/
redgpu
/
ezEngine
Обзор
Документация
Войти
/
redgpu
/
ezEngine
Код
Запросы
0
Задачи
Вики
Пакеты
0
Релизы
0
Аналитика
Безопасность
dev
Code/ThirdParty/MiniAudio/miniaudio.h
95 844 строки
4 MB
Jan Krassnigg
Updated MiniAudio to version 0.11.24 (#1783)
20 янв 2026, 12:01
Не верифицирован
20 янв 2026, 12:01
b18e629
Код
Авторство
О чём код?
/* Audio playback and capture library. Choice of public domain or MIT-0. See license statements at the end of this file. miniaudio - v0.11.24 - 2026-01-17 David Reid - mackron@gmail.com Website: https://miniaud.io Documentation: https://miniaud.io/docs GitHub: https://github.com/mackron/miniaudio */ /* 1. Introduction =============== To use miniaudio, just include "miniaudio.h" like any other header and add "miniaudio.c" to your source tree. If you don't want to add it to your source tree you can compile and link to it like any other library. Note that ABI compatibility is not guaranteed between versions, even with bug fix releases, so take care if compiling as a shared object. miniaudio includes both low level and high level APIs. The low level API is good for those who want to do all of their mixing themselves and only require a light weight interface to the underlying audio device. The high level API is good for those who have complex mixing and effect requirements. In miniaudio, objects are transparent structures. Unlike many other libraries, there are no handles to opaque objects which means you need to allocate memory for objects yourself. In the examples presented in this documentation you will often see objects declared on the stack. You need to be careful when translating these examples to your own code so that you don't accidentally declare your objects on the stack and then cause them to become invalid once the function returns. In addition, you must ensure the memory address of your objects remain the same throughout their lifetime. You therefore cannot be making copies of your objects. A config/init pattern is used throughout the entire library. The idea is that you set up a config object and pass that into the initialization routine. The advantage to this system is that the config object can be initialized with logical defaults and new properties added to it without breaking the API. The config object can be allocated on the stack and does not need to be maintained after initialization of the corresponding object. 1.1. Low Level API ------------------ The low level API gives you access to the raw audio data of an audio device. It supports playback, capture, full-duplex and loopback (WASAPI only). You can enumerate over devices to determine which physical device(s) you want to connect to. The low level API uses the concept of a "device" as the abstraction for physical devices. The idea is that you choose a physical device to emit or capture audio from, and then move data to/from the device when miniaudio tells you to. Data is delivered to and from devices asynchronously via a callback which you specify when initializing the device. When initializing the device you first need to configure it. The device configuration allows you to specify things like the format of the data delivered via the callback, the size of the internal buffer and the ID of the device you want to emit or capture audio from. Once you have the device configuration set up you can initialize the device. When initializing a device you need to allocate memory for the device object beforehand. This gives the application complete control over how the memory is allocated. In the example below we initialize a playback device on the stack, but you could allocate it on the heap if that suits your situation better. ```c void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount) { // In playback mode copy data to pOutput. In capture mode read data from pInput. In full-duplex mode, both // pOutput and pInput will be valid and you can move data from pInput into pOutput. Never process more than // frameCount frames. } int main() { ma_device_config config = ma_device_config_init(ma_device_type_playback); config.playback.format = ma_format_f32; // Set to ma_format_unknown to use the device's native format. config.playback.channels = 2; // Set to 0 to use the device's native channel count. config.sampleRate = 48000; // Set to 0 to use the device's native sample rate. config.dataCallback = data_callback; // This function will be called when miniaudio needs more data. config.pUserData = pMyCustomData; // Can be accessed from the device object (device.pUserData). ma_device device; if (ma_device_init(NULL, &config, &device) != MA_SUCCESS) { return -1; // Failed to initialize the device. } ma_device_start(&device); // The device is sleeping by default so you'll need to start it manually. // Do something here. Probably your program's main loop. ma_device_uninit(&device); return 0; } ``` In the example above, `data_callback()` is where audio data is written and read from the device. The idea is in playback mode you cause sound to be emitted from the speakers by writing audio data to the output buffer (`pOutput` in the example). In capture mode you read data from the input buffer (`pInput`) to extract sound captured by the microphone. The `frameCount` parameter tells you how many frames can be written to the output buffer and read from the input buffer. A "frame" is one sample for each channel. For example, in a stereo stream (2 channels), one frame is 2 samples: one for the left, one for the right. The channel count is defined by the device config. The size in bytes of an individual sample is defined by the sample format which is also specified in the device config. Multi-channel audio data is always interleaved, which means the samples for each frame are stored next to each other in memory. For example, in a stereo stream the first pair of samples will be the left and right samples for the first frame, the second pair of samples will be the left and right samples for the second frame, etc. The configuration of the device is defined by the `ma_device_config` structure. The config object is always initialized with `ma_device_config_init()`. It's important to always initialize the config with this function as it initializes it with logical defaults and ensures your program doesn't break when new members are added to the `ma_device_config` structure. The example above uses a fairly simple and standard device configuration. The call to `ma_device_config_init()` takes a single parameter, which is whether or not the device is a playback, capture, duplex or loopback device (loopback devices are not supported on all backends). The `config.playback.format` member sets the sample format which can be one of the following (all formats are native-endian): +---------------+----------------------------------------+---------------------------+ | Symbol | Description | Range | +---------------+----------------------------------------+---------------------------+ | ma_format_f32 | 32-bit floating point | [-1, 1] | | ma_format_s16 | 16-bit signed integer | [-32768, 32767] | | ma_format_s24 | 24-bit signed integer (tightly packed) | [-8388608, 8388607] | | ma_format_s32 | 32-bit signed integer | [-2147483648, 2147483647] | | ma_format_u8 | 8-bit unsigned integer | [0, 255] | +---------------+----------------------------------------+---------------------------+ The `config.playback.channels` member sets the number of channels to use with the device. The channel count cannot exceed MA_MAX_CHANNELS. The `config.sampleRate` member sets the sample rate (which must be the same for both playback and capture in full-duplex configurations). This is usually set to 44100 or 48000, but can be set to anything. It's recommended to keep this between 8000 and 384000, however. Note that leaving the format, channel count and/or sample rate at their default values will result in the internal device's native configuration being used which is useful if you want to avoid the overhead of miniaudio's automatic data conversion. In addition to the sample format, channel count and sample rate, the data callback and user data pointer are also set via the config. The user data pointer is not passed into the callback as a parameter, but is instead set to the `pUserData` member of `ma_device` which you can access directly since all miniaudio structures are transparent. Initializing the device is done with `ma_device_init()`. This will return a result code telling you what went wrong, if anything. On success it will return `MA_SUCCESS`. After initialization is complete the device will be in a stopped state. To start it, use `ma_device_start()`. Uninitializing the device will stop it, which is what the example above does, but you can also stop the device with `ma_device_stop()`. To resume the device simply call `ma_device_start()` again. Note that it's important to never stop or start the device from inside the callback. This will result in a deadlock. Instead you set a variable or signal an event indicating that the device needs to stop and handle it in a different thread. The following APIs must never be called inside the callback: ```c ma_device_init() ma_device_init_ex() ma_device_uninit() ma_device_start() ma_device_stop() ``` You must never try uninitializing and reinitializing a device inside the callback. You must also never try to stop and start it from inside the callback. There are a few other things you shouldn't do in the callback depending on your requirements, however this isn't so much a thread-safety thing, but rather a real-time processing thing which is beyond the scope of this introduction. The example above demonstrates the initialization of a playback device, but it works exactly the same for capture. All you need to do is change the device type from `ma_device_type_playback` to `ma_device_type_capture` when setting up the config, like so: ```c ma_device_config config = ma_device_config_init(ma_device_type_capture); config.capture.format = MY_FORMAT; config.capture.channels = MY_CHANNEL_COUNT; ``` In the data callback you just read from the input buffer (`pInput` in the example above) and leave the output buffer alone (it will be set to NULL when the device type is set to `ma_device_type_capture`). These are the available device types and how you should handle the buffers in the callback: +-------------------------+--------------------------------------------------------+ | Device Type | Callback Behavior | +-------------------------+--------------------------------------------------------+ | ma_device_type_playback | Write to output buffer, leave input buffer untouched. | | ma_device_type_capture | Read from input buffer, leave output buffer untouched. | | ma_device_type_duplex | Read from input buffer, write to output buffer. | | ma_device_type_loopback | Read from input buffer, leave output buffer untouched. | +-------------------------+--------------------------------------------------------+ You will notice in the example above that the sample format and channel count is specified separately for playback and capture. This is to support different data formats between the playback and capture devices in a full-duplex system. An example may be that you want to capture audio data as a monaural stream (one channel), but output sound to a stereo speaker system. Note that if you use different formats between playback and capture in a full-duplex configuration you will need to convert the data yourself. There are functions available to help you do this which will be explained later. The example above did not specify a physical device to connect to which means it will use the operating system's default device. If you have multiple physical devices connected and you want to use a specific one you will need to specify the device ID in the configuration, like so: ```c config.playback.pDeviceID = pMyPlaybackDeviceID; // Only if requesting a playback or duplex device. config.capture.pDeviceID = pMyCaptureDeviceID; // Only if requesting a capture, duplex or loopback device. ``` To retrieve the device ID you will need to perform device enumeration, however this requires the use of a new concept called the "context". Conceptually speaking the context sits above the device. There is one context to many devices. The purpose of the context is to represent the backend at a more global level and to perform operations outside the scope of an individual device. Mainly it is used for performing run-time linking against backend libraries, initializing backends and enumerating devices. The example below shows how to enumerate devices. ```c ma_context context; if (ma_context_init(NULL, 0, NULL, &context) != MA_SUCCESS) { // Error. } ma_device_info* pPlaybackInfos; ma_uint32 playbackCount; ma_device_info* pCaptureInfos; ma_uint32 captureCount; if (ma_context_get_devices(&context, &pPlaybackInfos, &playbackCount, &pCaptureInfos, &captureCount) != MA_SUCCESS) { // Error. } // Loop over each device info and do something with it. Here we just print the name with their index. You may want // to give the user the opportunity to choose which device they'd prefer. for (ma_uint32 iDevice = 0; iDevice < playbackCount; iDevice += 1) { printf("%d - %s\n", iDevice, pPlaybackInfos[iDevice].name); } ma_device_config config = ma_device_config_init(ma_device_type_playback); config.playback.pDeviceID = &pPlaybackInfos[chosenPlaybackDeviceIndex].id; config.playback.format = MY_FORMAT; config.playback.channels = MY_CHANNEL_COUNT; config.sampleRate = MY_SAMPLE_RATE; config.dataCallback = data_callback; config.pUserData = pMyCustomData; ma_device device; if (ma_device_init(&context, &config, &device) != MA_SUCCESS) { // Error } ... ma_device_uninit(&device); ma_context_uninit(&context); ``` The first thing we do in this example is initialize a `ma_context` object with `ma_context_init()`. The first parameter is a pointer to a list of `ma_backend` values which are used to override the default backend priorities. When this is NULL, as in this example, miniaudio's default priorities are used. The second parameter is the number of backends listed in the array pointed to by the first parameter. The third parameter is a pointer to a `ma_context_config` object which can be NULL, in which case defaults are used. The context configuration is used for setting the logging callback, custom memory allocation callbacks, user-defined data and some backend-specific configurations. Once the context has been initialized you can enumerate devices. In the example above we use the simpler `ma_context_get_devices()`, however you can also use a callback for handling devices by using `ma_context_enumerate_devices()`. When using `ma_context_get_devices()` you provide a pointer to a pointer that will, upon output, be set to a pointer to a buffer containing a list of `ma_device_info` structures. You also provide a pointer to an unsigned integer that will receive the number of items in the returned buffer. Do not free the returned buffers as their memory is managed internally by miniaudio. The `ma_device_info` structure contains an `id` member which is the ID you pass to the device config. It also contains the name of the device which is useful for presenting a list of devices to the user via the UI. When creating your own context you will want to pass it to `ma_device_init()` when initializing the device. Passing in NULL, like we do in the first example, will result in miniaudio creating the context for you, which you don't want to do since you've already created a context. Note that internally the context is only tracked by it's pointer which means you must not change the location of the `ma_context` object. If this is an issue, consider using `malloc()` to allocate memory for the context. 1.2. High Level API ------------------- The high level API consists of three main parts: * Resource management for loading and streaming sounds. * A node graph for advanced mixing and effect processing. * A high level "engine" that wraps around the resource manager and node graph. The resource manager (`ma_resource_manager`) is used for loading sounds. It supports loading sounds fully into memory and also streaming. It will also deal with reference counting for you which avoids the same sound being loaded multiple times. The node graph is used for mixing and effect processing. The idea is that you connect a number of nodes into the graph by connecting each node's outputs to another node's inputs. Each node can implement its own effect. By chaining nodes together, advanced mixing and effect processing can be achieved. The engine encapsulates both the resource manager and the node graph to create a simple, easy to use high level API. The resource manager and node graph APIs are covered in more later sections of this manual. The code below shows how you can initialize an engine using its default configuration. ```c ma_result result; ma_engine engine; result = ma_engine_init(NULL, &engine); if (result != MA_SUCCESS) { return result; // Failed to initialize the engine. } ``` This creates an engine instance which will initialize a device internally which you can access with `ma_engine_get_device()`. It will also initialize a resource manager for you which can be accessed with `ma_engine_get_resource_manager()`. The engine itself is a node graph (`ma_node_graph`) which means you can pass a pointer to the engine object into any of the `ma_node_graph` APIs (with a cast). Alternatively, you can use `ma_engine_get_node_graph()` instead of a cast. Note that all objects in miniaudio, including the `ma_engine` object in the example above, are transparent structures. There are no handles to opaque structures in miniaudio which means you need to be mindful of how you declare them. In the example above we are declaring it on the stack, but this will result in the struct being invalidated once the function encapsulating it returns. If allocating the engine on the heap is more appropriate, you can easily do so with a standard call to `malloc()` or whatever heap allocation routine you like: ```c ma_engine* pEngine = malloc(sizeof(*pEngine)); ``` The `ma_engine` API uses the same config/init pattern used all throughout miniaudio. To configure an engine, you can fill out a `ma_engine_config` object and pass it into the first parameter of `ma_engine_init()`: ```c ma_result result; ma_engine engine; ma_engine_config engineConfig; engineConfig = ma_engine_config_init(); engineConfig.pResourceManager = &myCustomResourceManager; // <-- Initialized as some earlier stage. result = ma_engine_init(&engineConfig, &engine); if (result != MA_SUCCESS) { return result; } ``` This creates an engine instance using a custom config. In this particular example it's showing how you can specify a custom resource manager rather than having the engine initialize one internally. This is particularly useful if you want to have multiple engine's share the same resource manager. The engine must be uninitialized with `ma_engine_uninit()` when it's no longer needed. By default the engine will be started, but nothing will be playing because no sounds have been initialized. The easiest but least flexible way of playing a sound is like so: ```c ma_engine_play_sound(&engine, "my_sound.wav", NULL); ``` This plays what miniaudio calls an "inline" sound. It plays the sound once, and then puts the internal sound up for recycling. The last parameter is used to specify which sound group the sound should be associated with which will be explained later. This particular way of playing a sound is simple, but lacks flexibility and features. A more flexible way of playing a sound is to first initialize a sound: ```c ma_result result; ma_sound sound; result = ma_sound_init_from_file(&engine, "my_sound.wav", 0, NULL, NULL, &sound); if (result != MA_SUCCESS) { return result; } ma_sound_start(&sound); ``` This returns a `ma_sound` object which represents a single instance of the specified sound file. If you want to play the same file multiple times simultaneously, you need to create one sound for each instance. Sounds should be uninitialized with `ma_sound_uninit()`. Sounds are not started by default. Start a sound with `ma_sound_start()` and stop it with `ma_sound_stop()`. When a sound is stopped, it is not rewound to the start. Use `ma_sound_seek_to_pcm_frame(&sound, 0)` to seek back to the start of a sound. By default, starting and stopping sounds happens immediately, but sometimes it might be convenient to schedule the sound to be started and/or stopped at a specific time. This can be done with the following functions: ```c ma_sound_set_start_time_in_pcm_frames() ma_sound_set_start_time_in_milliseconds() ma_sound_set_stop_time_in_pcm_frames() ma_sound_set_stop_time_in_milliseconds() ``` The start/stop time needs to be specified based on the absolute timer which is controlled by the engine. The current global time in PCM frames can be retrieved with `ma_engine_get_time_in_pcm_frames()`. The engine's global time can be changed with `ma_engine_set_time_in_pcm_frames()` for synchronization purposes if required. Note that scheduling a start time still requires an explicit call to `ma_sound_start()` before anything will play: ```c ma_sound_set_start_time_in_pcm_frames(&sound, ma_engine_get_time_in_pcm_frames(&engine) + (ma_engine_get_sample_rate(&engine) * 2); ma_sound_start(&sound); ``` The third parameter of `ma_sound_init_from_file()` is a set of flags that control how the sound be loaded and a few options on which features should be enabled for that sound. By default, the sound is synchronously loaded fully into memory straight from the file system without any kind of decoding. If you want to decode the sound before storing it in memory, you need to specify the `MA_SOUND_FLAG_DECODE` flag. This is useful if you want to incur the cost of decoding at an earlier stage, such as a loading stage. Without this option, decoding will happen dynamically at mixing time which might be too expensive on the audio thread. If you want to load the sound asynchronously, you can specify the `MA_SOUND_FLAG_ASYNC` flag. This will result in `ma_sound_init_from_file()` returning quickly, but the sound will not start playing until the sound has had some audio decoded. The fourth parameter is a pointer to sound group. A sound group is used as a mechanism to organise sounds into groups which have their own effect processing and volume control. An example is a game which might have separate groups for sfx, voice and music. Each of these groups have their own independent volume control. Use `ma_sound_group_init()` or `ma_sound_group_init_ex()` to initialize a sound group. Sounds and sound groups are nodes in the engine's node graph and can be plugged into any `ma_node` API. This makes it possible to connect sounds and sound groups to effect nodes to produce complex effect chains. A sound can have its volume changed with `ma_sound_set_volume()`. If you prefer decibel volume control you can use `ma_volume_db_to_linear()` to convert from decibel representation to linear. Panning and pitching is supported with `ma_sound_set_pan()` and `ma_sound_set_pitch()`. If you know a sound will never have its pitch changed with `ma_sound_set_pitch()` or via the doppler effect, you can specify the `MA_SOUND_FLAG_NO_PITCH` flag when initializing the sound for an optimization. By default, sounds and sound groups have spatialization enabled. If you don't ever want to spatialize your sounds, initialize the sound with the `MA_SOUND_FLAG_NO_SPATIALIZATION` flag. The spatialization model is fairly simple and is roughly on feature parity with OpenAL. HRTF and environmental occlusion are not currently supported, but planned for the future. The supported features include: * Sound and listener positioning and orientation with cones * Attenuation models: none, inverse, linear and exponential * Doppler effect Sounds can be faded in and out with `ma_sound_set_fade_in_pcm_frames()`. To check if a sound is currently playing, you can use `ma_sound_is_playing()`. To check if a sound is at the end, use `ma_sound_at_end()`. Looping of a sound can be controlled with `ma_sound_set_looping()`. Use `ma_sound_is_looping()` to check whether or not the sound is looping. 2. Building =========== miniaudio should work cleanly out of the box without the need to download or install any dependencies. See below for platform-specific details. This library has been designed to be added directly to your source tree which is the preferred way of using it, but you can compile it as a normal library if that's your preference. Be careful if compiling as a shared object because miniaudio is not ABI compatible between any release, including bug fix releases. It's recommended you link statically. Note that GCC and Clang require `-msse2`, `-mavx2`, etc. for SIMD optimizations. If you get errors about undefined references to `__sync_val_compare_and_swap_8`, `__atomic_load_8`, etc. you need to link with `-latomic`. 2.1. Windows ------------ The Windows build should compile cleanly on all popular compilers without the need to configure any include paths nor link to any libraries. The UWP build may require linking to mmdevapi.lib if you get errors about an unresolved external symbol for `ActivateAudioInterfaceAsync()`. 2.2. macOS and iOS ------------------ The macOS build should compile cleanly without the need to download any dependencies nor link to any libraries or frameworks. The iOS build needs to be compiled as Objective-C and will need to link the relevant frameworks but should compile cleanly out of the box with Xcode. Compiling through the command line requires linking to `-lpthread` and `-lm`. Due to the way miniaudio links to frameworks at runtime, your application may not pass Apple's notarization process. To fix this there are two options. The first is to compile with `-DMA_NO_RUNTIME_LINKING` which in turn will require linking with `-framework CoreFoundation -framework CoreAudio -framework AudioToolbox`. If you get errors about AudioToolbox, try with `-framework AudioUnit` instead. You may get this when using older versions of iOS. Alternatively, if you would rather keep using runtime linking you can add the following to your entitlements.xcent file: ``` <key>com.apple.security.cs.allow-dyld-environment-variables</key> <true/> <key>com.apple.security.cs.allow-unsigned-executable-memory</key> <true/> ``` See this discussion for more info: https://github.com/mackron/miniaudio/issues/203. 2.3. Linux ---------- The Linux build only requires linking to `-ldl`, `-lpthread` and `-lm`. You do not need any development packages. You may need to link with `-latomic` if you're compiling for 32-bit ARM. 2.4. BSD -------- The BSD build only requires linking to `-lpthread` and `-lm`. NetBSD uses audio(4), OpenBSD uses sndio and FreeBSD uses OSS. You may need to link with `-latomic` if you're compiling for 32-bit ARM. 2.5. Android ------------ AAudio is the highest priority backend on Android. This should work out of the box without needing any kind of compiler configuration. Support for AAudio starts with Android 8 which means older versions will fall back to OpenSL|ES which requires API level 16+. There have been reports that the OpenSL|ES backend fails to initialize on some Android based devices due to `dlopen()` failing to open "libOpenSLES.so". If this happens on your platform you'll need to disable run-time linking with `MA_NO_RUNTIME_LINKING` and link with -lOpenSLES. 2.6. Emscripten --------------- The Emscripten build emits Web Audio JavaScript directly and should compile cleanly out of the box. You cannot use `-std=c*` compiler flags, nor `-ansi`. You can enable the use of AudioWorklets by defining `MA_ENABLE_AUDIO_WORKLETS` and then compiling with the following options: -sAUDIO_WORKLET=1 -sWASM_WORKERS=1 -sASYNCIFY An example for compiling with AudioWorklet support might look like this: emcc program.c -o bin/program.html -DMA_ENABLE_AUDIO_WORKLETS -sAUDIO_WORKLET=1 -sWASM_WORKERS=1 -sASYNCIFY To run locally, you'll need to use emrun: emrun bin/program.html 2.7. Build Options ------------------ `#define` these options before including miniaudio.c, or pass them as compiler flags: +----------------------------------+--------------------------------------------------------------------+ | Option | Description | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_WASAPI | Disables the WASAPI backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_DSOUND | Disables the DirectSound backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_WINMM | Disables the WinMM backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_ALSA | Disables the ALSA backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_PULSEAUDIO | Disables the PulseAudio backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_JACK | Disables the JACK backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_COREAUDIO | Disables the Core Audio backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_SNDIO | Disables the sndio backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_AUDIO4 | Disables the audio(4) backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_OSS | Disables the OSS backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_AAUDIO | Disables the AAudio backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_OPENSL | Disables the OpenSL|ES backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_WEBAUDIO | Disables the Web Audio backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_CUSTOM | Disables support for custom backends. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_NULL | Disables the null backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_ONLY_SPECIFIC_BACKENDS | Disables all backends by default and requires `MA_ENABLE_*` to | | | enable specific backends. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_WASAPI | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the WASAPI backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_DSOUND | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the DirectSound backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_WINMM | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the WinMM backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_ALSA | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the ALSA backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_PULSEAUDIO | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the PulseAudio backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_JACK | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the JACK backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_COREAUDIO | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the Core Audio backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_SNDIO | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the sndio backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_AUDIO4 | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the audio(4) backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_OSS | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the OSS backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_AAUDIO | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the AAudio backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_OPENSL | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the OpenSL|ES backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_WEBAUDIO | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the Web Audio backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_CUSTOM | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable custom backends. | +----------------------------------+--------------------------------------------------------------------+ | MA_ENABLE_NULL | Used in conjunction with MA_ENABLE_ONLY_SPECIFIC_BACKENDS to | | | enable the null backend. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_DECODING | Disables decoding APIs. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_ENCODING | Disables encoding APIs. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_WAV | Disables the built-in WAV decoder and encoder. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_FLAC | Disables the built-in FLAC decoder. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_MP3 | Disables the built-in MP3 decoder. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_DEVICE_IO | Disables playback and recording. This will disable `ma_context` | | | and `ma_device` APIs. This is useful if you only want to use | | | miniaudio's data conversion and/or decoding APIs. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_RESOURCE_MANAGER | Disables the resource manager. When using the engine this will | | | also disable the following functions: | | | | | | ``` | | | ma_sound_init_from_file() | | | ma_sound_init_from_file_w() | | | ma_sound_init_copy() | | | ma_engine_play_sound_ex() | | | ma_engine_play_sound() | | | ``` | | | | | | The only way to initialize a `ma_sound` object is to initialize it | | | from a data source. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_NODE_GRAPH | Disables the node graph API. This will also disable the engine API | | | because it depends on the node graph. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_ENGINE | Disables the engine API. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_THREADING | Disables the `ma_thread`, `ma_mutex`, `ma_semaphore` and | | | `ma_event` APIs. This option is useful if you only need to use | | | miniaudio for data conversion, decoding and/or encoding. Some | | | families of APIs require threading which means the following | | | options must also be set: | | | | | | ``` | | | MA_NO_DEVICE_IO | | | ``` | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_GENERATION | Disables generation APIs such a `ma_waveform` and `ma_noise`. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_SSE2 | Disables SSE2 optimizations. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_AVX2 | Disables AVX2 optimizations. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_NEON | Disables NEON optimizations. | +----------------------------------+--------------------------------------------------------------------+ | MA_NO_RUNTIME_LINKING | Disables runtime linking. This is useful for passing Apple's | | | notarization process. When enabling this, you may need to avoid | | | using `-std=c89` or `-std=c99` on Linux builds or else you may end | | | up with compilation errors due to conflicts with `timespec` and | | | `timeval` data types. | | | | | | You may need to enable this if your target platform does not allow | | | runtime linking via `dlopen()`. | +----------------------------------+--------------------------------------------------------------------+ | MA_USE_STDINT | (Pass this in as a compiler flag. Do not `#define` this before | | | miniaudio.c) Forces the use of stdint.h for sized types. | +----------------------------------+--------------------------------------------------------------------+ | MA_DEBUG_OUTPUT | Enable `printf()` output of debug logs (`MA_LOG_LEVEL_DEBUG`). | +----------------------------------+--------------------------------------------------------------------+ | MA_COINIT_VALUE | Windows only. The value to pass to internal calls to | | | `CoInitializeEx()`. Defaults to `COINIT_MULTITHREADED`. | +----------------------------------+--------------------------------------------------------------------+ | MA_FORCE_UWP | Windows only. Affects only the WASAPI backend. Will force the | | | WASAPI backend to use the UWP code path instead of the regular | | | desktop path. This is normally auto-detected and should rarely be | | | needed to be used explicitly, but can be useful for debugging. | +----------------------------------+--------------------------------------------------------------------+ | MA_ON_THREAD_ENTRY | Defines some code that will be executed as soon as an internal | | | miniaudio-managed thread is created. This will be the first thing | | | to be executed by the thread entry point. | +----------------------------------+--------------------------------------------------------------------+ | MA_ON_THREAD_EXIT | Defines some code that will be executed from the entry point of an | | | internal miniaudio-managed thread upon exit. This will be the last | | | thing to be executed before the thread's entry point exits. | +----------------------------------+--------------------------------------------------------------------+ | MA_THREAD_DEFAULT_STACK_SIZE | If set, specifies the default stack size used by miniaudio-managed | | | threads. | +----------------------------------+--------------------------------------------------------------------+ | MA_API | Controls how public APIs should be decorated. Default is `extern`. | +----------------------------------+--------------------------------------------------------------------+ 3. Definitions ============== This section defines common terms used throughout miniaudio. Unfortunately there is often ambiguity in the use of terms throughout the audio space, so this section is intended to clarify how miniaudio uses each term. 3.1. Sample ----------- A sample is a single unit of audio data. If the sample format is f32, then one sample is one 32-bit floating point number. 3.2. Frame / PCM Frame ---------------------- A frame is a group of samples equal to the number of channels. For a stereo stream a frame is 2 samples, a mono frame is 1 sample, a 5.1 surround sound frame is 6 samples, etc. The terms "frame" and "PCM frame" are the same thing in miniaudio. Note that this is different to a compressed frame. If ever miniaudio needs to refer to a compressed frame, such as a FLAC frame, it will always clarify what it's referring to with something like "FLAC frame". 3.3. Channel ------------ A stream of monaural audio that is emitted from an individual speaker in a speaker system, or received from an individual microphone in a microphone system. A stereo stream has two channels (a left channel, and a right channel), a 5.1 surround sound system has 6 channels, etc. Some audio systems refer to a channel as a complex audio stream that's mixed with other channels to produce the final mix - this is completely different to miniaudio's use of the term "channel" and should not be confused. 3.4. Sample Rate ---------------- The sample rate in miniaudio is always expressed in Hz, such as 44100, 48000, etc. It's the number of PCM frames that are processed per second. 3.5. Formats ------------ Throughout miniaudio you will see references to different sample formats: +---------------+----------------------------------------+---------------------------+ | Symbol | Description | Range | +---------------+----------------------------------------+---------------------------+ | ma_format_f32 | 32-bit floating point | [-1, 1] | | ma_format_s16 | 16-bit signed integer | [-32768, 32767] | | ma_format_s24 | 24-bit signed integer (tightly packed) | [-8388608, 8388607] | | ma_format_s32 | 32-bit signed integer | [-2147483648, 2147483647] | | ma_format_u8 | 8-bit unsigned integer | [0, 255] | +---------------+----------------------------------------+---------------------------+ All formats are native-endian. 4. Data Sources =============== The data source abstraction in miniaudio is used for retrieving audio data from some source. A few examples include `ma_decoder`, `ma_noise` and `ma_waveform`. You will need to be familiar with data sources in order to make sense of some of the higher level concepts in miniaudio. The `ma_data_source` API is a generic interface for reading from a data source. Any object that implements the data source interface can be plugged into any `ma_data_source` function. To read data from a data source: ```c ma_result result; ma_uint64 framesRead; result = ma_data_source_read_pcm_frames(pDataSource, pFramesOut, frameCount, &framesRead); if (result != MA_SUCCESS) { return result; // Failed to read data from the data source. } ``` If you don't need the number of frames that were successfully read you can pass in `NULL` to the `pFramesRead` parameter. If this returns a value less than the number of frames requested it means the end of the file has been reached. `MA_AT_END` will be returned only when the number of frames read is 0. When calling any data source function, with the exception of `ma_data_source_init()` and `ma_data_source_uninit()`, you can pass in any object that implements a data source. For example, you could plug in a decoder like so: ```c ma_result result; ma_uint64 framesRead; ma_decoder decoder; // <-- This would be initialized with `ma_decoder_init_*()`. result = ma_data_source_read_pcm_frames(&decoder, pFramesOut, frameCount, &framesRead); if (result != MA_SUCCESS) { return result; // Failed to read data from the decoder. } ``` If you want to seek forward you can pass in `NULL` to the `pFramesOut` parameter. Alternatively you can use `ma_data_source_seek_pcm_frames()`. To seek to a specific PCM frame: ```c result = ma_data_source_seek_to_pcm_frame(pDataSource, frameIndex); if (result != MA_SUCCESS) { return result; // Failed to seek to PCM frame. } ``` You can retrieve the total length of a data source in PCM frames, but note that some data sources may not have the notion of a length, such as noise and waveforms, and others may just not have a way of determining the length such as some decoders. To retrieve the length: ```c ma_uint64 length; result = ma_data_source_get_length_in_pcm_frames(pDataSource, &length); if (result != MA_SUCCESS) { return result; // Failed to retrieve the length. } ``` Care should be taken when retrieving the length of a data source where the underlying decoder is pulling data from a data stream with an undefined length, such as internet radio or some kind of broadcast. If you do this, `ma_data_source_get_length_in_pcm_frames()` may never return. The current position of the cursor in PCM frames can also be retrieved: ```c ma_uint64 cursor; result = ma_data_source_get_cursor_in_pcm_frames(pDataSource, &cursor); if (result != MA_SUCCESS) { return result; // Failed to retrieve the cursor. } ``` You will often need to know the data format that will be returned after reading. This can be retrieved like so: ```c ma_format format; ma_uint32 channels; ma_uint32 sampleRate; ma_channel channelMap[MA_MAX_CHANNELS]; result = ma_data_source_get_data_format(pDataSource, &format, &channels, &sampleRate, channelMap, MA_MAX_CHANNELS); if (result != MA_SUCCESS) { return result; // Failed to retrieve data format. } ``` If you do not need a specific data format property, just pass in NULL to the respective parameter. There may be cases where you want to implement something like a sound bank where you only want to read data within a certain range of the underlying data. To do this you can use a range: ```c result = ma_data_source_set_range_in_pcm_frames(pDataSource, rangeBegInFrames, rangeEndInFrames); if (result != MA_SUCCESS) { return result; // Failed to set the range. } ``` This is useful if you have a sound bank where many sounds are stored in the same file and you want the data source to only play one of those sub-sounds. Note that once the range is set, everything that takes a position, such as cursors and loop points, should always be relative to the start of the range. When the range is set, any previously defined loop point will be reset. Custom loop points can also be used with data sources. By default, data sources will loop after they reach the end of the data source, but if you need to loop at a specific location, you can do the following: ```c result = ma_data_source_set_loop_point_in_pcm_frames(pDataSource, loopBegInFrames, loopEndInFrames); if (result != MA_SUCCESS) { return result; // Failed to set the loop point. } ``` The loop point is relative to the current range. It's sometimes useful to chain data sources together so that a seamless transition can be achieved. To do this, you can use chaining: ```c ma_decoder decoder1; ma_decoder decoder2; // ... initialize decoders with ma_decoder_init_*() ... result = ma_data_source_set_next(&decoder1, &decoder2); if (result != MA_SUCCESS) { return result; // Failed to set the next data source. } result = ma_data_source_read_pcm_frames(&decoder1, pFramesOut, frameCount, pFramesRead); if (result != MA_SUCCESS) { return result; // Failed to read from the decoder. } ``` In the example above we're using decoders. When reading from a chain, you always want to read from the top level data source in the chain. In the example above, `decoder1` is the top level data source in the chain. When `decoder1` reaches the end, `decoder2` will start seamlessly without any gaps. Note that when looping is enabled, only the current data source will be looped. You can loop the entire chain by linking in a loop like so: ```c ma_data_source_set_next(&decoder1, &decoder2); // decoder1 -> decoder2 ma_data_source_set_next(&decoder2, &decoder1); // decoder2 -> decoder1 (loop back to the start). ``` Note that setting up chaining is not thread safe, so care needs to be taken if you're dynamically changing links while the audio thread is in the middle of reading. Do not use `ma_decoder_seek_to_pcm_frame()` as a means to reuse a data source to play multiple instances of the same sound simultaneously. This can be extremely inefficient depending on the type of data source and can result in glitching due to subtle changes to the state of internal filters. Instead, initialize multiple data sources for each instance. 4.1. Custom Data Sources ------------------------ You can implement a custom data source by implementing the functions in `ma_data_source_vtable`. Your custom object must have `ma_data_source_base` as it's first member: ```c struct my_data_source { ma_data_source_base base; ... }; ``` In your initialization routine, you need to call `ma_data_source_init()` in order to set up the base object (`ma_data_source_base`): ```c static ma_result my_data_source_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead) { // Read data here. Output in the same format returned by my_data_source_get_data_format(). } static ma_result my_data_source_seek(ma_data_source* pDataSource, ma_uint64 frameIndex) { // Seek to a specific PCM frame here. Return MA_NOT_IMPLEMENTED if seeking is not supported. } static ma_result my_data_source_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap) { // Return the format of the data here. } static ma_result my_data_source_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor) { // Retrieve the current position of the cursor here. Return MA_NOT_IMPLEMENTED and set *pCursor to 0 if there is no notion of a cursor. } static ma_result my_data_source_get_length(ma_data_source* pDataSource, ma_uint64* pLength) { // Retrieve the length in PCM frames here. Return MA_NOT_IMPLEMENTED and set *pLength to 0 if there is no notion of a length or if the length is unknown. } static ma_data_source_vtable g_my_data_source_vtable = { my_data_source_read, my_data_source_seek, my_data_source_get_data_format, my_data_source_get_cursor, my_data_source_get_length }; ma_result my_data_source_init(my_data_source* pMyDataSource) { ma_result result; ma_data_source_config baseConfig; baseConfig = ma_data_source_config_init(); baseConfig.vtable = &g_my_data_source_vtable; result = ma_data_source_init(&baseConfig, &pMyDataSource->base); if (result != MA_SUCCESS) { return result; } // ... do the initialization of your custom data source here ... return MA_SUCCESS; } void my_data_source_uninit(my_data_source* pMyDataSource) { // ... do the uninitialization of your custom data source here ... // You must uninitialize the base data source. ma_data_source_uninit(&pMyDataSource->base); } ``` Note that `ma_data_source_init()` and `ma_data_source_uninit()` are never called directly outside of the custom data source. It's up to the custom data source itself to call these within their own init/uninit functions. 5. Engine ========= The `ma_engine` API is a high level API for managing and mixing sounds and effect processing. The `ma_engine` object encapsulates a resource manager and a node graph, both of which will be explained in more detail later. Sounds are called `ma_sound` and are created from an engine. Sounds can be associated with a mixing group called `ma_sound_group` which are also created from the engine. Both `ma_sound` and `ma_sound_group` objects are nodes within the engine's node graph. When the engine is initialized, it will normally create a device internally. If you would rather manage the device yourself, you can do so and just pass a pointer to it via the engine config when you initialize the engine. You can also just use the engine without a device, which again can be configured via the engine config. The most basic way to initialize the engine is with a default config, like so: ```c ma_result result; ma_engine engine; result = ma_engine_init(NULL, &engine); if (result != MA_SUCCESS) { return result; // Failed to initialize the engine. } ``` This will result in the engine initializing a playback device using the operating system's default device. This will be sufficient for many use cases, but if you need more flexibility you'll want to configure the engine with an engine config: ```c ma_result result; ma_engine engine; ma_engine_config engineConfig; engineConfig = ma_engine_config_init(); engineConfig.pDevice = &myDevice; result = ma_engine_init(&engineConfig, &engine); if (result != MA_SUCCESS) { return result; // Failed to initialize the engine. } ``` In the example above we're passing in a pre-initialized device. Since the caller is the one in control of the device's data callback, it's their responsibility to manually call `ma_engine_read_pcm_frames()` from inside their data callback: ```c void playback_data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount) { ma_engine_read_pcm_frames(&g_Engine, pOutput, frameCount, NULL); } ``` You can also use the engine independent of a device entirely: ```c ma_result result; ma_engine engine; ma_engine_config engineConfig; engineConfig = ma_engine_config_init(); engineConfig.noDevice = MA_TRUE; engineConfig.channels = 2; // Must be set when not using a device. engineConfig.sampleRate = 48000; // Must be set when not using a device. result = ma_engine_init(&engineConfig, &engine); if (result != MA_SUCCESS) { return result; // Failed to initialize the engine. } ``` Note that when you're not using a device, you must set the channel count and sample rate in the config or else miniaudio won't know what to use (miniaudio will use the device to determine this normally). When not using a device, you need to use `ma_engine_read_pcm_frames()` to process audio data from the engine. This kind of setup is useful if you want to do something like offline processing or want to use a different audio system for playback such as SDL. When a sound is loaded it goes through a resource manager. By default the engine will initialize a resource manager internally, but you can also specify a pre-initialized resource manager: ```c ma_result result; ma_engine engine1; ma_engine engine2; ma_engine_config engineConfig; engineConfig = ma_engine_config_init(); engineConfig.pResourceManager = &myResourceManager; ma_engine_init(&engineConfig, &engine1); ma_engine_init(&engineConfig, &engine2); ``` In this example we are initializing two engines, both of which are sharing the same resource manager. This is especially useful for saving memory when loading the same file across multiple engines. If you were not to use a shared resource manager, each engine instance would use their own which would result in any sounds that are used between both engine's being loaded twice. By using a shared resource manager, it would only be loaded once. Using multiple engine's is useful when you need to output to multiple playback devices, such as in a local multiplayer game where each player is using their own set of headphones. By default an engine will be in a started state. To make it so the engine is not automatically started you can configure it as such: ```c engineConfig.noAutoStart = MA_TRUE; // The engine will need to be started manually. ma_engine_start(&engine); // Later on the engine can be stopped with ma_engine_stop(). ma_engine_stop(&engine); ``` The concept of starting or stopping an engine is only relevant when using the engine with a device. Attempting to start or stop an engine that is not associated with a device will result in `MA_INVALID_OPERATION`. The master volume of the engine can be controlled with `ma_engine_set_volume()` which takes a linear scale, with 0 resulting in silence and anything above 1 resulting in amplification. If you prefer decibel based volume control, use `ma_volume_db_to_linear()` to convert from dB to linear. When a sound is spatialized, it is done so relative to a listener. An engine can be configured to have multiple listeners which can be configured via the config: ```c engineConfig.listenerCount = 2; ``` The maximum number of listeners is restricted to `MA_ENGINE_MAX_LISTENERS`. By default, when a sound is spatialized, it will be done so relative to the closest listener. You can also pin a sound to a specific listener which will be explained later. Listener's have a position, direction, cone, and velocity (for doppler effect). A listener is referenced by an index, the meaning of which is up to the caller (the index is 0 based and cannot go beyond the listener count, minus 1). The position, direction and velocity are all specified in absolute terms: ```c ma_engine_listener_set_position(&engine, listenerIndex, worldPosX, worldPosY, worldPosZ); ``` The direction of the listener represents it's forward vector. The listener's up vector can also be specified and defaults to +1 on the Y axis. ```c ma_engine_listener_set_direction(&engine, listenerIndex, forwardX, forwardY, forwardZ); ma_engine_listener_set_world_up(&engine, listenerIndex, 0, 1, 0); ``` The engine supports directional attenuation. The listener can have a cone the controls how sound is attenuated based on the listener's direction. When a sound is between the inner and outer cones, it will be attenuated between 1 and the cone's outer gain: ```c ma_engine_listener_set_cone(&engine, listenerIndex, innerAngleInRadians, outerAngleInRadians, outerGain); ``` When a sound is inside the inner code, no directional attenuation is applied. When the sound is outside of the outer cone, the attenuation will be set to `outerGain` in the example above. When the sound is in between the inner and outer cones, the attenuation will be interpolated between 1 and the outer gain. The engine's coordinate system follows the OpenGL coordinate system where positive X points right, positive Y points up and negative Z points forward. The simplest and least flexible way to play a sound is like so: ```c ma_engine_play_sound(&engine, "my_sound.wav", pGroup); ``` This is a "fire and forget" style of function. The engine will manage the `ma_sound` object internally. When the sound finishes playing, it'll be put up for recycling. For more flexibility you'll want to initialize a sound object: ```c ma_sound sound; result = ma_sound_init_from_file(&engine, "my_sound.wav", flags, pGroup, NULL, &sound); if (result != MA_SUCCESS) { return result; // Failed to load sound. } ``` Sounds need to be uninitialized with `ma_sound_uninit()`. The example above loads a sound from a file. If the resource manager has been disabled you will not be able to use this function and instead you'll need to initialize a sound directly from a data source: ```c ma_sound sound; result = ma_sound_init_from_data_source(&engine, &dataSource, flags, pGroup, &sound); if (result != MA_SUCCESS) { return result; } ``` Each `ma_sound` object represents a single instance of the sound. If you want to play the same sound multiple times at the same time, you need to initialize a separate `ma_sound` object. For the most flexibility when initializing sounds, use `ma_sound_init_ex()`. This uses miniaudio's standard config/init pattern: ```c ma_sound sound; ma_sound_config soundConfig; soundConfig = ma_sound_config_init(); soundConfig.pFilePath = NULL; // Set this to load from a file path. soundConfig.pDataSource = NULL; // Set this to initialize from an existing data source. soundConfig.pInitialAttachment = &someNodeInTheNodeGraph; soundConfig.initialAttachmentInputBusIndex = 0; soundConfig.channelsIn = 1; soundConfig.channelsOut = 0; // Set to 0 to use the engine's native channel count. result = ma_sound_init_ex(&soundConfig, &sound); if (result != MA_SUCCESS) { return result; } ``` In the example above, the sound is being initialized without a file nor a data source. This is valid, in which case the sound acts as a node in the middle of the node graph. This means you can connect other sounds to this sound and allow it to act like a sound group. Indeed, this is exactly what a `ma_sound_group` is. When loading a sound, you specify a set of flags that control how the sound is loaded and what features are enabled for that sound. When no flags are set, the sound will be fully loaded into memory in exactly the same format as how it's stored on the file system. The resource manager will allocate a block of memory and then load the file directly into it. When reading audio data, it will be decoded dynamically on the fly. In order to save processing time on the audio thread, it might be beneficial to pre-decode the sound. You can do this with the `MA_SOUND_FLAG_DECODE` flag: ```c ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_DECODE, pGroup, NULL, &sound); ``` By default, sounds will be loaded synchronously, meaning `ma_sound_init_*()` will not return until the sound has been fully loaded. If this is prohibitive you can instead load sounds asynchronously by specifying the `MA_SOUND_FLAG_ASYNC` flag: ```c ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_DECODE | MA_SOUND_FLAG_ASYNC, pGroup, NULL, &sound); ``` This will result in `ma_sound_init_*()` returning quickly, but the sound won't yet have been fully loaded. When you start the sound, it won't output anything until some sound is available. The sound will start outputting audio before the sound has been fully decoded when the `MA_SOUND_FLAG_DECODE` is specified. If you need to wait for an asynchronously loaded sound to be fully loaded, you can use a fence. A fence in miniaudio is a simple synchronization mechanism which simply blocks until it's internal counter hit's zero. You can specify a fence like so: ```c ma_result result; ma_fence fence; ma_sound sounds[4]; result = ma_fence_init(&fence); if (result != MA_SUCCESS) { return result; } // Load some sounds asynchronously. for (int iSound = 0; iSound < 4; iSound += 1) { ma_sound_init_from_file(&engine, mySoundFilesPaths[iSound], MA_SOUND_FLAG_DECODE | MA_SOUND_FLAG_ASYNC, pGroup, &fence, &sounds[iSound]); } // ... do some other stuff here in the mean time ... // Wait for all sounds to finish loading. ma_fence_wait(&fence); ``` If loading the entire sound into memory is prohibitive, you can also configure the engine to stream the audio data: ```c ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_STREAM, pGroup, NULL, &sound); ``` When streaming sounds, 2 seconds worth of audio data is stored in memory. Although it should work fine, it's inefficient to use streaming for short sounds. Streaming is useful for things like music tracks in games. When loading a sound from a file path, the engine will reference count the file to prevent it from being loaded if it's already in memory. When you uninitialize a sound, the reference count will be decremented, and if it hits zero, the sound will be unloaded from memory. This reference counting system is not used for streams. The engine will use a 64-bit hash of the file name when comparing file paths which means there's a small chance you might encounter a name collision. If this is an issue, you'll need to use a different name for one of the colliding file paths, or just not load from files and instead load from a data source. You can use `ma_sound_init_copy()` to initialize a copy of another sound. Note, however, that this only works for sounds that were initialized with `ma_sound_init_from_file()` and without the `MA_SOUND_FLAG_STREAM` flag. When you initialize a sound, if you specify a sound group the sound will be attached to that group automatically. If you set it to NULL, it will be automatically attached to the engine's endpoint. If you would instead rather leave the sound unattached by default, you can specify the `MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT` flag. This is useful if you want to set up a complex node graph. Sounds are not started by default. To start a sound, use `ma_sound_start()`. Stop a sound with `ma_sound_stop()`. Sounds can have their volume controlled with `ma_sound_set_volume()` in the same way as the engine's master volume. Sounds support stereo panning and pitching. Set the pan with `ma_sound_set_pan()`. Setting the pan to 0 will result in an unpanned sound. Setting it to -1 will shift everything to the left, whereas +1 will shift it to the right. The pitch can be controlled with `ma_sound_set_pitch()`. A larger value will result in a higher pitch. The pitch must be greater than 0. The engine supports 3D spatialization of sounds. By default sounds will have spatialization enabled, but if a sound does not need to be spatialized it's best to disable it. There are two ways to disable spatialization of a sound: ```c // Disable spatialization at initialization time via a flag: ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_NO_SPATIALIZATION, NULL, NULL, &sound); // Dynamically disable or enable spatialization post-initialization: ma_sound_set_spatialization_enabled(&sound, isSpatializationEnabled); ``` By default sounds will be spatialized based on the closest listener. If a sound should always be spatialized relative to a specific listener it can be pinned to one: ```c ma_sound_set_pinned_listener_index(&sound, listenerIndex); ``` Like listeners, sounds have a position. By default, the position of a sound is in absolute space, but it can be changed to be relative to a listener: ```c ma_sound_set_positioning(&sound, ma_positioning_relative); ``` Note that relative positioning of a sound only makes sense if there is either only one listener, or the sound is pinned to a specific listener. To set the position of a sound: ```c ma_sound_set_position(&sound, posX, posY, posZ); ``` The direction works the same way as a listener and represents the sound's forward direction: ```c ma_sound_set_direction(&sound, forwardX, forwardY, forwardZ); ``` Sound's also have a cone for controlling directional attenuation. This works exactly the same as listeners: ```c ma_sound_set_cone(&sound, innerAngleInRadians, outerAngleInRadians, outerGain); ``` The velocity of a sound is used for doppler effect and can be set as such: ```c ma_sound_set_velocity(&sound, velocityX, velocityY, velocityZ); ``` The engine supports different attenuation models which can be configured on a per-sound basis. By default the attenuation model is set to `ma_attenuation_model_inverse` which is the equivalent to OpenAL's `AL_INVERSE_DISTANCE_CLAMPED`. Configure the attenuation model like so: ```c ma_sound_set_attenuation_model(&sound, ma_attenuation_model_inverse); ``` The supported attenuation models include the following: +----------------------------------+----------------------------------------------+ | ma_attenuation_model_none | No distance attenuation. | +----------------------------------+----------------------------------------------+ | ma_attenuation_model_inverse | Equivalent to `AL_INVERSE_DISTANCE_CLAMPED`. | +----------------------------------+----------------------------------------------+ | ma_attenuation_model_linear | Linear attenuation. | +----------------------------------+----------------------------------------------+ | ma_attenuation_model_exponential | Exponential attenuation. | +----------------------------------+----------------------------------------------+ To control how quickly a sound rolls off as it moves away from the listener, you need to configure the rolloff: ```c ma_sound_set_rolloff(&sound, rolloff); ``` You can control the minimum and maximum gain to apply from spatialization: ```c ma_sound_set_min_gain(&sound, minGain); ma_sound_set_max_gain(&sound, maxGain); ``` Likewise, in the calculation of attenuation, you can control the minimum and maximum distances for the attenuation calculation. This is useful if you want to ensure sounds don't drop below a certain volume after the listener moves further away and to have sounds play a maximum volume when the listener is within a certain distance: ```c ma_sound_set_min_distance(&sound, minDistance); ma_sound_set_max_distance(&sound, maxDistance); ``` The engine's spatialization system supports doppler effect. The doppler factor can be configure on a per-sound basis like so: ```c ma_sound_set_doppler_factor(&sound, dopplerFactor); ``` You can fade sounds in and out with `ma_sound_set_fade_in_pcm_frames()` and `ma_sound_set_fade_in_milliseconds()`. Set the volume to -1 to use the current volume as the starting volume: ```c // Fade in over 1 second. ma_sound_set_fade_in_milliseconds(&sound, 0, 1, 1000); // ... sometime later ... // Fade out over 1 second, starting from the current volume. ma_sound_set_fade_in_milliseconds(&sound, -1, 0, 1000); ``` By default sounds will start immediately, but sometimes for timing and synchronization purposes it can be useful to schedule a sound to start or stop: ```c // Start the sound in 1 second from now. ma_sound_set_start_time_in_pcm_frames(&sound, ma_engine_get_time_in_pcm_frames(&engine) + (ma_engine_get_sample_rate(&engine) * 1)); // Stop the sound in 2 seconds from now. ma_sound_set_stop_time_in_pcm_frames(&sound, ma_engine_get_time_in_pcm_frames(&engine) + (ma_engine_get_sample_rate(&engine) * 2)); ``` Note that scheduling a start time still requires an explicit call to `ma_sound_start()` before anything will play. The time is specified in global time which is controlled by the engine. You can get the engine's current time with `ma_engine_get_time_in_pcm_frames()`. The engine's global time is incremented automatically as audio data is read, but it can be reset with `ma_engine_set_time_in_pcm_frames()` in case it needs to be resynchronized for some reason. To determine whether or not a sound is currently playing, use `ma_sound_is_playing()`. This will take the scheduled start and stop times into account. Whether or not a sound should loop can be controlled with `ma_sound_set_looping()`. Sounds will not be looping by default. Use `ma_sound_is_looping()` to determine whether or not a sound is looping. Use `ma_sound_at_end()` to determine whether or not a sound is currently at the end. For a looping sound this should never return true. Alternatively, you can configure a callback that will be fired when the sound reaches the end. Note that the callback is fired from the audio thread which means you cannot be uninitializing sound from the callback. To set the callback you can use `ma_sound_set_end_callback()`. Alternatively, if you're using `ma_sound_init_ex()`, you can pass it into the config like so: ```c soundConfig.endCallback = my_end_callback; soundConfig.pEndCallbackUserData = pMyEndCallbackUserData; ``` The end callback is declared like so: ```c void my_end_callback(void* pUserData, ma_sound* pSound) { ... } ``` Internally a sound wraps around a data source. Some APIs exist to control the underlying data source, mainly for convenience: ```c ma_sound_seek_to_pcm_frame(&sound, frameIndex); ma_sound_get_data_format(&sound, &format, &channels, &sampleRate, pChannelMap, channelMapCapacity); ma_sound_get_cursor_in_pcm_frames(&sound, &cursor); ma_sound_get_length_in_pcm_frames(&sound, &length); ``` Sound groups have the same API as sounds, only they are called `ma_sound_group`, and since they do not have any notion of a data source, anything relating to a data source is unavailable. Internally, sound data is loaded via the `ma_decoder` API which means by default it only supports file formats that have built-in support in miniaudio. You can extend this to support any kind of file format through the use of custom decoders. To do this you'll need to use a self-managed resource manager and configure it appropriately. See the "Resource Management" section below for details on how to set this up. 6. Resource Management ====================== Many programs will want to manage sound resources for things such as reference counting and streaming. This is supported by miniaudio via the `ma_resource_manager` API. The resource manager is mainly responsible for the following: * Loading of sound files into memory with reference counting. * Streaming of sound data. When loading a sound file, the resource manager will give you back a `ma_data_source` compatible object called `ma_resource_manager_data_source`. This object can be passed into any `ma_data_source` API which is how you can read and seek audio data. When loading a sound file, you specify whether or not you want the sound to be fully loaded into memory (and optionally pre-decoded) or streamed. When loading into memory, you can also specify whether or not you want the data to be loaded asynchronously. The example below is how you can initialize a resource manager using it's default configuration: ```c ma_resource_manager_config config; ma_resource_manager resourceManager; config = ma_resource_manager_config_init(); result = ma_resource_manager_init(&config, &resourceManager); if (result != MA_SUCCESS) { ma_device_uninit(&device); printf("Failed to initialize the resource manager."); return -1; } ``` You can configure the format, channels and sample rate of the decoded audio data. By default it will use the file's native data format, but you can configure it to use a consistent format. This is useful for offloading the cost of data conversion to load time rather than dynamically converting at mixing time. To do this, you configure the decoded format, channels and sample rate like the code below: ```c config = ma_resource_manager_config_init(); config.decodedFormat = device.playback.format; config.decodedChannels = device.playback.channels; config.decodedSampleRate = device.sampleRate; ``` In the code above, the resource manager will be configured so that any decoded audio data will be pre-converted at load time to the device's native data format. If instead you used defaults and the data format of the file did not match the device's data format, you would need to convert the data at mixing time which may be prohibitive in high-performance and large scale scenarios like games. Internally the resource manager uses the `ma_decoder` API to load sounds. This means by default it only supports decoders that are built into miniaudio. It's possible to support additional encoding formats through the use of custom decoders. To do so, pass in your `ma_decoding_backend_vtable` vtables into the resource manager config: ```c ma_decoding_backend_vtable* pCustomBackendVTables[] = { &g_ma_decoding_backend_vtable_libvorbis, &g_ma_decoding_backend_vtable_libopus }; ... resourceManagerConfig.ppCustomDecodingBackendVTables = pCustomBackendVTables; resourceManagerConfig.customDecodingBackendCount = sizeof(pCustomBackendVTables) / sizeof(pCustomBackendVTables[0]); resourceManagerConfig.pCustomDecodingBackendUserData = NULL; ``` This system can allow you to support any kind of file format. See the "Decoding" section for details on how to implement custom decoders. The miniaudio repository includes examples for Opus via libopus and libopusfile and Vorbis via libvorbis and libvorbisfile. Asynchronicity is achieved via a job system. When an operation needs to be performed, such as the decoding of a page, a job will be posted to a queue which will then be processed by a job thread. By default there will be only one job thread running, but this can be configured, like so: ```c config = ma_resource_manager_config_init(); config.jobThreadCount = MY_JOB_THREAD_COUNT; ``` By default job threads are managed internally by the resource manager, however you can also self manage your job threads if, for example, you want to integrate the job processing into your existing job infrastructure, or if you simply don't like the way the resource manager does it. To do this, just set the job thread count to 0 and process jobs manually. To process jobs, you first need to retrieve a job using `ma_resource_manager_next_job()` and then process it using `ma_job_process()`: ```c config = ma_resource_manager_config_init(); config.jobThreadCount = 0; // Don't manage any job threads internally. config.flags = MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING; // Optional. Makes `ma_resource_manager_next_job()` non-blocking. // ... Initialize your custom job threads ... void my_custom_job_thread(...) { for (;;) { ma_job job; ma_result result = ma_resource_manager_next_job(pMyResourceManager, &job); if (result != MA_SUCCESS) { if (result == MA_NO_DATA_AVAILABLE) { // No jobs are available. Keep going. Will only get this if the resource manager was initialized // with MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING. continue; } else if (result == MA_CANCELLED) { // MA_JOB_TYPE_QUIT was posted. Exit. break; } else { // Some other error occurred. break; } } ma_job_process(&job); } } ``` In the example above, the `MA_JOB_TYPE_QUIT` event is the used as the termination indicator, but you can use whatever you would like to terminate the thread. The call to `ma_resource_manager_next_job()` is blocking by default, but can be configured to be non-blocking by initializing the resource manager with the `MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING` configuration flag. Note that the `MA_JOB_TYPE_QUIT` will never be removed from the job queue. This is to give every thread the opportunity to catch the event and terminate naturally. When loading a file, it's sometimes convenient to be able to customize how files are opened and read instead of using standard `fopen()`, `fclose()`, etc. which is what miniaudio will use by default. This can be done by setting `pVFS` member of the resource manager's config: ```c // Initialize your custom VFS object. See documentation for VFS for information on how to do this. my_custom_vfs vfs = my_custom_vfs_init(); config = ma_resource_manager_config_init(); config.pVFS = &vfs; ``` This is particularly useful in programs like games where you want to read straight from an archive rather than the normal file system. If you do not specify a custom VFS, the resource manager will use the operating system's normal file operations. To load a sound file and create a data source, call `ma_resource_manager_data_source_init()`. When loading a sound you need to specify the file path and options for how the sounds should be loaded. By default a sound will be loaded synchronously. The returned data source is owned by the caller which means the caller is responsible for the allocation and freeing of the data source. Below is an example for initializing a data source: ```c ma_resource_manager_data_source dataSource; ma_result result = ma_resource_manager_data_source_init(pResourceManager, pFilePath, flags, &dataSource); if (result != MA_SUCCESS) { // Error. } // ... // A ma_resource_manager_data_source object is compatible with the `ma_data_source` API. To read data, just call // the `ma_data_source_read_pcm_frames()` like you would with any normal data source. result = ma_data_source_read_pcm_frames(&dataSource, pDecodedData, frameCount, &framesRead); if (result != MA_SUCCESS) { // Failed to read PCM frames. } // ... ma_resource_manager_data_source_uninit(&dataSource); ``` The `flags` parameter specifies how you want to perform loading of the sound file. It can be a combination of the following flags: ``` MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_LOOPING ``` When no flags are specified (set to 0), the sound will be fully loaded into memory, but not decoded, meaning the raw file data will be stored in memory, and then dynamically decoded when `ma_data_source_read_pcm_frames()` is called. To instead decode the audio data before storing it in memory, use the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE` flag. By default, the sound file will be loaded synchronously, meaning `ma_resource_manager_data_source_init()` will only return after the entire file has been loaded. This is good for simplicity, but can be prohibitively slow. You can instead load the sound asynchronously using the `MA_RESOURCE_MANAGER_DATA_SOURCE_ASYNC` flag. This will result in `ma_resource_manager_data_source_init()` returning quickly, but no data will be returned by `ma_data_source_read_pcm_frames()` until some data is available. When no data is available because the asynchronous decoding hasn't caught up, `MA_BUSY` will be returned by `ma_data_source_read_pcm_frames()`. For large sounds, it's often prohibitive to store the entire file in memory. To mitigate this, you can instead stream audio data which you can do by specifying the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag. When streaming, data will be decoded in 1 second pages. When a new page needs to be decoded, a job will be posted to the job queue and then subsequently processed in a job thread. The `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_LOOPING` flag can be used so that the sound will loop when it reaches the end by default. It's recommended you use this flag when you want to have a looping streaming sound. If you try loading a very short sound as a stream, you will get a glitch. This is because the resource manager needs to pre-fill the initial buffer at initialization time, and if you don't specify the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_LOOPING` flag, the resource manager will assume the sound is not looping and will stop filling the buffer when it reaches the end, therefore resulting in a discontinuous buffer. For in-memory sounds, reference counting is used to ensure the data is loaded only once. This means multiple calls to `ma_resource_manager_data_source_init()` with the same file path will result in the file data only being loaded once. Each call to `ma_resource_manager_data_source_init()` must be matched up with a call to `ma_resource_manager_data_source_uninit()`. Sometimes it can be useful for a program to register self-managed raw audio data and associate it with a file path. Use the `ma_resource_manager_register_*()` and `ma_resource_manager_unregister_*()` APIs to do this. `ma_resource_manager_register_decoded_data()` is used to associate a pointer to raw, self-managed decoded audio data in the specified data format with the specified name. Likewise, `ma_resource_manager_register_encoded_data()` is used to associate a pointer to raw self-managed encoded audio data (the raw file data) with the specified name. Note that these names need not be actual file paths. When `ma_resource_manager_data_source_init()` is called (without the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag), the resource manager will look for these explicitly registered data buffers and, if found, will use it as the backing data for the data source. Note that the resource manager does *not* make a copy of this data so it is up to the caller to ensure the pointer stays valid for its lifetime. Use `ma_resource_manager_unregister_data()` to unregister the self-managed data. You can also use `ma_resource_manager_register_file()` and `ma_resource_manager_unregister_file()` to register and unregister a file. It does not make sense to use the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag with a self-managed data pointer. 6.1. Asynchronous Loading and Synchronization --------------------------------------------- When loading asynchronously, it can be useful to poll whether or not loading has finished. Use `ma_resource_manager_data_source_result()` to determine this. For in-memory sounds, this will return `MA_SUCCESS` when the file has been *entirely* decoded. If the sound is still being decoded, `MA_BUSY` will be returned. Otherwise, some other error code will be returned if the sound failed to load. For streaming data sources, `MA_SUCCESS` will be returned when the first page has been decoded and the sound is ready to be played. If the first page is still being decoded, `MA_BUSY` will be returned. Otherwise, some other error code will be returned if the sound failed to load. In addition to polling, you can also use a simple synchronization object called a "fence" to wait for asynchronously loaded sounds to finish. This is called `ma_fence`. The advantage to using a fence is that it can be used to wait for a group of sounds to finish loading rather than waiting for sounds on an individual basis. There are two stages to loading a sound: * Initialization of the internal decoder; and * Completion of decoding of the file (the file is fully decoded) You can specify separate fences for each of the different stages. Waiting for the initialization of the internal decoder is important for when you need to know the sample format, channels and sample rate of the file. The example below shows how you could use a fence when loading a number of sounds: ```c // This fence will be released when all sounds are finished loading entirely. ma_fence fence; ma_fence_init(&fence); // This will be passed into the initialization routine for each sound. ma_resource_manager_pipeline_notifications notifications = ma_resource_manager_pipeline_notifications_init(); notifications.done.pFence = &fence; // Now load a bunch of sounds: for (iSound = 0; iSound < soundCount; iSound += 1) { ma_resource_manager_data_source_init(pResourceManager, pSoundFilePaths[iSound], flags, ¬ifications, &pSoundSources[iSound]); } // ... DO SOMETHING ELSE WHILE SOUNDS ARE LOADING ... // Wait for loading of sounds to finish. ma_fence_wait(&fence); ``` In the example above we used a fence for waiting until the entire file has been fully decoded. If you only need to wait for the initialization of the internal decoder to complete, you can use the `init` member of the `ma_resource_manager_pipeline_notifications` object: ```c notifications.init.pFence = &fence; ``` If a fence is not appropriate for your situation, you can instead use a callback that is fired on an individual sound basis. This is done in a very similar way to fences: ```c typedef struct { ma_async_notification_callbacks cb; void* pMyData; } my_notification; void my_notification_callback(ma_async_notification* pNotification) { my_notification* pMyNotification = (my_notification*)pNotification; // Do something in response to the sound finishing loading. } ... my_notification myCallback; myCallback.cb.onSignal = my_notification_callback; myCallback.pMyData = pMyData; ma_resource_manager_pipeline_notifications notifications = ma_resource_manager_pipeline_notifications_init(); notifications.done.pNotification = &myCallback; ma_resource_manager_data_source_init(pResourceManager, "my_sound.wav", flags, ¬ifications, &mySound); ``` In the example above we just extend the `ma_async_notification_callbacks` object and pass an instantiation into the `ma_resource_manager_pipeline_notifications` in the same way as we did with the fence, only we set `pNotification` instead of `pFence`. You can set both of these at the same time and they should both work as expected. If using the `pNotification` system, you need to ensure your `ma_async_notification_callbacks` object stays valid. 6.2. Resource Manager Implementation Details -------------------------------------------- Resources are managed in two main ways: * By storing the entire sound inside an in-memory buffer (referred to as a data buffer) * By streaming audio data on the fly (referred to as a data stream) A resource managed data source (`ma_resource_manager_data_source`) encapsulates a data buffer or data stream, depending on whether or not the data source was initialized with the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag. If so, it will make use of a `ma_resource_manager_data_stream` object. Otherwise it will use a `ma_resource_manager_data_buffer` object. Both of these objects are data sources which means they can be used with any `ma_data_source_*()` API. Another major feature of the resource manager is the ability to asynchronously decode audio files. This relieves the audio thread of time-consuming decoding which can negatively affect scalability due to the audio thread needing to complete it's work extremely quickly to avoid glitching. Asynchronous decoding is achieved through a job system. There is a central multi-producer, multi-consumer, fixed-capacity job queue. When some asynchronous work needs to be done, a job is posted to the queue which is then read by a job thread. The number of job threads can be configured for improved scalability, and job threads can all run in parallel without needing to worry about the order of execution (how this is achieved is explained below). When a sound is being loaded asynchronously, playback can begin before the sound has been fully decoded. This enables the application to start playback of the sound quickly, while at the same time allowing to resource manager to keep loading in the background. Since there may be less threads than the number of sounds being loaded at a given time, a simple scheduling system is used to keep decoding time balanced and fair. The resource manager solves this by splitting decoding into chunks called pages. By default, each page is 1 second long. When a page has been decoded, a new job will be posted to start decoding the next page. By dividing up decoding into pages, an individual sound shouldn't ever delay every other sound from having their first page decoded. Of course, when loading many sounds at the same time, there will always be an amount of time required to process jobs in the queue so in heavy load situations there will still be some delay. To determine if a data source is ready to have some frames read, use `ma_resource_manager_data_source_get_available_frames()`. This will return the number of frames available starting from the current position. 6.2.1. Job Queue ---------------- The resource manager uses a job queue which is multi-producer, multi-consumer, and fixed-capacity. This job queue is not currently lock-free, and instead uses a spinlock to achieve thread-safety. Only a fixed number of jobs can be allocated and inserted into the queue which is done through a lock-free data structure for allocating an index into a fixed sized array, with reference counting for mitigation of the ABA problem. The reference count is 32-bit. For many types of jobs it's important that they execute in a specific order. In these cases, jobs are executed serially. For the resource manager, serial execution of jobs is only required on a per-object basis (per data buffer or per data stream). Each of these objects stores an execution counter. When a job is posted it is associated with an execution counter. When the job is processed, it checks if the execution counter of the job equals the execution counter of the owning object and if so, processes the job. If the counters are not equal, the job will be posted back onto the job queue for later processing. When the job finishes processing the execution order of the main object is incremented. This system means the no matter how many job threads are executing, decoding of an individual sound will always get processed serially. The advantage to having multiple threads comes into play when loading multiple sounds at the same time. The resource manager's job queue is not 100% lock-free and will use a spinlock to achieve thread-safety for a very small section of code. This is only relevant when the resource manager uses more than one job thread. If only using a single job thread, which is the default, the lock should never actually wait in practice. The amount of time spent locking should be quite short, but it's something to be aware of for those who have pedantic lock-free requirements and need to use more than one job thread. There are plans to remove this lock in a future version. In addition, posting a job will release a semaphore, which on Win32 is implemented with `ReleaseSemaphore` and on POSIX platforms via a condition variable: ```c pthread_mutex_lock(&pSemaphore->lock); { pSemaphore->value += 1; pthread_cond_signal(&pSemaphore->cond); } pthread_mutex_unlock(&pSemaphore->lock); ``` Again, this is relevant for those with strict lock-free requirements in the audio thread. To avoid this, you can use non-blocking mode (via the `MA_JOB_QUEUE_FLAG_NON_BLOCKING` flag) and implement your own job processing routine (see the "Resource Manager" section above for details on how to do this). 6.2.2. Data Buffers ------------------- When the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag is excluded at initialization time, the resource manager will try to load the data into an in-memory data buffer. Before doing so, however, it will first check if the specified file is already loaded. If so, it will increment a reference counter and just use the already loaded data. This saves both time and memory. When the data buffer is uninitialized, the reference counter will be decremented. If the counter hits zero, the file will be unloaded. This is a detail to keep in mind because it could result in excessive loading and unloading of a sound. For example, the following sequence will result in a file be loaded twice, once after the other: ```c ma_resource_manager_data_source_init(pResourceManager, "my_file", ..., &myDataBuffer0); // Refcount = 1. Initial load. ma_resource_manager_data_source_uninit(&myDataBuffer0); // Refcount = 0. Unloaded. ma_resource_manager_data_source_init(pResourceManager, "my_file", ..., &myDataBuffer1); // Refcount = 1. Reloaded because previous uninit() unloaded it. ma_resource_manager_data_source_uninit(&myDataBuffer1); // Refcount = 0. Unloaded. ``` A binary search tree (BST) is used for storing data buffers as it has good balance between efficiency and simplicity. The key of the BST is a 64-bit hash of the file path that was passed into `ma_resource_manager_data_source_init()`. The advantage of using a hash is that it saves memory over storing the entire path, has faster comparisons, and results in a mostly balanced BST due to the random nature of the hash. The disadvantages are that file names are case-sensitive and there's a small chance of name collisions. If case-sensitivity is an issue, you should normalize your file names to upper- or lower-case before initializing your data sources. If name collisions become an issue, you'll need to change the name of one of the colliding names or just not use the resource manager. When a sound file has not already been loaded and the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC` flag is excluded, the file will be decoded synchronously by the calling thread. There are two options for controlling how the audio is stored in the data buffer - encoded or decoded. When the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE` option is excluded, the raw file data will be stored in memory. Otherwise the sound will be decoded before storing it in memory. Synchronous loading is a very simple and standard process of simply adding an item to the BST, allocating a block of memory and then decoding (if `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE` is specified). When the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC` flag is specified, loading of the data buffer is done asynchronously. In this case, a job is posted to the queue to start loading and then the function immediately returns, setting an internal result code to `MA_BUSY`. This result code is returned when the program calls `ma_resource_manager_data_source_result()`. When decoding has fully completed `MA_SUCCESS` will be returned. This can be used to know if loading has fully completed. When loading asynchronously, a single job is posted to the queue of the type `MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER_NODE`. This involves making a copy of the file path and associating it with job. When the job is processed by the job thread, it will first load the file using the VFS associated with the resource manager. When using a custom VFS, it's important that it be completely thread-safe because it will be used from one or more job threads at the same time. Individual files should only ever be accessed by one thread at a time, however. After opening the file via the VFS, the job will determine whether or not the file is being decoded. If not, it simply allocates a block of memory and loads the raw file contents into it and returns. On the other hand, when the file is being decoded, it will first allocate a decoder on the heap and initialize it. Then it will check if the length of the file is known. If so it will allocate a block of memory to store the decoded output and initialize it to silence. If the size is unknown, it will allocate room for one page. After memory has been allocated, the first page will be decoded. If the sound is shorter than a page, the result code will be set to `MA_SUCCESS` and the completion event will be signalled and loading is now complete. If, however, there is more to decode, a job with the code `MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE` is posted. This job will decode the next page and perform the same process if it reaches the end. If there is more to decode, the job will post another `MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE` job which will keep on happening until the sound has been fully decoded. For sounds of an unknown length, each page will be linked together as a linked list. Internally this is implemented via the `ma_paged_audio_buffer` object. 6.2.3. Data Streams ------------------- Data streams only ever store two pages worth of data for each instance. They are most useful for large sounds like music tracks in games that would consume too much memory if fully decoded in memory. After every frame from a page has been read, a job will be posted to load the next page which is done from the VFS. For data streams, the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC` flag will determine whether or not initialization of the data source waits until the two pages have been decoded. When unset, `ma_resource_manager_data_source_init()` will wait until the two pages have been loaded, otherwise it will return immediately. When frames are read from a data stream using `ma_resource_manager_data_source_read_pcm_frames()`, `MA_BUSY` will be returned if there are no frames available. If there are some frames available, but less than the number requested, `MA_SUCCESS` will be returned, but the actual number of frames read will be less than the number requested. Due to the asynchronous nature of data streams, seeking is also asynchronous. If the data stream is in the middle of a seek, `MA_BUSY` will be returned when trying to read frames. When `ma_resource_manager_data_source_read_pcm_frames()` results in a page getting fully consumed a job is posted to load the next page. This will be posted from the same thread that called `ma_resource_manager_data_source_read_pcm_frames()`. Data streams are uninitialized by posting a job to the queue, but the function won't return until that job has been processed. The reason for this is that the caller owns the data stream object and therefore miniaudio needs to ensure everything completes before handing back control to the caller. Also, if the data stream is uninitialized while pages are in the middle of decoding, they must complete before destroying any underlying object and the job system handles this cleanly. Note that when a new page needs to be loaded, a job will be posted to the resource manager's job thread from the audio thread. You must keep in mind the details mentioned in the "Job Queue" section above regarding locking when posting an event if you require a strictly lock-free audio thread. 7. Node Graph ============= miniaudio's routing infrastructure follows a node graph paradigm. The idea is that you create a node whose outputs are attached to inputs of another node, thereby creating a graph. There are different types of nodes, with each node in the graph processing input data to produce output, which is then fed through the chain. Each node in the graph can apply their own custom effects. At the start of the graph will usually be one or more data source nodes which have no inputs and instead pull their data from a data source. At the end of the graph is an endpoint which represents the end of the chain and is where the final output is ultimately extracted from. Each node has a number of input buses and a number of output buses. An output bus from a node is attached to an input bus of another. Multiple nodes can connect their output buses to another node's input bus, in which case their outputs will be mixed before processing by the node. Below is a diagram that illustrates a hypothetical node graph setup: ``` >>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Data flows left to right >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> +---------------+ +-----------------+ | Data Source 1 =----+ +----------+ +----= Low Pass Filter =----+ +---------------+ | | =----+ +-----------------+ | +----------+ +----= Splitter | +----= ENDPOINT | +---------------+ | | =----+ +-----------------+ | +----------+ | Data Source 2 =----+ +----------+ +----= Echo / Delay =----+ +---------------+ +-----------------+ ``` In the above graph, it starts with two data sources whose outputs are attached to the input of a splitter node. It's at this point that the two data sources are mixed. After mixing, the splitter performs it's processing routine and produces two outputs which is simply a duplication of the input stream. One output is attached to a low pass filter, whereas the other output is attached to a echo/delay. The outputs of the low pass filter and the echo are attached to the endpoint, and since they're both connected to the same input bus, they'll be mixed. Each input bus must be configured to accept the same number of channels, but the number of channels used by input buses can be different to the number of channels for output buses in which case miniaudio will automatically convert the input data to the output channel count before processing. The number of channels of an output bus of one node must match the channel count of the input bus it's attached to. The channel counts cannot be changed after the node has been initialized. If you attempt to attach an output bus to an input bus with a different channel count, attachment will fail. To use a node graph, you first need to initialize a `ma_node_graph` object. This is essentially a container around the entire graph. The `ma_node_graph` object is required for some thread-safety issues which will be explained later. A `ma_node_graph` object is initialized using miniaudio's standard config/init system: ```c ma_node_graph_config nodeGraphConfig = ma_node_graph_config_init(myChannelCount); result = ma_node_graph_init(&nodeGraphConfig, NULL, &nodeGraph); // Second parameter is a pointer to allocation callbacks. if (result != MA_SUCCESS) { // Failed to initialize node graph. } ``` When you initialize the node graph, you're specifying the channel count of the endpoint. The endpoint is a special node which has one input bus and one output bus, both of which have the same channel count, which is specified in the config. Any nodes that connect directly to the endpoint must be configured such that their output buses have the same channel count. When you read audio data from the node graph, it'll have the channel count you specified in the config. To read data from the graph: ```c ma_uint32 framesRead; result = ma_node_graph_read_pcm_frames(&nodeGraph, pFramesOut, frameCount, &framesRead); if (result != MA_SUCCESS) { // Failed to read data from the node graph. } ``` When you read audio data, miniaudio starts at the node graph's endpoint node which then pulls in data from its input attachments, which in turn recursively pull in data from their inputs, and so on. At the start of the graph there will be some kind of data source node which will have zero inputs and will instead read directly from a data source. The base nodes don't literally need to read from a `ma_data_source` object, but they will always have some kind of underlying object that sources some kind of audio. The `ma_data_source_node` node can be used to read from a `ma_data_source`. Data is always in floating-point format and in the number of channels you specified when the graph was initialized. The sample rate is defined by the underlying data sources. It's up to you to ensure they use a consistent and appropriate sample rate. The `ma_node` API is designed to allow custom nodes to be implemented with relative ease, but miniaudio includes a few stock nodes for common functionality. This is how you would initialize a node which reads directly from a data source (`ma_data_source_node`) which is an example of one of the stock nodes that comes with miniaudio: ```c ma_data_source_node_config config = ma_data_source_node_config_init(pMyDataSource); ma_data_source_node dataSourceNode; result = ma_data_source_node_init(&nodeGraph, &config, NULL, &dataSourceNode); if (result != MA_SUCCESS) { // Failed to create data source node. } ``` The data source node will use the output channel count to determine the channel count of the output bus. There will be 1 output bus and 0 input buses (data will be drawn directly from the data source). The data source must output to floating-point (`ma_format_f32`) or else an error will be returned from `ma_data_source_node_init()`. By default the node will not be attached to the graph. To do so, use `ma_node_attach_output_bus()`: ```c result = ma_node_attach_output_bus(&dataSourceNode, 0, ma_node_graph_get_endpoint(&nodeGraph), 0); if (result != MA_SUCCESS) { // Failed to attach node. } ``` The code above connects the data source node directly to the endpoint. Since the data source node has only a single output bus, the index will always be 0. Likewise, the endpoint only has a single input bus which means the input bus index will also always be 0. To detach a specific output bus, use `ma_node_detach_output_bus()`. To detach all output buses, use `ma_node_detach_all_output_buses()`. If you want to just move the output bus from one attachment to another, you do not need to detach first. You can just call `ma_node_attach_output_bus()` and it'll deal with it for you. Less frequently you may want to create a specialized node. This will be a node where you implement your own processing callback to apply a custom effect of some kind. This is similar to initializing one of the stock node types, only this time you need to specify a pointer to a vtable containing a pointer to the processing function and the number of input and output buses. Example: ```c static void my_custom_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut) { // Do some processing of ppFramesIn (one stream of audio data per input bus) const float* pFramesIn_0 = ppFramesIn[0]; // Input bus @ index 0. const float* pFramesIn_1 = ppFramesIn[1]; // Input bus @ index 1. float* pFramesOut_0 = ppFramesOut[0]; // Output bus @ index 0. // Do some processing. On input, `pFrameCountIn` will be the number of input frames in each // buffer in `ppFramesIn` and `pFrameCountOut` will be the capacity of each of the buffers // in `ppFramesOut`. On output, `pFrameCountIn` should be set to the number of input frames // your node consumed and `pFrameCountOut` should be set the number of output frames that // were produced. // // You should process as many frames as you can. If your effect consumes input frames at the // same rate as output frames (always the case, unless you're doing resampling), you need // only look at `ppFramesOut` and process that exact number of frames. If you're doing // resampling, you'll need to be sure to set both `pFrameCountIn` and `pFrameCountOut` // properly. } static ma_node_vtable my_custom_node_vtable = { my_custom_node_process_pcm_frames, // The function that will be called to process your custom node. This is where you'd implement your effect processing. NULL, // Optional. A callback for calculating the number of input frames that are required to process a specified number of output frames. 2, // 2 input buses. 1, // 1 output bus. 0 // Default flags. }; ... // Each bus needs to have a channel count specified. To do this you need to specify the channel // counts in an array and then pass that into the node config. ma_uint32 inputChannels[2]; // Equal in size to the number of input channels specified in the vtable. ma_uint32 outputChannels[1]; // Equal in size to the number of output channels specified in the vtable. inputChannels[0] = channelsIn; inputChannels[1] = channelsIn; outputChannels[0] = channelsOut; ma_node_config nodeConfig = ma_node_config_init(); nodeConfig.vtable = &my_custom_node_vtable; nodeConfig.pInputChannels = inputChannels; nodeConfig.pOutputChannels = outputChannels; ma_node_base node; result = ma_node_init(&nodeGraph, &nodeConfig, NULL, &node); if (result != MA_SUCCESS) { // Failed to initialize node. } ``` When initializing a custom node, as in the code above, you'll normally just place your vtable in static space. The number of input and output buses are specified as part of the vtable. If you need a variable number of buses on a per-node bases, the vtable should have the relevant bus count set to `MA_NODE_BUS_COUNT_UNKNOWN`. In this case, the bus count should be set in the node config: ```c static ma_node_vtable my_custom_node_vtable = { my_custom_node_process_pcm_frames, // The function that will be called process your custom node. This is where you'd implement your effect processing. NULL, // Optional. A callback for calculating the number of input frames that are required to process a specified number of output frames. MA_NODE_BUS_COUNT_UNKNOWN, // The number of input buses is determined on a per-node basis. 1, // 1 output bus. 0 // Default flags. }; ... ma_node_config nodeConfig = ma_node_config_init(); nodeConfig.vtable = &my_custom_node_vtable; nodeConfig.inputBusCount = myBusCount; // <-- Since the vtable specifies MA_NODE_BUS_COUNT_UNKNOWN, the input bus count should be set here. nodeConfig.pInputChannels = inputChannels; // <-- Make sure there are nodeConfig.inputBusCount elements in this array. nodeConfig.pOutputChannels = outputChannels; // <-- The vtable specifies 1 output bus, so there must be 1 element in this array. ``` In the above example it's important to never set the `inputBusCount` and `outputBusCount` members to anything other than their defaults if the vtable specifies an explicit count. They can only be set if the vtable specifies MA_NODE_BUS_COUNT_UNKNOWN in the relevant bus count. Most often you'll want to create a structure to encapsulate your node with some extra data. You need to make sure the `ma_node_base` object is your first member of the structure: ```c typedef struct { ma_node_base base; // <-- Make sure this is always the first member. float someCustomData; } my_custom_node; ``` By doing this, your object will be compatible with all `ma_node` APIs and you can attach it to the graph just like any other node. In the custom processing callback (`my_custom_node_process_pcm_frames()` in the example above), the number of channels for each bus is what was specified by the config when the node was initialized with `ma_node_init()`. In addition, all attachments to each of the input buses will have been pre-mixed by miniaudio. The config allows you to specify different channel counts for each individual input and output bus. It's up to the effect to handle it appropriate, and if it can't, return an error in it's initialization routine. Custom nodes can be assigned some flags to describe their behaviour. These are set via the vtable and include the following: +-----------------------------------------+---------------------------------------------------+ | Flag Name | Description | +-----------------------------------------+---------------------------------------------------+ | MA_NODE_FLAG_PASSTHROUGH | Useful for nodes that do not do any kind of audio | | | processing, but are instead used for tracking | | | time, handling events, etc. Also used by the | | | internal endpoint node. It reads directly from | | | the input bus to the output bus. Nodes with this | | | flag must have exactly 1 input bus and 1 output | | | bus, and both buses must have the same channel | | | counts. | +-----------------------------------------+---------------------------------------------------+ | MA_NODE_FLAG_CONTINUOUS_PROCESSING | Causes the processing callback to be called even | | | when no data is available to be read from input | | | attachments. When a node has at least one input | | | bus, but there are no inputs attached or the | | | inputs do not deliver any data, the node's | | | processing callback will not get fired. This flag | | | will make it so the callback is always fired | | | regardless of whether or not any input data is | | | received. This is useful for effects like | | | echos where there will be a tail of audio data | | | that still needs to be processed even when the | | | original data sources have reached their ends. It | | | may also be useful for nodes that must always | | | have their processing callback fired when there | | | are no inputs attached. | +-----------------------------------------+---------------------------------------------------+ | MA_NODE_FLAG_ALLOW_NULL_INPUT | Used in conjunction with | | | `MA_NODE_FLAG_CONTINUOUS_PROCESSING`. When this | | | is set, the `ppFramesIn` parameter of the | | | processing callback will be set to NULL when | | | there are no input frames are available. When | | | this is unset, silence will be posted to the | | | processing callback. | +-----------------------------------------+---------------------------------------------------+ | MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES | Used to tell miniaudio that input and output | | | frames are processed at different rates. You | | | should set this for any nodes that perform | | | resampling. | +-----------------------------------------+---------------------------------------------------+ | MA_NODE_FLAG_SILENT_OUTPUT | Used to tell miniaudio that a node produces only | | | silent output. This is useful for nodes where you | | | don't want the output to contribute to the final | | | mix. An example might be if you want split your | | | stream and have one branch be output to a file. | | | When using this flag, you should avoid writing to | | | the output buffer of the node's processing | | | callback because miniaudio will ignore it anyway. | +-----------------------------------------+---------------------------------------------------+ If you need to make a copy of an audio stream for effect processing you can use a splitter node called `ma_splitter_node`. This takes has 1 input bus and splits the stream into 2 output buses. You can use it like this: ```c ma_splitter_node_config splitterNodeConfig = ma_splitter_node_config_init(channels); ma_splitter_node splitterNode; result = ma_splitter_node_init(&nodeGraph, &splitterNodeConfig, NULL, &splitterNode); if (result != MA_SUCCESS) { // Failed to create node. } // Attach your output buses to two different input buses (can be on two different nodes). ma_node_attach_output_bus(&splitterNode, 0, ma_node_graph_get_endpoint(&nodeGraph), 0); // Attach directly to the endpoint. ma_node_attach_output_bus(&splitterNode, 1, &myEffectNode, 0); // Attach to input bus 0 of some effect node. ``` The volume of an output bus can be configured on a per-bus basis: ```c ma_node_set_output_bus_volume(&splitterNode, 0, 0.5f); ma_node_set_output_bus_volume(&splitterNode, 1, 0.5f); ``` In the code above we're using the splitter node from before and changing the volume of each of the copied streams. You can start and stop a node with the following: ```c ma_node_set_state(&splitterNode, ma_node_state_started); // The default state. ma_node_set_state(&splitterNode, ma_node_state_stopped); ``` By default the node is in a started state, but since it won't be connected to anything won't actually be invoked by the node graph until it's connected. When you stop a node, data will not be read from any of its input connections. You can use this property to stop a group of sounds atomically. You can configure the initial state of a node in it's config: ```c nodeConfig.initialState = ma_node_state_stopped; ``` Note that for the stock specialized nodes, all of their configs will have a `nodeConfig` member which is the config to use with the base node. This is where the initial state can be configured for specialized nodes: ```c dataSourceNodeConfig.nodeConfig.initialState = ma_node_state_stopped; ``` When using a specialized node like `ma_data_source_node` or `ma_splitter_node`, be sure to not modify the `vtable` member of the `nodeConfig` object. 7.1. Timing ----------- The node graph supports starting and stopping nodes at scheduled times. This is especially useful for data source nodes where you want to get the node set up, but only start playback at a specific time. There are two clocks: local and global. A local clock is per-node, whereas the global clock is per graph. Scheduling starts and stops can only be done based on the global clock because the local clock will not be running while the node is stopped. The global clocks advances whenever `ma_node_graph_read_pcm_frames()` is called. On the other hand, the local clock only advances when the node's processing callback is fired, and is advanced based on the output frame count. To retrieve the global time, use `ma_node_graph_get_time()`. The global time can be set with `ma_node_graph_set_time()` which might be useful if you want to do seeking on a global timeline. Getting and setting the local time is similar. Use `ma_node_get_time()` to retrieve the local time, and `ma_node_set_time()` to set the local time. The global and local times will be advanced by the audio thread, so care should be taken to avoid data races. Ideally you should avoid calling these outside of the node processing callbacks which are always run on the audio thread. There is basic support for scheduling the starting and stopping of nodes. You can only schedule one start and one stop at a time. This is mainly intended for putting nodes into a started or stopped state in a frame-exact manner. Without this mechanism, starting and stopping of a node is limited to the resolution of a call to `ma_node_graph_read_pcm_frames()` which would typically be in blocks of several milliseconds. The following APIs can be used for scheduling node states: ```c ma_node_set_state_time() ma_node_get_state_time() ``` The time is absolute and must be based on the global clock. An example is below: ```c ma_node_set_state_time(&myNode, ma_node_state_started, sampleRate*1); // Delay starting to 1 second. ma_node_set_state_time(&myNode, ma_node_state_stopped, sampleRate*5); // Delay stopping to 5 seconds. ``` An example for changing the state using a relative time. ```c ma_node_set_state_time(&myNode, ma_node_state_started, sampleRate*1 + ma_node_graph_get_time(&myNodeGraph)); ma_node_set_state_time(&myNode, ma_node_state_stopped, sampleRate*5 + ma_node_graph_get_time(&myNodeGraph)); ``` Note that due to the nature of multi-threading the times may not be 100% exact. If this is an issue, consider scheduling state changes from within a processing callback. An idea might be to have some kind of passthrough trigger node that is used specifically for tracking time and handling events. 7.2. Thread Safety and Locking ------------------------------ When processing audio, it's ideal not to have any kind of locking in the audio thread. Since it's expected that `ma_node_graph_read_pcm_frames()` would be run on the audio thread, it does so without the use of any locks. This section discusses the implementation used by miniaudio and goes over some of the compromises employed by miniaudio to achieve this goal. Note that the current implementation may not be ideal - feedback and critiques are most welcome. The node graph API is not *entirely* lock-free. Only `ma_node_graph_read_pcm_frames()` is expected to be lock-free. Attachment, detachment and uninitialization of nodes use locks to simplify the implementation, but are crafted in a way such that such locking is not required when reading audio data from the graph. Locking in these areas are achieved by means of spinlocks. The main complication with keeping `ma_node_graph_read_pcm_frames()` lock-free stems from the fact that a node can be uninitialized, and it's memory potentially freed, while in the middle of being processed on the audio thread. There are times when the audio thread will be referencing a node, which means the uninitialization process of a node needs to make sure it delays returning until the audio thread is finished so that control is not handed back to the caller thereby giving them a chance to free the node's memory. When the audio thread is processing a node, it does so by reading from each of the output buses of the node. In order for a node to process data for one of its output buses, it needs to read from each of its input buses, and so on an so forth. It follows that once all output buses of a node are detached, the node as a whole will be disconnected and no further processing will occur unless it's output buses are reattached, which won't be happening when the node is being uninitialized. By having `ma_node_detach_output_bus()` wait until the audio thread is finished with it, we can simplify a few things, at the expense of making `ma_node_detach_output_bus()` a bit slower. By doing this, the implementation of `ma_node_uninit()` becomes trivial - just detach all output nodes, followed by each of the attachments to each of its input nodes, and then do any final clean up. With the above design, the worst-case scenario is `ma_node_detach_output_bus()` taking as long as it takes to process the output bus being detached. This will happen if it's called at just the wrong moment where the audio thread has just iterated it and has just started processing. The caller of `ma_node_detach_output_bus()` will stall until the audio thread is finished, which includes the cost of recursively processing its inputs. This is the biggest compromise made with the approach taken by miniaudio for its lock-free processing system. The cost of detaching nodes earlier in the pipeline (data sources, for example) will be cheaper than the cost of detaching higher level nodes, such as some kind of final post-processing endpoint. If you need to do mass detachments, detach starting from the lowest level nodes and work your way towards the final endpoint node (but don't try detaching the node graph's endpoint). If the audio thread is not running, detachment will be fast and detachment in any order will be the same. The reason nodes need to wait for their input attachments to complete is due to the potential for desyncs between data sources. If the node was to terminate processing mid way through processing its inputs, there's a chance that some of the underlying data sources will have been read, but then others not. That will then result in a potential desynchronization when detaching and reattaching higher-level nodes. A possible solution to this is to have an option when detaching to terminate processing before processing all input attachments which should be fairly simple. Another compromise, albeit less significant, is locking when attaching and detaching nodes. This locking is achieved by means of a spinlock in order to reduce memory overhead. A lock is present for each input bus and output bus. When an output bus is connected to an input bus, both the output bus and input bus is locked. This locking is specifically for attaching and detaching across different threads and does not affect `ma_node_graph_read_pcm_frames()` in any way. The locking and unlocking is mostly self-explanatory, but a slightly less intuitive aspect comes into it when considering that iterating over attachments must not break as a result of attaching or detaching a node while iteration is occurring. Attaching and detaching are both quite simple. When an output bus of a node is attached to an input bus of another node, it's added to a linked list. Basically, an input bus is a linked list, where each item in the list is and output bus. We have some intentional (and convenient) restrictions on what can done with the linked list in order to simplify the implementation. First of all, whenever something needs to iterate over the list, it must do so in a forward direction. Backwards iteration is not supported. Also, items can only be added to the start of the list. The linked list is a doubly-linked list where each item in the list (an output bus) holds a pointer to the next item in the list, and another to the previous item. A pointer to the previous item is only required for fast detachment of the node - it is never used in iteration. This is an important property because it means from the perspective of iteration, attaching and detaching of an item can be done with a single atomic assignment. This is exploited by both the attachment and detachment process. When attaching the node, the first thing that is done is the setting of the local "next" and "previous" pointers of the node. After that, the item is "attached" to the list by simply performing an atomic exchange with the head pointer. After that, the node is "attached" to the list from the perspective of iteration. Even though the "previous" pointer of the next item hasn't yet been set, from the perspective of iteration it's been attached because iteration will only be happening in a forward direction which means the "previous" pointer won't actually ever get used. The same general process applies to detachment. See `ma_node_attach_output_bus()` and `ma_node_detach_output_bus()` for the implementation of this mechanism. 8. Decoding =========== The `ma_decoder` API is used for reading audio files. Decoders are completely decoupled from devices and can be used independently. Built-in support is included for the following formats: +---------+ | Format | +---------+ | WAV | | MP3 | | FLAC | +---------+ You can disable the built-in decoders by specifying one or more of the following options before the miniaudio implementation: ```c #define MA_NO_WAV #define MA_NO_MP3 #define MA_NO_FLAC ``` miniaudio supports the ability to plug in custom decoders. See the section below for details on how to use custom decoders. A decoder can be initialized from a file with `ma_decoder_init_file()`, a block of memory with `ma_decoder_init_memory()`, or from data delivered via callbacks with `ma_decoder_init()`. Here is an example for loading a decoder from a file: ```c ma_decoder decoder; ma_result result = ma_decoder_init_file("MySong.mp3", NULL, &decoder); if (result != MA_SUCCESS) { return false; // An error occurred. } ... ma_decoder_uninit(&decoder); ``` When initializing a decoder, you can optionally pass in a pointer to a `ma_decoder_config` object (the `NULL` argument in the example above) which allows you to configure the output format, channel count, sample rate and channel map: ```c ma_decoder_config config = ma_decoder_config_init(ma_format_f32, 2, 48000); ``` When passing in `NULL` for decoder config in `ma_decoder_init*()`, the output format will be the same as that defined by the decoding backend. Data is read from the decoder as PCM frames. This will output the number of PCM frames actually read. If this is less than the requested number of PCM frames it means you've reached the end. The return value will be `MA_AT_END` if no samples have been read and the end has been reached. ```c ma_result result = ma_decoder_read_pcm_frames(pDecoder, pFrames, framesToRead, &framesRead); if (framesRead < framesToRead) { // Reached the end. } ``` You can also seek to a specific frame like so: ```c ma_result result = ma_decoder_seek_to_pcm_frame(pDecoder, targetFrame); if (result != MA_SUCCESS) { return false; // An error occurred. } ``` If you want to loop back to the start, you can simply seek back to the first PCM frame: ```c ma_decoder_seek_to_pcm_frame(pDecoder, 0); ``` When loading a decoder, miniaudio uses a trial and error technique to find the appropriate decoding backend. This can be unnecessarily inefficient if the type is already known. In this case you can use `encodingFormat` variable in the device config to specify a specific encoding format you want to decode: ```c decoderConfig.encodingFormat = ma_encoding_format_wav; ``` See the `ma_encoding_format` enum for possible encoding formats. The `ma_decoder_init_file()` API will try using the file extension to determine which decoding backend to prefer. 8.1. Custom Decoders -------------------- It's possible to implement a custom decoder and plug it into miniaudio. This is extremely useful when you want to use the `ma_decoder` API, but need to support an encoding format that's not one of the stock formats supported by miniaudio. This can be put to particularly good use when using the `ma_engine` and/or `ma_resource_manager` APIs because they use `ma_decoder` internally. If, for example, you wanted to support Opus, you can do so with a custom decoder (there if a reference Opus decoder in the "extras" folder of the miniaudio repository which uses libopus + libopusfile). A custom decoder must implement a data source. A vtable called `ma_decoding_backend_vtable` needs to be implemented which is then passed into the decoder config: ```c ma_decoding_backend_vtable* pCustomBackendVTables[] = { &g_ma_decoding_backend_vtable_libvorbis, &g_ma_decoding_backend_vtable_libopus }; ... decoderConfig = ma_decoder_config_init_default(); decoderConfig.pCustomBackendUserData = NULL; decoderConfig.ppCustomBackendVTables = pCustomBackendVTables; decoderConfig.customBackendCount = sizeof(pCustomBackendVTables) / sizeof(pCustomBackendVTables[0]); ``` The `ma_decoding_backend_vtable` vtable has the following functions: ``` onInit onInitFile onInitFileW onInitMemory onUninit ``` There are only two functions that must be implemented - `onInit` and `onUninit`. The other functions can be implemented for a small optimization for loading from a file path or memory. If these are not specified, miniaudio will deal with it for you via a generic implementation. When you initialize a custom data source (by implementing the `onInit` function in the vtable) you will need to output a pointer to a `ma_data_source` which implements your custom decoder. See the section about data sources for details on how to implement this. Alternatively, see the "custom_decoders" example in the miniaudio repository. The `onInit` function takes a pointer to some callbacks for the purpose of reading raw audio data from some arbitrary source. You'll use these functions to read from the raw data and perform the decoding. When you call them, you will pass in the `pReadSeekTellUserData` pointer to the relevant parameter. The `pConfig` parameter in `onInit` can be used to configure the backend if appropriate. It's only used as a hint and can be ignored. However, if any of the properties are relevant to your decoder, an optimal implementation will handle the relevant properties appropriately. If memory allocation is required, it should be done so via the specified allocation callbacks if possible (the `pAllocationCallbacks` parameter). If an error occurs when initializing the decoder, you should leave `ppBackend` unset, or set to NULL, and make sure everything is cleaned up appropriately and an appropriate result code returned. When multiple custom backends are specified, miniaudio will cycle through the vtables in the order they're listed in the array that's passed into the decoder config so it's important that your initialization routine is clean. When a decoder is uninitialized, the `onUninit` callback will be fired which will give you an opportunity to clean up and internal data. 9. Encoding =========== The `ma_encoding` API is used for writing audio files. The only supported output format is WAV. This can be disabled by specifying the following option before the implementation of miniaudio: ```c #define MA_NO_WAV ``` An encoder can be initialized to write to a file with `ma_encoder_init_file()` or from data delivered via callbacks with `ma_encoder_init()`. Below is an example for initializing an encoder to output to a file. ```c ma_encoder_config config = ma_encoder_config_init(ma_encoding_format_wav, FORMAT, CHANNELS, SAMPLE_RATE); ma_encoder encoder; ma_result result = ma_encoder_init_file("my_file.wav", &config, &encoder); if (result != MA_SUCCESS) { // Error } ... ma_encoder_uninit(&encoder); ``` When initializing an encoder you must specify a config which is initialized with `ma_encoder_config_init()`. Here you must specify the file type, the output sample format, output channel count and output sample rate. The following file types are supported: +------------------------+-------------+ | Enum | Description | +------------------------+-------------+ | ma_encoding_format_wav | WAV | +------------------------+-------------+ If the format, channel count or sample rate is not supported by the output file type an error will be returned. The encoder will not perform data conversion so you will need to convert it before outputting any audio data. To output audio data, use `ma_encoder_write_pcm_frames()`, like in the example below: ```c ma_uint64 framesWritten; result = ma_encoder_write_pcm_frames(&encoder, pPCMFramesToWrite, framesToWrite, &framesWritten); if (result != MA_SUCCESS) { ... handle error ... } ``` The `framesWritten` variable will contain the number of PCM frames that were actually written. This is optionally and you can pass in `NULL` if you need this. Encoders must be uninitialized with `ma_encoder_uninit()`. 10. Data Conversion =================== A data conversion API is included with miniaudio which supports the majority of data conversion requirements. This supports conversion between sample formats, channel counts (with channel mapping) and sample rates. 10.1. Sample Format Conversion ------------------------------ Conversion between sample formats is achieved with the `ma_pcm_*_to_*()`, `ma_pcm_convert()` and `ma_convert_pcm_frames_format()` APIs. Use `ma_pcm_*_to_*()` to convert between two specific formats. Use `ma_pcm_convert()` to convert based on a `ma_format` variable. Use `ma_convert_pcm_frames_format()` to convert PCM frames where you want to specify the frame count and channel count as a variable instead of the total sample count. 10.1.1. Dithering ----------------- Dithering can be set using the ditherMode parameter. The different dithering modes include the following, in order of efficiency: +-----------+--------------------------+ | Type | Enum Token | +-----------+--------------------------+ | None | ma_dither_mode_none | | Rectangle | ma_dither_mode_rectangle | | Triangle | ma_dither_mode_triangle | +-----------+--------------------------+ Note that even if the dither mode is set to something other than `ma_dither_mode_none`, it will be ignored for conversions where dithering is not needed. Dithering is available for the following conversions: ``` s16 -> u8 s24 -> u8 s32 -> u8 f32 -> u8 s24 -> s16 s32 -> s16 f32 -> s16 ``` Note that it is not an error to pass something other than ma_dither_mode_none for conversions where dither is not used. It will just be ignored. 10.2. Channel Conversion ------------------------ Channel conversion is used for channel rearrangement and conversion from one channel count to another. The `ma_channel_converter` API is used for channel conversion. Below is an example of initializing a simple channel converter which converts from mono to stereo. ```c ma_channel_converter_config config = ma_channel_converter_config_init( ma_format, // Sample format 1, // Input channels NULL, // Input channel map 2, // Output channels NULL, // Output channel map ma_channel_mix_mode_default); // The mixing algorithm to use when combining channels. result = ma_channel_converter_init(&config, NULL, &converter); if (result != MA_SUCCESS) { // Error. } ``` To perform the conversion simply call `ma_channel_converter_process_pcm_frames()` like so: ```c ma_result result = ma_channel_converter_process_pcm_frames(&converter, pFramesOut, pFramesIn, frameCount); if (result != MA_SUCCESS) { // Error. } ``` It is up to the caller to ensure the output buffer is large enough to accommodate the new PCM frames. Input and output PCM frames are always interleaved. Deinterleaved layouts are not supported. 10.2.1. Channel Mapping ----------------------- In addition to converting from one channel count to another, like the example above, the channel converter can also be used to rearrange channels. When initializing the channel converter, you can optionally pass in channel maps for both the input and output frames. If the channel counts are the same, and each channel map contains the same channel positions with the exception that they're in a different order, a simple shuffling of the channels will be performed. If, however, there is not a 1:1 mapping of channel positions, or the channel counts differ, the input channels will be mixed based on a mixing mode which is specified when initializing the `ma_channel_converter_config` object. When converting from mono to multi-channel, the mono channel is simply copied to each output channel. When going the other way around, the audio of each output channel is simply averaged and copied to the mono channel. In more complicated cases blending is used. The `ma_channel_mix_mode_simple` mode will drop excess channels and silence extra channels. For example, converting from 4 to 2 channels, the 3rd and 4th channels will be dropped, whereas converting from 2 to 4 channels will put silence into the 3rd and 4th channels. The `ma_channel_mix_mode_rectangle` mode uses spacial locality based on a rectangle to compute a simple distribution between input and output. Imagine sitting in the middle of a room, with speakers on the walls representing channel positions. The `MA_CHANNEL_FRONT_LEFT` position can be thought of as being in the corner of the front and left walls. Finally, the `ma_channel_mix_mode_custom_weights` mode can be used to use custom user-defined weights. Custom weights can be passed in as the last parameter of `ma_channel_converter_config_init()`. Predefined channel maps can be retrieved with `ma_channel_map_init_standard()`. This takes a `ma_standard_channel_map` enum as its first parameter, which can be one of the following: +-----------------------------------+-----------------------------------------------------------+ | Name | Description | +-----------------------------------+-----------------------------------------------------------+ | ma_standard_channel_map_default | Default channel map used by miniaudio. See below. | | ma_standard_channel_map_microsoft | Channel map used by Microsoft's bitfield channel maps. | | ma_standard_channel_map_alsa | Default ALSA channel map. | | ma_standard_channel_map_rfc3551 | RFC 3551. Based on AIFF. | | ma_standard_channel_map_flac | FLAC channel map. | | ma_standard_channel_map_vorbis | Vorbis channel map. | | ma_standard_channel_map_sound4 | FreeBSD's sound(4). | | ma_standard_channel_map_sndio | sndio channel map. http://www.sndio.org/tips.html. | | ma_standard_channel_map_webaudio | https://webaudio.github.io/web-audio-api/#ChannelOrdering | +-----------------------------------+-----------------------------------------------------------+ Below are the channel maps used by default in miniaudio (`ma_standard_channel_map_default`): +---------------+---------------------------------+ | Channel Count | Mapping | +---------------+---------------------------------+ | 1 (Mono) | 0: MA_CHANNEL_MONO | +---------------+---------------------------------+ | 2 (Stereo) | 0: MA_CHANNEL_FRONT_LEFT <br> | | | 1: MA_CHANNEL_FRONT_RIGHT | +---------------+---------------------------------+ | 3 | 0: MA_CHANNEL_FRONT_LEFT <br> | | | 1: MA_CHANNEL_FRONT_RIGHT <br> | | | 2: MA_CHANNEL_FRONT_CENTER | +---------------+---------------------------------+ | 4 (Surround) | 0: MA_CHANNEL_FRONT_LEFT <br> | | | 1: MA_CHANNEL_FRONT_RIGHT <br> | | | 2: MA_CHANNEL_FRONT_CENTER <br> | | | 3: MA_CHANNEL_BACK_CENTER | +---------------+---------------------------------+ | 5 | 0: MA_CHANNEL_FRONT_LEFT <br> | | | 1: MA_CHANNEL_FRONT_RIGHT <br> | | | 2: MA_CHANNEL_FRONT_CENTER <br> | | | 3: MA_CHANNEL_BACK_LEFT <br> | | | 4: MA_CHANNEL_BACK_RIGHT | +---------------+---------------------------------+ | 6 (5.1) | 0: MA_CHANNEL_FRONT_LEFT <br> | | | 1: MA_CHANNEL_FRONT_RIGHT <br> | | | 2: MA_CHANNEL_FRONT_CENTER <br> | | | 3: MA_CHANNEL_LFE <br> | | | 4: MA_CHANNEL_SIDE_LEFT <br> | | | 5: MA_CHANNEL_SIDE_RIGHT | +---------------+---------------------------------+ | 7 | 0: MA_CHANNEL_FRONT_LEFT <br> | | | 1: MA_CHANNEL_FRONT_RIGHT <br> | | | 2: MA_CHANNEL_FRONT_CENTER <br> | | | 3: MA_CHANNEL_LFE <br> | | | 4: MA_CHANNEL_BACK_CENTER <br> | | | 4: MA_CHANNEL_SIDE_LEFT <br> | | | 5: MA_CHANNEL_SIDE_RIGHT | +---------------+---------------------------------+ | 8 (7.1) | 0: MA_CHANNEL_FRONT_LEFT <br> | | | 1: MA_CHANNEL_FRONT_RIGHT <br> | | | 2: MA_CHANNEL_FRONT_CENTER <br> | | | 3: MA_CHANNEL_LFE <br> | | | 4: MA_CHANNEL_BACK_LEFT <br> | | | 5: MA_CHANNEL_BACK_RIGHT <br> | | | 6: MA_CHANNEL_SIDE_LEFT <br> | | | 7: MA_CHANNEL_SIDE_RIGHT | +---------------+---------------------------------+ | Other | All channels set to 0. This | | | is equivalent to the same | | | mapping as the device. | +---------------+---------------------------------+ 10.3. Resampling ---------------- Resampling is achieved with the `ma_resampler` object. To create a resampler object, do something like the following: ```c ma_resampler_config config = ma_resampler_config_init( ma_format_s16, channels, sampleRateIn, sampleRateOut, ma_resample_algorithm_linear); ma_resampler resampler; ma_result result = ma_resampler_init(&config, NULL, &resampler); if (result != MA_SUCCESS) { // An error occurred... } ``` Do the following to uninitialize the resampler: ```c ma_resampler_uninit(&resampler); ``` The following example shows how data can be processed ```c ma_uint64 frameCountIn = 1000; ma_uint64 frameCountOut = 2000; ma_result result = ma_resampler_process_pcm_frames(&resampler, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut); if (result != MA_SUCCESS) { // An error occurred... } // At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the // number of output frames written. ``` To initialize the resampler you first need to set up a config (`ma_resampler_config`) with `ma_resampler_config_init()`. You need to specify the sample format you want to use, the number of channels, the input and output sample rate, and the algorithm. The sample format can be either `ma_format_s16` or `ma_format_f32`. If you need a different format you will need to perform pre- and post-conversions yourself where necessary. Note that the format is the same for both input and output. The format cannot be changed after initialization. The resampler supports multiple channels and is always interleaved (both input and output). The channel count cannot be changed after initialization. The sample rates can be anything other than zero, and are always specified in hertz. They should be set to something like 44100, etc. The sample rate is the only configuration property that can be changed after initialization. The miniaudio resampler has built-in support for the following algorithms: +-----------+------------------------------+ | Algorithm | Enum Token | +-----------+------------------------------+ | Linear | ma_resample_algorithm_linear | | Custom | ma_resample_algorithm_custom | +-----------+------------------------------+ The algorithm cannot be changed after initialization. Processing always happens on a per PCM frame basis and always assumes interleaved input and output. De-interleaved processing is not supported. To process frames, use `ma_resampler_process_pcm_frames()`. On input, this function takes the number of output frames you can fit in the output buffer and the number of input frames contained in the input buffer. On output these variables contain the number of output frames that were written to the output buffer and the number of input frames that were consumed in the process. You can pass in NULL for the input buffer in which case it will be treated as an infinitely large buffer of zeros. The output buffer can also be NULL, in which case the processing will be treated as seek. The sample rate can be changed dynamically on the fly. You can change this with explicit sample rates with `ma_resampler_set_rate()` and also with a decimal ratio with `ma_resampler_set_rate_ratio()`. The ratio is in/out. Sometimes it's useful to know exactly how many input frames will be required to output a specific number of frames. You can calculate this with `ma_resampler_get_required_input_frame_count()`. Likewise, it's sometimes useful to know exactly how many frames would be output given a certain number of input frames. You can do this with `ma_resampler_get_expected_output_frame_count()`. Due to the nature of how resampling works, the resampler introduces some latency. This can be retrieved in terms of both the input rate and the output rate with `ma_resampler_get_input_latency()` and `ma_resampler_get_output_latency()`. 10.3.1. Resampling Algorithms ----------------------------- The choice of resampling algorithm depends on your situation and requirements. 10.3.1.1. Linear Resampling --------------------------- The linear resampler is the fastest, but comes at the expense of poorer quality. There is, however, some control over the quality of the linear resampler which may make it a suitable option depending on your requirements. The linear resampler performs low-pass filtering before or after downsampling or upsampling, depending on the sample rates you're converting between. When decreasing the sample rate, the low-pass filter will be applied before downsampling. When increasing the rate it will be performed after upsampling. By default a fourth order low-pass filter will be applied. This can be configured via the `lpfOrder` configuration variable. Setting this to 0 will disable filtering. The low-pass filter has a cutoff frequency which defaults to half the sample rate of the lowest of the input and output sample rates (Nyquist Frequency). The API for the linear resampler is the same as the main resampler API, only it's called `ma_linear_resampler`. 10.3.2. Custom Resamplers ------------------------- You can implement a custom resampler by using the `ma_resample_algorithm_custom` resampling algorithm and setting a vtable in the resampler config: ```c ma_resampler_config config = ma_resampler_config_init(..., ma_resample_algorithm_custom); config.pBackendVTable = &g_customResamplerVTable; ``` Custom resamplers are useful if the stock algorithms are not appropriate for your use case. You need to implement the required functions in `ma_resampling_backend_vtable`. Note that not all functions in the vtable need to be implemented, but if it's possible to implement, they should be. You can use the `ma_linear_resampler` object for an example on how to implement the vtable. The `onGetHeapSize` callback is used to calculate the size of any internal heap allocation the custom resampler will need to make given the supplied config. When you initialize the resampler via the `onInit` callback, you'll be given a pointer to a heap allocation which is where you should store the heap allocated data. You should not free this data in `onUninit` because miniaudio will manage it for you. The `onProcess` callback is where the actual resampling takes place. On input, `pFrameCountIn` points to a variable containing the number of frames in the `pFramesIn` buffer and `pFrameCountOut` points to a variable containing the capacity in frames of the `pFramesOut` buffer. On output, `pFrameCountIn` should be set to the number of input frames that were fully consumed, whereas `pFrameCountOut` should be set to the number of frames that were written to `pFramesOut`. The `onSetRate` callback is optional and is used for dynamically changing the sample rate. If dynamic rate changes are not supported, you can set this callback to NULL. The `onGetInputLatency` and `onGetOutputLatency` functions are used for retrieving the latency in input and output rates respectively. These can be NULL in which case latency calculations will be assumed to be NULL. The `onGetRequiredInputFrameCount` callback is used to give miniaudio a hint as to how many input frames are required to be available to produce the given number of output frames. Likewise, the `onGetExpectedOutputFrameCount` callback is used to determine how many output frames will be produced given the specified number of input frames. miniaudio will use these as a hint, but they are optional and can be set to NULL if you're unable to implement them. 10.4. General Data Conversion ----------------------------- The `ma_data_converter` API can be used to wrap sample format conversion, channel conversion and resampling into one operation. This is what miniaudio uses internally to convert between the format requested when the device was initialized and the format of the backend's native device. The API for general data conversion is very similar to the resampling API. Create a `ma_data_converter` object like this: ```c ma_data_converter_config config = ma_data_converter_config_init( inputFormat, outputFormat, inputChannels, outputChannels, inputSampleRate, outputSampleRate ); ma_data_converter converter; ma_result result = ma_data_converter_init(&config, NULL, &converter); if (result != MA_SUCCESS) { // An error occurred... } ``` In the example above we use `ma_data_converter_config_init()` to initialize the config, however there's many more properties that can be configured, such as channel maps and resampling quality. Something like the following may be more suitable depending on your requirements: ```c ma_data_converter_config config = ma_data_converter_config_init_default(); config.formatIn = inputFormat; config.formatOut = outputFormat; config.channelsIn = inputChannels; config.channelsOut = outputChannels; config.sampleRateIn = inputSampleRate; config.sampleRateOut = outputSampleRate; ma_channel_map_init_standard(ma_standard_channel_map_flac, config.channelMapIn, sizeof(config.channelMapIn)/sizeof(config.channelMapIn[0]), config.channelCountIn); config.resampling.linear.lpfOrder = MA_MAX_FILTER_ORDER; ``` Do the following to uninitialize the data converter: ```c ma_data_converter_uninit(&converter, NULL); ``` The following example shows how data can be processed ```c ma_uint64 frameCountIn = 1000; ma_uint64 frameCountOut = 2000; ma_result result = ma_data_converter_process_pcm_frames(&converter, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut); if (result != MA_SUCCESS) { // An error occurred... } // At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the number // of output frames written. ``` The data converter supports multiple channels and is always interleaved (both input and output). The channel count cannot be changed after initialization. Sample rates can be anything other than zero, and are always specified in hertz. They should be set to something like 44100, etc. The sample rate is the only configuration property that can be changed after initialization, but only if the `resampling.allowDynamicSampleRate` member of `ma_data_converter_config` is set to `MA_TRUE`. To change the sample rate, use `ma_data_converter_set_rate()` or `ma_data_converter_set_rate_ratio()`. The ratio must be in/out. The resampling algorithm cannot be changed after initialization. Processing always happens on a per PCM frame basis and always assumes interleaved input and output. De-interleaved processing is not supported. To process frames, use `ma_data_converter_process_pcm_frames()`. On input, this function takes the number of output frames you can fit in the output buffer and the number of input frames contained in the input buffer. On output these variables contain the number of output frames that were written to the output buffer and the number of input frames that were consumed in the process. You can pass in NULL for the input buffer in which case it will be treated as an infinitely large buffer of zeros. The output buffer can also be NULL, in which case the processing will be treated as seek. Sometimes it's useful to know exactly how many input frames will be required to output a specific number of frames. You can calculate this with `ma_data_converter_get_required_input_frame_count()`. Likewise, it's sometimes useful to know exactly how many frames would be output given a certain number of input frames. You can do this with `ma_data_converter_get_expected_output_frame_count()`. Due to the nature of how resampling works, the data converter introduces some latency if resampling is required. This can be retrieved in terms of both the input rate and the output rate with `ma_data_converter_get_input_latency()` and `ma_data_converter_get_output_latency()`. 11. Filtering ============= 11.1. Biquad Filtering ---------------------- Biquad filtering is achieved with the `ma_biquad` API. Example: ```c ma_biquad_config config = ma_biquad_config_init(ma_format_f32, channels, b0, b1, b2, a0, a1, a2); ma_result result = ma_biquad_init(&config, NULL, &biquad); if (result != MA_SUCCESS) { // Error. } ... ma_biquad_process_pcm_frames(&biquad, pFramesOut, pFramesIn, frameCount); ``` Biquad filtering is implemented using transposed direct form 2. The numerator coefficients are b0, b1 and b2, and the denominator coefficients are a0, a1 and a2. The a0 coefficient is required and coefficients must not be pre-normalized. Supported formats are `ma_format_s16` and `ma_format_f32`. If you need to use a different format you need to convert it yourself beforehand. When using `ma_format_s16` the biquad filter will use fixed point arithmetic. When using `ma_format_f32`, floating point arithmetic will be used. Input and output frames are always interleaved. Filtering can be applied in-place by passing in the same pointer for both the input and output buffers, like so: ```c ma_biquad_process_pcm_frames(&biquad, pMyData, pMyData, frameCount); ``` If you need to change the values of the coefficients, but maintain the values in the registers you can do so with `ma_biquad_reinit()`. This is useful if you need to change the properties of the filter while keeping the values of registers valid to avoid glitching. Do not use `ma_biquad_init()` for this as it will do a full initialization which involves clearing the registers to 0. Note that changing the format or channel count after initialization is invalid and will result in an error. 11.2. Low-Pass Filtering ------------------------ Low-pass filtering is achieved with the following APIs: +---------+------------------------------------------+ | API | Description | +---------+------------------------------------------+ | ma_lpf1 | First order low-pass filter | | ma_lpf2 | Second order low-pass filter | | ma_lpf | High order low-pass filter (Butterworth) | +---------+------------------------------------------+ Low-pass filter example: ```c ma_lpf_config config = ma_lpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency, order); ma_result result = ma_lpf_init(&config, &lpf); if (result != MA_SUCCESS) { // Error. } ... ma_lpf_process_pcm_frames(&lpf, pFramesOut, pFramesIn, frameCount); ``` Supported formats are `ma_format_s16` and` ma_format_f32`. If you need to use a different format you need to convert it yourself beforehand. Input and output frames are always interleaved. Filtering can be applied in-place by passing in the same pointer for both the input and output buffers, like so: ```c ma_lpf_process_pcm_frames(&lpf, pMyData, pMyData, frameCount); ``` The maximum filter order is limited to `MA_MAX_FILTER_ORDER` which is set to 8. If you need more, you can chain first and second order filters together. ```c for (iFilter = 0; iFilter < filterCount; iFilter += 1) { ma_lpf2_process_pcm_frames(&lpf2[iFilter], pMyData, pMyData, frameCount); } ``` If you need to change the configuration of the filter, but need to maintain the state of internal registers you can do so with `ma_lpf_reinit()`. This may be useful if you need to change the sample rate and/or cutoff frequency dynamically while maintaining smooth transitions. Note that changing the format or channel count after initialization is invalid and will result in an error. The `ma_lpf` object supports a configurable order, but if you only need a first order filter you may want to consider using `ma_lpf1`. Likewise, if you only need a second order filter you can use `ma_lpf2`. The advantage of this is that they're lighter weight and a bit more efficient. If an even filter order is specified, a series of second order filters will be processed in a chain. If an odd filter order is specified, a first order filter will be applied, followed by a series of second order filters in a chain. 11.3. High-Pass Filtering ------------------------- High-pass filtering is achieved with the following APIs: +---------+-------------------------------------------+ | API | Description | +---------+-------------------------------------------+ | ma_hpf1 | First order high-pass filter | | ma_hpf2 | Second order high-pass filter | | ma_hpf | High order high-pass filter (Butterworth) | +---------+-------------------------------------------+ High-pass filters work exactly the same as low-pass filters, only the APIs are called `ma_hpf1`, `ma_hpf2` and `ma_hpf`. See example code for low-pass filters for example usage. 11.4. Band-Pass Filtering ------------------------- Band-pass filtering is achieved with the following APIs: +---------+-------------------------------+ | API | Description | +---------+-------------------------------+ | ma_bpf2 | Second order band-pass filter | | ma_bpf | High order band-pass filter | +---------+-------------------------------+ Band-pass filters work exactly the same as low-pass filters, only the APIs are called `ma_bpf2` and `ma_hpf`. See example code for low-pass filters for example usage. Note that the order for band-pass filters must be an even number which means there is no first order band-pass filter, unlike low-pass and high-pass filters. 11.5. Notch Filtering --------------------- Notch filtering is achieved with the following APIs: +-----------+------------------------------------------+ | API | Description | +-----------+------------------------------------------+ | ma_notch2 | Second order notching filter | +-----------+------------------------------------------+ 11.6. Peaking EQ Filtering ------------------------- Peaking filtering is achieved with the following APIs: +----------+------------------------------------------+ | API | Description | +----------+------------------------------------------+ | ma_peak2 | Second order peaking filter | +----------+------------------------------------------+ 11.7. Low Shelf Filtering ------------------------- Low shelf filtering is achieved with the following APIs: +-------------+------------------------------------------+ | API | Description | +-------------+------------------------------------------+ | ma_loshelf2 | Second order low shelf filter | +-------------+------------------------------------------+ Where a high-pass filter is used to eliminate lower frequencies, a low shelf filter can be used to just turn them down rather than eliminate them entirely. 11.8. High Shelf Filtering -------------------------- High shelf filtering is achieved with the following APIs: +-------------+------------------------------------------+ | API | Description | +-------------+------------------------------------------+ | ma_hishelf2 | Second order high shelf filter | +-------------+------------------------------------------+ The high shelf filter has the same API as the low shelf filter, only you would use `ma_hishelf` instead of `ma_loshelf`. Where a low shelf filter is used to adjust the volume of low frequencies, the high shelf filter does the same thing for high frequencies. 12. Waveform and Noise Generation ================================= 12.1. Waveforms --------------- miniaudio supports generation of sine, square, triangle and sawtooth waveforms. This is achieved with the `ma_waveform` API. Example: ```c ma_waveform_config config = ma_waveform_config_init( FORMAT, CHANNELS, SAMPLE_RATE, ma_waveform_type_sine, amplitude, frequency); ma_waveform waveform; ma_result result = ma_waveform_init(&config, &waveform); if (result != MA_SUCCESS) { // Error. } ... ma_waveform_read_pcm_frames(&waveform, pOutput, frameCount); ``` The amplitude, frequency, type, and sample rate can be changed dynamically with `ma_waveform_set_amplitude()`, `ma_waveform_set_frequency()`, `ma_waveform_set_type()`, and `ma_waveform_set_sample_rate()` respectively. You can invert the waveform by setting the amplitude to a negative value. You can use this to control whether or not a sawtooth has a positive or negative ramp, for example. Below are the supported waveform types: +---------------------------+ | Enum Name | +---------------------------+ | ma_waveform_type_sine | | ma_waveform_type_square | | ma_waveform_type_triangle | | ma_waveform_type_sawtooth | +---------------------------+ 12.2. Noise ----------- miniaudio supports generation of white, pink and Brownian noise via the `ma_noise` API. Example: ```c ma_noise_config config = ma_noise_config_init( FORMAT, CHANNELS, ma_noise_type_white, SEED, amplitude); ma_noise noise; ma_result result = ma_noise_init(&config, &noise); if (result != MA_SUCCESS) { // Error. } ... ma_noise_read_pcm_frames(&noise, pOutput, frameCount); ``` The noise API uses simple LCG random number generation. It supports a custom seed which is useful for things like automated testing requiring reproducibility. Setting the seed to zero will default to `MA_DEFAULT_LCG_SEED`. The amplitude and seed can be changed dynamically with `ma_noise_set_amplitude()` and `ma_noise_set_seed()` respectively. By default, the noise API will use different values for different channels. So, for example, the left side in a stereo stream will be different to the right side. To instead have each channel use the same random value, set the `duplicateChannels` member of the noise config to true, like so: ```c config.duplicateChannels = MA_TRUE; ``` Below are the supported noise types. +------------------------+ | Enum Name | +------------------------+ | ma_noise_type_white | | ma_noise_type_pink | | ma_noise_type_brownian | +------------------------+ 13. Audio Buffers ================= miniaudio supports reading from a buffer of raw audio data via the `ma_audio_buffer` API. This can read from memory that's managed by the application, but can also handle the memory management for you internally. Memory management is flexible and should support most use cases. Audio buffers are initialized using the standard configuration system used everywhere in miniaudio: ```c ma_audio_buffer_config config = ma_audio_buffer_config_init( format, channels, sizeInFrames, pExistingData, &allocationCallbacks); ma_audio_buffer buffer; result = ma_audio_buffer_init(&config, &buffer); if (result != MA_SUCCESS) { // Error. } ... ma_audio_buffer_uninit(&buffer); ``` In the example above, the memory pointed to by `pExistingData` will *not* be copied and is how an application can do self-managed memory allocation. If you would rather make a copy of the data, use `ma_audio_buffer_init_copy()`. To uninitialize the buffer, use `ma_audio_buffer_uninit()`. Sometimes it can be convenient to allocate the memory for the `ma_audio_buffer` structure and the raw audio data in a contiguous block of memory. That is, the raw audio data will be located immediately after the `ma_audio_buffer` structure. To do this, use `ma_audio_buffer_alloc_and_init()`: ```c ma_audio_buffer_config config = ma_audio_buffer_config_init( format, channels, sizeInFrames, pExistingData, &allocationCallbacks); ma_audio_buffer* pBuffer result = ma_audio_buffer_alloc_and_init(&config, &pBuffer); if (result != MA_SUCCESS) { // Error } ... ma_audio_buffer_uninit_and_free(&buffer); ``` If you initialize the buffer with `ma_audio_buffer_alloc_and_init()` you should uninitialize it with `ma_audio_buffer_uninit_and_free()`. In the example above, the memory pointed to by `pExistingData` will be copied into the buffer, which is contrary to the behavior of `ma_audio_buffer_init()`. An audio buffer has a playback cursor just like a decoder. As you read frames from the buffer, the cursor moves forward. The last parameter (`loop`) can be used to determine if the buffer should loop. The return value is the number of frames actually read. If this is less than the number of frames requested it means the end has been reached. This should never happen if the `loop` parameter is set to true. If you want to manually loop back to the start, you can do so with with `ma_audio_buffer_seek_to_pcm_frame(pAudioBuffer, 0)`. Below is an example for reading data from an audio buffer. ```c ma_uint64 framesRead = ma_audio_buffer_read_pcm_frames(pAudioBuffer, pFramesOut, desiredFrameCount, isLooping); if (framesRead < desiredFrameCount) { // If not looping, this means the end has been reached. This should never happen in looping mode with valid input. } ``` Sometimes you may want to avoid the cost of data movement between the internal buffer and the output buffer. Instead you can use memory mapping to retrieve a pointer to a segment of data: ```c void* pMappedFrames; ma_uint64 frameCount = frameCountToTryMapping; ma_result result = ma_audio_buffer_map(pAudioBuffer, &pMappedFrames, &frameCount); if (result == MA_SUCCESS) { // Map was successful. The value in frameCount will be how many frames were _actually_ mapped, which may be // less due to the end of the buffer being reached. ma_copy_pcm_frames(pFramesOut, pMappedFrames, frameCount, pAudioBuffer->format, pAudioBuffer->channels); // You must unmap the buffer. ma_audio_buffer_unmap(pAudioBuffer, frameCount); } ``` When you use memory mapping, the read cursor is increment by the frame count passed in to `ma_audio_buffer_unmap()`. If you decide not to process every frame you can pass in a value smaller than the value returned by `ma_audio_buffer_map()`. The disadvantage to using memory mapping is that it does not handle looping for you. You can determine if the buffer is at the end for the purpose of looping with `ma_audio_buffer_at_end()` or by inspecting the return value of `ma_audio_buffer_unmap()` and checking if it equals `MA_AT_END`. You should not treat `MA_AT_END` as an error when returned by `ma_audio_buffer_unmap()`. 14. Ring Buffers ================ miniaudio supports lock free (single producer, single consumer) ring buffers which are exposed via the `ma_rb` and `ma_pcm_rb` APIs. The `ma_rb` API operates on bytes, whereas the `ma_pcm_rb` operates on PCM frames. They are otherwise identical as `ma_pcm_rb` is just a wrapper around `ma_rb`. Unlike most other APIs in miniaudio, ring buffers support both interleaved and deinterleaved streams. The caller can also allocate their own backing memory for the ring buffer to use internally for added flexibility. Otherwise the ring buffer will manage it's internal memory for you. The examples below use the PCM frame variant of the ring buffer since that's most likely the one you will want to use. To initialize a ring buffer, do something like the following: ```c ma_pcm_rb rb; ma_result result = ma_pcm_rb_init(FORMAT, CHANNELS, BUFFER_SIZE_IN_FRAMES, NULL, NULL, &rb); if (result != MA_SUCCESS) { // Error } ``` The `ma_pcm_rb_init()` function takes the sample format and channel count as parameters because it's the PCM variant of the ring buffer API. For the regular ring buffer that operates on bytes you would call `ma_rb_init()` which leaves these out and just takes the size of the buffer in bytes instead of frames. The fourth parameter is an optional pre-allocated buffer and the fifth parameter is a pointer to a `ma_allocation_callbacks` structure for custom memory allocation routines. Passing in `NULL` for this results in `MA_MALLOC()` and `MA_FREE()` being used. Use `ma_pcm_rb_init_ex()` if you need a deinterleaved buffer. The data for each sub-buffer is offset from each other based on the stride. To manage your sub-buffers you can use `ma_pcm_rb_get_subbuffer_stride()`, `ma_pcm_rb_get_subbuffer_offset()` and `ma_pcm_rb_get_subbuffer_ptr()`. Use `ma_pcm_rb_acquire_read()` and `ma_pcm_rb_acquire_write()` to retrieve a pointer to a section of the ring buffer. You specify the number of frames you need, and on output it will set to what was actually acquired. If the read or write pointer is positioned such that the number of frames requested will require a loop, it will be clamped to the end of the buffer. Therefore, the number of frames you're given may be less than the number you requested. After calling `ma_pcm_rb_acquire_read()` or `ma_pcm_rb_acquire_write()`, you do your work on the buffer and then "commit" it with `ma_pcm_rb_commit_read()` or `ma_pcm_rb_commit_write()`. This is where the read/write pointers are updated. When you commit you need to pass in the buffer that was returned by the earlier call to `ma_pcm_rb_acquire_read()` or `ma_pcm_rb_acquire_write()` and is only used for validation. The number of frames passed to `ma_pcm_rb_commit_read()` and `ma_pcm_rb_commit_write()` is what's used to increment the pointers, and can be less that what was originally requested. If you want to correct for drift between the write pointer and the read pointer you can use a combination of `ma_pcm_rb_pointer_distance()`, `ma_pcm_rb_seek_read()` and `ma_pcm_rb_seek_write()`. Note that you can only move the pointers forward, and you should only move the read pointer forward via the consumer thread, and the write pointer forward by the producer thread. If there is too much space between the pointers, move the read pointer forward. If there is too little space between the pointers, move the write pointer forward. You can use a ring buffer at the byte level instead of the PCM frame level by using the `ma_rb` API. This is exactly the same, only you will use the `ma_rb` functions instead of `ma_pcm_rb` and instead of frame counts you will pass around byte counts. The maximum size of the buffer in bytes is `0x7FFFFFFF-(MA_SIMD_ALIGNMENT-1)` due to the most significant bit being used to encode a loop flag and the internally managed buffers always being aligned to `MA_SIMD_ALIGNMENT`. Note that the ring buffer is only thread safe when used by a single consumer thread and single producer thread. 15. Backends ============ The following backends are supported by miniaudio. These are listed in order of default priority. When no backend is specified when initializing a context or device, miniaudio will attempt to use each of these backends in the order listed in the table below. Note that backends that are not usable by the build target will not be included in the build. For example, ALSA, which is specific to Linux, will not be included in the Windows build. +-------------+-----------------------+--------------------------------------------------------+ | Name | Enum Name | Supported Operating Systems | +-------------+-----------------------+--------------------------------------------------------+ | WASAPI | ma_backend_wasapi | Windows Vista+ | | DirectSound | ma_backend_dsound | Windows XP+ | | WinMM | ma_backend_winmm | Windows 95+ | | Core Audio | ma_backend_coreaudio | macOS, iOS | | sndio | ma_backend_sndio | OpenBSD | | audio(4) | ma_backend_audio4 | NetBSD, OpenBSD | | OSS | ma_backend_oss | FreeBSD | | PulseAudio | ma_backend_pulseaudio | Cross Platform (disabled on Windows, BSD and Android) | | ALSA | ma_backend_alsa | Linux | | JACK | ma_backend_jack | Cross Platform (disabled on BSD and Android) | | AAudio | ma_backend_aaudio | Android 8+ | | OpenSL ES | ma_backend_opensl | Android (API level 16+) | | Web Audio | ma_backend_webaudio | Web (via Emscripten) | | Custom | ma_backend_custom | Cross Platform | | Null | ma_backend_null | Cross Platform (not used on Web) | +-------------+-----------------------+--------------------------------------------------------+ Some backends have some nuance details you may want to be aware of. 15.1. WASAPI ------------ - Low-latency shared mode will be disabled when using an application-defined sample rate which is different to the device's native sample rate. To work around this, set `wasapi.noAutoConvertSRC` to true in the device config. This is due to IAudioClient3_InitializeSharedAudioStream() failing when the `AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM` flag is specified. Setting wasapi.noAutoConvertSRC will result in miniaudio's internal resampler being used instead which will in turn enable the use of low-latency shared mode. 15.2. PulseAudio ---------------- - If you experience bad glitching/noise on Arch Linux, consider this fix from the Arch wiki: https://wiki.archlinux.org/index.php/PulseAudio/Troubleshooting#Glitches,_skips_or_crackling. Alternatively, consider using a different backend such as ALSA. 15.3. Android ------------- - To capture audio on Android, remember to add the RECORD_AUDIO permission to your manifest: `<uses-permission android:name="android.permission.RECORD_AUDIO" />` - With OpenSL|ES, only a single ma_context can be active at any given time. This is due to a limitation with OpenSL|ES. - With AAudio, only default devices are enumerated. This is due to AAudio not having an enumeration API (devices are enumerated through Java). You can however perform your own device enumeration through Java and then set the ID in the ma_device_id structure (ma_device_id.aaudio) and pass it to ma_device_init(). - The backend API will perform resampling where possible. The reason for this as opposed to using miniaudio's built-in resampler is to take advantage of any potential device-specific optimizations the driver may implement. BSD --- - The sndio backend is currently only enabled on OpenBSD builds. - The audio(4) backend is supported on OpenBSD, but you may need to disable sndiod before you can use it. 15.4. UWP --------- - UWP only supports default playback and capture devices. - UWP requires the Microphone capability to be enabled in the application's manifest (Package.appxmanifest): ``` <Package ...> ... <Capabilities> <DeviceCapability Name="microphone" /> </Capabilities> </Package> ``` 15.5. Web Audio / Emscripten ---------------------------- - You cannot use `-std=c*` compiler flags, nor `-ansi`. This only applies to the Emscripten build. - The first time a context is initialized it will create a global object called "miniaudio" whose primary purpose is to act as a factory for device objects. - Currently the Web Audio backend uses ScriptProcessorNode's, but this may need to change later as they've been deprecated. - Google has implemented a policy in their browsers that prevent automatic media output without first receiving some kind of user input. The following web page has additional details: https://developers.google.com/web/updates/2017/09/autoplay-policy-changes. Starting the device may fail if you try to start playback without first handling some kind of user input. 16. Optimization Tips ===================== See below for some tips on improving performance. 16.1. Low Level API ------------------- - In the data callback, if your data is already clipped prior to copying it into the output buffer, set the `noClip` config option in the device config to true. This will disable miniaudio's built in clipping function. - By default, miniaudio will pre-silence the data callback's output buffer. If you know that you will always write valid data to the output buffer you can disable pre-silencing by setting the `noPreSilence` config option in the device config to true. 16.2. High Level API -------------------- - If a sound does not require doppler or pitch shifting, consider disabling pitching by initializing the sound with the `MA_SOUND_FLAG_NO_PITCH` flag. - If a sound does not require spatialization, disable it by initializing the sound with the `MA_SOUND_FLAG_NO_SPATIALIZATION` flag. It can be re-enabled again post-initialization with `ma_sound_set_spatialization_enabled()`. - If you know all of your sounds will always be the same sample rate, set the engine's sample rate to match that of the sounds. Likewise, if you're using a self-managed resource manager, consider setting the decoded sample rate to match your sounds. By configuring everything to use a consistent sample rate, sample rate conversion can be avoided. 17. Miscellaneous Notes ======================= - Automatic stream routing is enabled on a per-backend basis. Support is explicitly enabled for WASAPI and Core Audio, however other backends such as PulseAudio may naturally support it, though not all have been tested. - When compiling with VC6 and earlier, decoding is restricted to files less than 2GB in size. This is due to 64-bit file APIs not being available. */ #ifndef miniaudio_h #define miniaudio_h #ifdef __cplusplus extern "C" { #endif #define MA_STRINGIFY(x) #x #define MA_XSTRINGIFY(x) MA_STRINGIFY(x) #define MA_VERSION_MAJOR 0 #define MA_VERSION_MINOR 11 #define MA_VERSION_REVISION 24 #define MA_VERSION_STRING MA_XSTRINGIFY(MA_VERSION_MAJOR) "." MA_XSTRINGIFY(MA_VERSION_MINOR) "." MA_XSTRINGIFY(MA_VERSION_REVISION) #if defined(_MSC_VER) && !defined(__clang__) #pragma warning(push) #pragma warning(disable:4201) /* nonstandard extension used: nameless struct/union */ #pragma warning(disable:4214) /* nonstandard extension used: bit field types other than int */ #pragma warning(disable:4324) /* structure was padded due to alignment specifier */ #elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))) #pragma GCC diagnostic push #pragma GCC diagnostic ignored "-Wpedantic" /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */ #if defined(__clang__) #pragma GCC diagnostic ignored "-Wc11-extensions" /* anonymous unions are a C11 extension */ #endif #endif #if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined(_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__) || defined(__ppc64__) #define MA_SIZEOF_PTR 8 #else #define MA_SIZEOF_PTR 4 #endif #include <stddef.h> /* For size_t. */ /* Sized types. */ #if defined(MA_USE_STDINT) #include <stdint.h> typedef int8_t ma_int8; typedef uint8_t ma_uint8; typedef int16_t ma_int16; typedef uint16_t ma_uint16; typedef int32_t ma_int32; typedef uint32_t ma_uint32; typedef int64_t ma_int64; typedef uint64_t ma_uint64; #else typedef signed char ma_int8; typedef unsigned char ma_uint8; typedef signed short ma_int16; typedef unsigned short ma_uint16; typedef signed int ma_int32; typedef unsigned int ma_uint32; #if defined(_MSC_VER) && !defined(__clang__) typedef signed __int64 ma_int64; typedef unsigned __int64 ma_uint64; #else #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) #pragma GCC diagnostic push #pragma GCC diagnostic ignored "-Wlong-long" #if defined(__clang__) #pragma GCC diagnostic ignored "-Wc++11-long-long" #endif #endif typedef signed long long ma_int64; typedef unsigned long long ma_uint64; #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) #pragma GCC diagnostic pop #endif #endif #endif /* MA_USE_STDINT */ #if MA_SIZEOF_PTR == 8 typedef ma_uint64 ma_uintptr; #else typedef ma_uint32 ma_uintptr; #endif typedef ma_uint8 ma_bool8; typedef ma_uint32 ma_bool32; #define MA_TRUE 1 #define MA_FALSE 0 /* These float types are not used universally by miniaudio. It's to simplify some macro expansion for atomic types. */ typedef float ma_float; typedef double ma_double; typedef void* ma_handle; typedef void* ma_ptr; /* ma_proc is annoying because when compiling with GCC we get pedantic warnings about converting between `void*` and `void (*)()`. We can't use `void (*)()` with MSVC however, because we'll get warning C4191 about "type cast between incompatible function types". To work around this I'm going to use a different data type depending on the compiler. */ #if defined(__GNUC__) typedef void (*ma_proc)(void); #else typedef void* ma_proc; #endif #if defined(_MSC_VER) && !defined(_WCHAR_T_DEFINED) typedef ma_uint16 wchar_t; #endif /* Define NULL for some compilers. */ #ifndef NULL #define NULL 0 #endif #if defined(SIZE_MAX) #define MA_SIZE_MAX SIZE_MAX #else #define MA_SIZE_MAX 0xFFFFFFFF /* When SIZE_MAX is not defined by the standard library just default to the maximum 32-bit unsigned integer. */ #endif #define MA_UINT64_MAX (((ma_uint64)0xFFFFFFFF << 32) | (ma_uint64)0xFFFFFFFF) /* Weird shifting syntax is for VC6 compatibility. */ /* Platform/backend detection. */ #if defined(_WIN32) #define MA_WIN32 #if defined(MA_FORCE_UWP) || (defined(WINAPI_FAMILY) && ((defined(WINAPI_FAMILY_PC_APP) && WINAPI_FAMILY == WINAPI_FAMILY_PC_APP) || (defined(WINAPI_FAMILY_PHONE_APP) && WINAPI_FAMILY == WINAPI_FAMILY_PHONE_APP))) #define MA_WIN32_UWP #elif defined(WINAPI_FAMILY) && (defined(WINAPI_FAMILY_GAMES) && WINAPI_FAMILY == WINAPI_FAMILY_GAMES) #define MA_WIN32_GDK #elif defined(NXDK) #define MA_WIN32_NXDK #else #define MA_WIN32_DESKTOP #endif /* The original Xbox. */ #if defined(NXDK) /* <-- Add other Xbox compiler toolchains here, and then add a toolchain-specific define in case we need to discriminate between them later. */ #define MA_XBOX #if defined(NXDK) #define MA_XBOX_NXDK #endif #endif #endif #if defined(__MSDOS__) || defined(MSDOS) || defined(_MSDOS) || defined(__DOS__) #define MA_DOS /* No threading allowed on DOS. */ #ifndef MA_NO_THREADING #define MA_NO_THREADING #endif /* No runtime linking allowed on DOS. */ #ifndef MA_NO_RUNTIME_LINKING #define MA_NO_RUNTIME_LINKING #endif #endif #if !defined(MA_WIN32) && !defined(MA_DOS) /* If it's not Win32, assume POSIX. */ #define MA_POSIX #if !defined(MA_NO_THREADING) /* Use the MA_NO_PTHREAD_IN_HEADER option at your own risk. This is intentionally undocumented. You can use this to avoid including pthread.h in the header section. The downside is that it results in some fixed sized structures being declared for the various types that are used in miniaudio. The risk here is that these types might be too small for a given platform. This risk is yours to take and no support will be offered if you enable this option. */ #ifndef MA_NO_PTHREAD_IN_HEADER #include <pthread.h> /* Unfortunate #include, but needed for pthread_t, pthread_mutex_t and pthread_cond_t types. */ typedef pthread_t ma_pthread_t; typedef pthread_mutex_t ma_pthread_mutex_t; typedef pthread_cond_t ma_pthread_cond_t; #else typedef ma_uintptr ma_pthread_t; typedef union ma_pthread_mutex_t { char __data[40]; ma_uint64 __alignment; } ma_pthread_mutex_t; typedef union ma_pthread_cond_t { char __data[48]; ma_uint64 __alignment; } ma_pthread_cond_t; #endif #endif #if defined(__unix__) #define MA_UNIX #endif #if defined(__linux__) #define MA_LINUX #endif #if defined(__APPLE__) #define MA_APPLE #endif #if defined(__DragonFly__) || defined(__FreeBSD__) || defined(__NetBSD__) || defined(__OpenBSD__) #define MA_BSD #endif #if defined(__ANDROID__) #define MA_ANDROID #endif #if defined(__EMSCRIPTEN__) #define MA_EMSCRIPTEN #endif #if defined(__ORBIS__) #define MA_ORBIS #endif #if defined(__PROSPERO__) #define MA_PROSPERO #endif #if defined(__3DS__) #define MA_3DS #endif #if defined(__SWITCH__) || defined(__NX__) #define MA_SWITCH #endif #if defined(__BEOS__) || defined(__HAIKU__) #define MA_BEOS #endif #if defined(__HAIKU__) #define MA_HAIKU #endif #endif #if !defined(MA_FALLTHROUGH) && defined(__cplusplus) && __cplusplus >= 201703L #define MA_FALLTHROUGH [[fallthrough]] #endif #if !defined(MA_FALLTHROUGH) && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 202000L #define MA_FALLTHROUGH [[fallthrough]] #endif #if !defined(MA_FALLTHROUGH) && defined(__has_attribute) #if __has_attribute(fallthrough) #define MA_FALLTHROUGH __attribute__((fallthrough)) #endif #endif #if !defined(MA_FALLTHROUGH) #define MA_FALLTHROUGH ((void)0) #endif #ifdef _MSC_VER #define MA_INLINE __forceinline /* noinline was introduced in Visual Studio 2005. */ #if _MSC_VER >= 1400 #define MA_NO_INLINE __declspec(noinline) #else #define MA_NO_INLINE #endif #elif defined(__GNUC__) /* I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue I am using "__inline__" only when we're compiling in strict ANSI mode. */ #if defined(__STRICT_ANSI__) #define MA_GNUC_INLINE_HINT __inline__ #else #define MA_GNUC_INLINE_HINT inline #endif #if (__GNUC__ > 3 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 2)) || defined(__clang__) #define MA_INLINE MA_GNUC_INLINE_HINT __attribute__((always_inline)) #define MA_NO_INLINE __attribute__((noinline)) #else #define MA_INLINE MA_GNUC_INLINE_HINT #define MA_NO_INLINE #endif #elif defined(__WATCOMC__) #define MA_INLINE __inline #define MA_NO_INLINE #else #define MA_INLINE #define MA_NO_INLINE #endif /* MA_DLL is not officially supported. You're on your own if you want to use this. */ #if defined(MA_DLL) #if defined(_WIN32) #define MA_DLL_IMPORT __declspec(dllimport) #define MA_DLL_EXPORT __declspec(dllexport) #define MA_DLL_PRIVATE static #else #if defined(__GNUC__) && __GNUC__ >= 4 #define MA_DLL_IMPORT __attribute__((visibility("default"))) #define MA_DLL_EXPORT __attribute__((visibility("default"))) #define MA_DLL_PRIVATE __attribute__((visibility("hidden"))) #else #define MA_DLL_IMPORT #define MA_DLL_EXPORT #define MA_DLL_PRIVATE static #endif #endif #endif #if !defined(MA_API) #if defined(MA_DLL) #if defined(MINIAUDIO_IMPLEMENTATION) || defined(MA_IMPLEMENTATION) #define MA_API MA_DLL_EXPORT #else #define MA_API MA_DLL_IMPORT #endif #else #define MA_API extern #endif #endif #if !defined(MA_STATIC) #if defined(MA_DLL) #define MA_PRIVATE MA_DLL_PRIVATE #else #define MA_PRIVATE static #endif #endif /* SIMD alignment in bytes. Currently set to 32 bytes in preparation for future AVX optimizations. */ #define MA_SIMD_ALIGNMENT 32 /* Special wchar_t type to ensure any structures in the public sections that reference it have a consistent size across all platforms. On Windows, wchar_t is 2 bytes, whereas everywhere else it's 4 bytes. Since Windows likes to use wchar_t for its IDs, we need a special explicitly sized wchar type that is always 2 bytes on all platforms. */ #if !defined(MA_POSIX) && defined(MA_WIN32) typedef wchar_t ma_wchar_win32; #else typedef ma_uint16 ma_wchar_win32; #endif /* Logging Levels ============== Log levels are only used to give logging callbacks some context as to the severity of a log message so they can do filtering. All log levels will be posted to registered logging callbacks. If you don't want to output a certain log level you can discriminate against the log level in the callback. MA_LOG_LEVEL_DEBUG Used for debugging. Useful for debug and test builds, but should be disabled in release builds. MA_LOG_LEVEL_INFO Informational logging. Useful for debugging. This will never be called from within the data callback. MA_LOG_LEVEL_WARNING Warnings. You should enable this in you development builds and action them when encountered. These logs usually indicate a potential problem or misconfiguration, but still allow you to keep running. This will never be called from within the data callback. MA_LOG_LEVEL_ERROR Error logging. This will be fired when an operation fails and is subsequently aborted. This can be fired from within the data callback, in which case the device will be stopped. You should always have this log level enabled. */ typedef enum { MA_LOG_LEVEL_DEBUG = 4, MA_LOG_LEVEL_INFO = 3, MA_LOG_LEVEL_WARNING = 2, MA_LOG_LEVEL_ERROR = 1 } ma_log_level; /* Variables needing to be accessed atomically should be declared with this macro for two reasons: 1) It allows people who read the code to identify a variable as such; and 2) It forces alignment on platforms where it's required or optimal. Note that for x86/64, alignment is not strictly necessary, but does have some performance implications. Where supported by the compiler, alignment will be used, but otherwise if the CPU architecture does not require it, it will simply leave it unaligned. This is the case with old versions of Visual Studio, which I've confirmed with at least VC6. */ #if !defined(_MSC_VER) && defined (__STDC_VERSION__) && (__STDC_VERSION__ >= 201112L) #include <stdalign.h> #define MA_ATOMIC(alignment, type) _Alignas(alignment) type #else #if defined(__GNUC__) /* GCC-style compilers. */ #define MA_ATOMIC(alignment, type) type __attribute__((aligned(alignment))) #elif defined(_MSC_VER) && _MSC_VER > 1200 /* 1200 = VC6. Alignment not supported, but not necessary because x86 is the only supported target. */ /* MSVC. */ #define MA_ATOMIC(alignment, type) __declspec(align(alignment)) type #else /* Other compilers. */ #define MA_ATOMIC(alignment, type) type #endif #endif typedef struct ma_context ma_context; typedef struct ma_device ma_device; typedef ma_uint8 ma_channel; typedef enum { MA_CHANNEL_NONE = 0, MA_CHANNEL_MONO = 1, MA_CHANNEL_FRONT_LEFT = 2, MA_CHANNEL_FRONT_RIGHT = 3, MA_CHANNEL_FRONT_CENTER = 4, MA_CHANNEL_LFE = 5, MA_CHANNEL_BACK_LEFT = 6, MA_CHANNEL_BACK_RIGHT = 7, MA_CHANNEL_FRONT_LEFT_CENTER = 8, MA_CHANNEL_FRONT_RIGHT_CENTER = 9, MA_CHANNEL_BACK_CENTER = 10, MA_CHANNEL_SIDE_LEFT = 11, MA_CHANNEL_SIDE_RIGHT = 12, MA_CHANNEL_TOP_CENTER = 13, MA_CHANNEL_TOP_FRONT_LEFT = 14, MA_CHANNEL_TOP_FRONT_CENTER = 15, MA_CHANNEL_TOP_FRONT_RIGHT = 16, MA_CHANNEL_TOP_BACK_LEFT = 17, MA_CHANNEL_TOP_BACK_CENTER = 18, MA_CHANNEL_TOP_BACK_RIGHT = 19, MA_CHANNEL_AUX_0 = 20, MA_CHANNEL_AUX_1 = 21, MA_CHANNEL_AUX_2 = 22, MA_CHANNEL_AUX_3 = 23, MA_CHANNEL_AUX_4 = 24, MA_CHANNEL_AUX_5 = 25, MA_CHANNEL_AUX_6 = 26, MA_CHANNEL_AUX_7 = 27, MA_CHANNEL_AUX_8 = 28, MA_CHANNEL_AUX_9 = 29, MA_CHANNEL_AUX_10 = 30, MA_CHANNEL_AUX_11 = 31, MA_CHANNEL_AUX_12 = 32, MA_CHANNEL_AUX_13 = 33, MA_CHANNEL_AUX_14 = 34, MA_CHANNEL_AUX_15 = 35, MA_CHANNEL_AUX_16 = 36, MA_CHANNEL_AUX_17 = 37, MA_CHANNEL_AUX_18 = 38, MA_CHANNEL_AUX_19 = 39, MA_CHANNEL_AUX_20 = 40, MA_CHANNEL_AUX_21 = 41, MA_CHANNEL_AUX_22 = 42, MA_CHANNEL_AUX_23 = 43, MA_CHANNEL_AUX_24 = 44, MA_CHANNEL_AUX_25 = 45, MA_CHANNEL_AUX_26 = 46, MA_CHANNEL_AUX_27 = 47, MA_CHANNEL_AUX_28 = 48, MA_CHANNEL_AUX_29 = 49, MA_CHANNEL_AUX_30 = 50, MA_CHANNEL_AUX_31 = 51, /* Count. */ MA_CHANNEL_POSITION_COUNT, /* Aliases. */ MA_CHANNEL_LEFT = MA_CHANNEL_FRONT_LEFT, MA_CHANNEL_RIGHT = MA_CHANNEL_FRONT_RIGHT, } _ma_channel_position; /* Do not use `_ma_channel_position` directly. Use `ma_channel` instead. */ typedef enum { MA_SUCCESS = 0, MA_ERROR = -1, /* A generic error. */ MA_INVALID_ARGS = -2, MA_INVALID_OPERATION = -3, MA_OUT_OF_MEMORY = -4, MA_OUT_OF_RANGE = -5, MA_ACCESS_DENIED = -6, MA_DOES_NOT_EXIST = -7, MA_ALREADY_EXISTS = -8, MA_TOO_MANY_OPEN_FILES = -9, MA_INVALID_FILE = -10, MA_TOO_BIG = -11, MA_PATH_TOO_LONG = -12, MA_NAME_TOO_LONG = -13, MA_NOT_DIRECTORY = -14, MA_IS_DIRECTORY = -15, MA_DIRECTORY_NOT_EMPTY = -16, MA_AT_END = -17, MA_NO_SPACE = -18, MA_BUSY = -19, MA_IO_ERROR = -20, MA_INTERRUPT = -21, MA_UNAVAILABLE = -22, MA_ALREADY_IN_USE = -23, MA_BAD_ADDRESS = -24, MA_BAD_SEEK = -25, MA_BAD_PIPE = -26, MA_DEADLOCK = -27, MA_TOO_MANY_LINKS = -28, MA_NOT_IMPLEMENTED = -29, MA_NO_MESSAGE = -30, MA_BAD_MESSAGE = -31, MA_NO_DATA_AVAILABLE = -32, MA_INVALID_DATA = -33, MA_TIMEOUT = -34, MA_NO_NETWORK = -35, MA_NOT_UNIQUE = -36, MA_NOT_SOCKET = -37, MA_NO_ADDRESS = -38, MA_BAD_PROTOCOL = -39, MA_PROTOCOL_UNAVAILABLE = -40, MA_PROTOCOL_NOT_SUPPORTED = -41, MA_PROTOCOL_FAMILY_NOT_SUPPORTED = -42, MA_ADDRESS_FAMILY_NOT_SUPPORTED = -43, MA_SOCKET_NOT_SUPPORTED = -44, MA_CONNECTION_RESET = -45, MA_ALREADY_CONNECTED = -46, MA_NOT_CONNECTED = -47, MA_CONNECTION_REFUSED = -48, MA_NO_HOST = -49, MA_IN_PROGRESS = -50, MA_CANCELLED = -51, MA_MEMORY_ALREADY_MAPPED = -52, /* General non-standard errors. */ MA_CRC_MISMATCH = -100, /* General miniaudio-specific errors. */ MA_FORMAT_NOT_SUPPORTED = -200, MA_DEVICE_TYPE_NOT_SUPPORTED = -201, MA_SHARE_MODE_NOT_SUPPORTED = -202, MA_NO_BACKEND = -203, MA_NO_DEVICE = -204, MA_API_NOT_FOUND = -205, MA_INVALID_DEVICE_CONFIG = -206, MA_LOOP = -207, MA_BACKEND_NOT_ENABLED = -208, /* State errors. */ MA_DEVICE_NOT_INITIALIZED = -300, MA_DEVICE_ALREADY_INITIALIZED = -301, MA_DEVICE_NOT_STARTED = -302, MA_DEVICE_NOT_STOPPED = -303, /* Operation errors. */ MA_FAILED_TO_INIT_BACKEND = -400, MA_FAILED_TO_OPEN_BACKEND_DEVICE = -401, MA_FAILED_TO_START_BACKEND_DEVICE = -402, MA_FAILED_TO_STOP_BACKEND_DEVICE = -403 } ma_result; #define MA_MIN_CHANNELS 1 #ifndef MA_MAX_CHANNELS #define MA_MAX_CHANNELS 254 #endif #ifndef MA_MAX_FILTER_ORDER #define MA_MAX_FILTER_ORDER 8 #endif typedef enum { ma_stream_format_pcm = 0 } ma_stream_format; typedef enum { ma_stream_layout_interleaved = 0, ma_stream_layout_deinterleaved } ma_stream_layout; typedef enum { ma_dither_mode_none = 0, ma_dither_mode_rectangle, ma_dither_mode_triangle } ma_dither_mode; typedef enum { /* I like to keep these explicitly defined because they're used as a key into a lookup table. When items are added to this, make sure there are no gaps and that they're added to the lookup table in ma_get_bytes_per_sample(). */ ma_format_unknown = 0, /* Mainly used for indicating an error, but also used as the default for the output format for decoders. */ ma_format_u8 = 1, ma_format_s16 = 2, /* Seems to be the most widely supported format. */ ma_format_s24 = 3, /* Tightly packed. 3 bytes per sample. */ ma_format_s32 = 4, ma_format_f32 = 5, ma_format_count } ma_format; typedef enum { /* Standard rates need to be in priority order. */ ma_standard_sample_rate_48000 = 48000, /* Most common */ ma_standard_sample_rate_44100 = 44100, ma_standard_sample_rate_32000 = 32000, /* Lows */ ma_standard_sample_rate_24000 = 24000, ma_standard_sample_rate_22050 = 22050, ma_standard_sample_rate_88200 = 88200, /* Highs */ ma_standard_sample_rate_96000 = 96000, ma_standard_sample_rate_176400 = 176400, ma_standard_sample_rate_192000 = 192000, ma_standard_sample_rate_16000 = 16000, /* Extreme lows */ ma_standard_sample_rate_11025 = 11025, ma_standard_sample_rate_8000 = 8000, ma_standard_sample_rate_352800 = 352800, /* Extreme highs */ ma_standard_sample_rate_384000 = 384000, ma_standard_sample_rate_min = ma_standard_sample_rate_8000, ma_standard_sample_rate_max = ma_standard_sample_rate_384000, ma_standard_sample_rate_count = 14 /* Need to maintain the count manually. Make sure this is updated if items are added to enum. */ } ma_standard_sample_rate; typedef enum { ma_channel_mix_mode_rectangular = 0, /* Simple averaging based on the plane(s) the channel is sitting on. */ ma_channel_mix_mode_simple, /* Drop excess channels; zeroed out extra channels. */ ma_channel_mix_mode_custom_weights, /* Use custom weights specified in ma_channel_converter_config. */ ma_channel_mix_mode_default = ma_channel_mix_mode_rectangular } ma_channel_mix_mode; typedef enum { ma_standard_channel_map_microsoft, ma_standard_channel_map_alsa, ma_standard_channel_map_rfc3551, /* Based off AIFF. */ ma_standard_channel_map_flac, ma_standard_channel_map_vorbis, ma_standard_channel_map_sound4, /* FreeBSD's sound(4). */ ma_standard_channel_map_sndio, /* www.sndio.org/tips.html */ ma_standard_channel_map_webaudio = ma_standard_channel_map_flac, /* https://webaudio.github.io/web-audio-api/#ChannelOrdering. Only 1, 2, 4 and 6 channels are defined, but can fill in the gaps with logical assumptions. */ ma_standard_channel_map_default = ma_standard_channel_map_microsoft } ma_standard_channel_map; typedef enum { ma_performance_profile_low_latency = 0, ma_performance_profile_conservative } ma_performance_profile; typedef struct { void* pUserData; void* (* onMalloc)(size_t sz, void* pUserData); void* (* onRealloc)(void* p, size_t sz, void* pUserData); void (* onFree)(void* p, void* pUserData); } ma_allocation_callbacks; typedef struct { ma_uint32 state; } ma_lcg; /* Atomics. These are typesafe structures to prevent errors as a result of forgetting to reference variables atomically. It's too easy to introduce subtle bugs where you accidentally do a regular assignment instead of an atomic load/store, etc. By using a struct we can enforce the use of atomics at compile time. These types are declared in the header section because we need to reference them in structs below, but functions for using them are only exposed in the implementation section. I do not want these to be part of the public API. There's a few downsides to this system. The first is that you need to declare a new struct for each type. Below are some macros to help with the declarations. They will be named like so: ma_atomic_uint32 - atomic ma_uint32 ma_atomic_int32 - atomic ma_int32 ma_atomic_uint64 - atomic ma_uint64 ma_atomic_float - atomic float ma_atomic_bool32 - atomic ma_bool32 The other downside is that atomic pointers are extremely messy. You need to declare a new struct for each specific type of pointer you need to make atomic. For example, an atomic ma_node* will look like this: MA_ATOMIC_SAFE_TYPE_IMPL_PTR(node) Which will declare a type struct that's named like so: ma_atomic_ptr_node Functions to use the atomic types are declared in the implementation section. All atomic functions are prefixed with the name of the struct. For example: ma_atomic_uint32_set() - Atomic store of ma_uint32 ma_atomic_uint32_get() - Atomic load of ma_uint32 etc. For pointer types it's the same, which makes them a bit messy to use due to the length of each function name, but in return you get type safety and enforcement of atomic operations. */ #define MA_ATOMIC_SAFE_TYPE_DECL(c89TypeExtension, typeSize, type) \ typedef struct \ { \ MA_ATOMIC(typeSize, ma_##type) value; \ } ma_atomic_##type; \ #define MA_ATOMIC_SAFE_TYPE_DECL_PTR(type) \ typedef struct \ { \ MA_ATOMIC(MA_SIZEOF_PTR, ma_##type*) value; \ } ma_atomic_ptr_##type; \ MA_ATOMIC_SAFE_TYPE_DECL(32, 4, uint32) MA_ATOMIC_SAFE_TYPE_DECL(i32, 4, int32) MA_ATOMIC_SAFE_TYPE_DECL(64, 8, uint64) MA_ATOMIC_SAFE_TYPE_DECL(f32, 4, float) MA_ATOMIC_SAFE_TYPE_DECL(32, 4, bool32) /* Spinlocks are 32-bit for compatibility reasons. */ typedef ma_uint32 ma_spinlock; #ifndef MA_NO_THREADING /* Thread priorities should be ordered such that the default priority of the worker thread is 0. */ typedef enum { ma_thread_priority_idle = -5, ma_thread_priority_lowest = -4, ma_thread_priority_low = -3, ma_thread_priority_normal = -2, ma_thread_priority_high = -1, ma_thread_priority_highest = 0, ma_thread_priority_realtime = 1, ma_thread_priority_default = 0 } ma_thread_priority; #if defined(MA_POSIX) typedef ma_pthread_t ma_thread; #elif defined(MA_WIN32) typedef ma_handle ma_thread; #endif #if defined(MA_POSIX) typedef ma_pthread_mutex_t ma_mutex; #elif defined(MA_WIN32) typedef ma_handle ma_mutex; #endif #if defined(MA_POSIX) typedef struct { ma_uint32 value; ma_pthread_mutex_t lock; ma_pthread_cond_t cond; } ma_event; #elif defined(MA_WIN32) typedef ma_handle ma_event; #endif #if defined(MA_POSIX) typedef struct { int value; ma_pthread_mutex_t lock; ma_pthread_cond_t cond; } ma_semaphore; #elif defined(MA_WIN32) typedef ma_handle ma_semaphore; #endif #else /* MA_NO_THREADING is set which means threading is disabled. Threading is required by some API families. If any of these are enabled we need to throw an error. */ #ifndef MA_NO_DEVICE_IO #error "MA_NO_THREADING cannot be used without MA_NO_DEVICE_IO"; #endif #endif /* MA_NO_THREADING */ /* Retrieves the version of miniaudio as separated integers. Each component can be NULL if it's not required. */ MA_API void ma_version(ma_uint32* pMajor, ma_uint32* pMinor, ma_uint32* pRevision); /* Retrieves the version of miniaudio as a string which can be useful for logging purposes. */ MA_API const char* ma_version_string(void); /************************************************************************************************************************************************************** Logging **************************************************************************************************************************************************************/ #include <stdarg.h> /* For va_list. */ #if defined(__has_attribute) #if __has_attribute(format) #define MA_ATTRIBUTE_FORMAT(fmt, va) __attribute__((format(printf, fmt, va))) #endif #endif #ifndef MA_ATTRIBUTE_FORMAT #define MA_ATTRIBUTE_FORMAT(fmt, va) #endif #ifndef MA_MAX_LOG_CALLBACKS #define MA_MAX_LOG_CALLBACKS 4 #endif /* The callback for handling log messages. Parameters ---------- pUserData (in) The user data pointer that was passed into ma_log_register_callback(). logLevel (in) The log level. This can be one of the following: +----------------------+ | Log Level | +----------------------+ | MA_LOG_LEVEL_DEBUG | | MA_LOG_LEVEL_INFO | | MA_LOG_LEVEL_WARNING | | MA_LOG_LEVEL_ERROR | +----------------------+ pMessage (in) The log message. */ typedef void (* ma_log_callback_proc)(void* pUserData, ma_uint32 level, const char* pMessage); typedef struct { ma_log_callback_proc onLog; void* pUserData; } ma_log_callback; MA_API ma_log_callback ma_log_callback_init(ma_log_callback_proc onLog, void* pUserData); typedef struct { ma_log_callback callbacks[MA_MAX_LOG_CALLBACKS]; ma_uint32 callbackCount; ma_allocation_callbacks allocationCallbacks; /* Need to store these persistently because ma_log_postv() might need to allocate a buffer on the heap. */ #ifndef MA_NO_THREADING ma_mutex lock; /* For thread safety just to make it easier and safer for the logging implementation. */ #endif } ma_log; MA_API ma_result ma_log_init(const ma_allocation_callbacks* pAllocationCallbacks, ma_log* pLog); MA_API void ma_log_uninit(ma_log* pLog); MA_API ma_result ma_log_register_callback(ma_log* pLog, ma_log_callback callback); MA_API ma_result ma_log_unregister_callback(ma_log* pLog, ma_log_callback callback); MA_API ma_result ma_log_post(ma_log* pLog, ma_uint32 level, const char* pMessage); MA_API ma_result ma_log_postv(ma_log* pLog, ma_uint32 level, const char* pFormat, va_list args); MA_API ma_result ma_log_postf(ma_log* pLog, ma_uint32 level, const char* pFormat, ...) MA_ATTRIBUTE_FORMAT(3, 4); /************************************************************************************************************************************************************** Biquad Filtering **************************************************************************************************************************************************************/ typedef union { float f32; ma_int32 s32; } ma_biquad_coefficient; typedef struct { ma_format format; ma_uint32 channels; double b0; double b1; double b2; double a0; double a1; double a2; } ma_biquad_config; MA_API ma_biquad_config ma_biquad_config_init(ma_format format, ma_uint32 channels, double b0, double b1, double b2, double a0, double a1, double a2); typedef struct { ma_format format; ma_uint32 channels; ma_biquad_coefficient b0; ma_biquad_coefficient b1; ma_biquad_coefficient b2; ma_biquad_coefficient a1; ma_biquad_coefficient a2; ma_biquad_coefficient* pR1; ma_biquad_coefficient* pR2; /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_biquad; MA_API ma_result ma_biquad_get_heap_size(const ma_biquad_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_biquad_init_preallocated(const ma_biquad_config* pConfig, void* pHeap, ma_biquad* pBQ); MA_API ma_result ma_biquad_init(const ma_biquad_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_biquad* pBQ); MA_API void ma_biquad_uninit(ma_biquad* pBQ, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_biquad_reinit(const ma_biquad_config* pConfig, ma_biquad* pBQ); MA_API ma_result ma_biquad_clear_cache(ma_biquad* pBQ); MA_API ma_result ma_biquad_process_pcm_frames(ma_biquad* pBQ, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_biquad_get_latency(const ma_biquad* pBQ); /************************************************************************************************************************************************************** Low-Pass Filtering **************************************************************************************************************************************************************/ typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; double cutoffFrequency; double q; } ma_lpf1_config, ma_lpf2_config; MA_API ma_lpf1_config ma_lpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency); MA_API ma_lpf2_config ma_lpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q); typedef struct { ma_format format; ma_uint32 channels; ma_biquad_coefficient a; ma_biquad_coefficient* pR1; /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_lpf1; MA_API ma_result ma_lpf1_get_heap_size(const ma_lpf1_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_lpf1_init_preallocated(const ma_lpf1_config* pConfig, void* pHeap, ma_lpf1* pLPF); MA_API ma_result ma_lpf1_init(const ma_lpf1_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf1* pLPF); MA_API void ma_lpf1_uninit(ma_lpf1* pLPF, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_lpf1_reinit(const ma_lpf1_config* pConfig, ma_lpf1* pLPF); MA_API ma_result ma_lpf1_clear_cache(ma_lpf1* pLPF); MA_API ma_result ma_lpf1_process_pcm_frames(ma_lpf1* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_lpf1_get_latency(const ma_lpf1* pLPF); typedef struct { ma_biquad bq; /* The second order low-pass filter is implemented as a biquad filter. */ } ma_lpf2; MA_API ma_result ma_lpf2_get_heap_size(const ma_lpf2_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_lpf2_init_preallocated(const ma_lpf2_config* pConfig, void* pHeap, ma_lpf2* pHPF); MA_API ma_result ma_lpf2_init(const ma_lpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf2* pLPF); MA_API void ma_lpf2_uninit(ma_lpf2* pLPF, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_lpf2_reinit(const ma_lpf2_config* pConfig, ma_lpf2* pLPF); MA_API ma_result ma_lpf2_clear_cache(ma_lpf2* pLPF); MA_API ma_result ma_lpf2_process_pcm_frames(ma_lpf2* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_lpf2_get_latency(const ma_lpf2* pLPF); typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; double cutoffFrequency; ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */ } ma_lpf_config; MA_API ma_lpf_config ma_lpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order); typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; ma_uint32 lpf1Count; ma_uint32 lpf2Count; ma_lpf1* pLPF1; ma_lpf2* pLPF2; /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_lpf; MA_API ma_result ma_lpf_get_heap_size(const ma_lpf_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_lpf_init_preallocated(const ma_lpf_config* pConfig, void* pHeap, ma_lpf* pLPF); MA_API ma_result ma_lpf_init(const ma_lpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf* pLPF); MA_API void ma_lpf_uninit(ma_lpf* pLPF, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_lpf_reinit(const ma_lpf_config* pConfig, ma_lpf* pLPF); MA_API ma_result ma_lpf_clear_cache(ma_lpf* pLPF); MA_API ma_result ma_lpf_process_pcm_frames(ma_lpf* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_lpf_get_latency(const ma_lpf* pLPF); /************************************************************************************************************************************************************** High-Pass Filtering **************************************************************************************************************************************************************/ typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; double cutoffFrequency; double q; } ma_hpf1_config, ma_hpf2_config; MA_API ma_hpf1_config ma_hpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency); MA_API ma_hpf2_config ma_hpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q); typedef struct { ma_format format; ma_uint32 channels; ma_biquad_coefficient a; ma_biquad_coefficient* pR1; /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_hpf1; MA_API ma_result ma_hpf1_get_heap_size(const ma_hpf1_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_hpf1_init_preallocated(const ma_hpf1_config* pConfig, void* pHeap, ma_hpf1* pLPF); MA_API ma_result ma_hpf1_init(const ma_hpf1_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf1* pHPF); MA_API void ma_hpf1_uninit(ma_hpf1* pHPF, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_hpf1_reinit(const ma_hpf1_config* pConfig, ma_hpf1* pHPF); MA_API ma_result ma_hpf1_process_pcm_frames(ma_hpf1* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_hpf1_get_latency(const ma_hpf1* pHPF); typedef struct { ma_biquad bq; /* The second order high-pass filter is implemented as a biquad filter. */ } ma_hpf2; MA_API ma_result ma_hpf2_get_heap_size(const ma_hpf2_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_hpf2_init_preallocated(const ma_hpf2_config* pConfig, void* pHeap, ma_hpf2* pHPF); MA_API ma_result ma_hpf2_init(const ma_hpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf2* pHPF); MA_API void ma_hpf2_uninit(ma_hpf2* pHPF, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_hpf2_reinit(const ma_hpf2_config* pConfig, ma_hpf2* pHPF); MA_API ma_result ma_hpf2_process_pcm_frames(ma_hpf2* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_hpf2_get_latency(const ma_hpf2* pHPF); typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; double cutoffFrequency; ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */ } ma_hpf_config; MA_API ma_hpf_config ma_hpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order); typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; ma_uint32 hpf1Count; ma_uint32 hpf2Count; ma_hpf1* pHPF1; ma_hpf2* pHPF2; /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_hpf; MA_API ma_result ma_hpf_get_heap_size(const ma_hpf_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_hpf_init_preallocated(const ma_hpf_config* pConfig, void* pHeap, ma_hpf* pLPF); MA_API ma_result ma_hpf_init(const ma_hpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf* pHPF); MA_API void ma_hpf_uninit(ma_hpf* pHPF, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_hpf_reinit(const ma_hpf_config* pConfig, ma_hpf* pHPF); MA_API ma_result ma_hpf_process_pcm_frames(ma_hpf* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_hpf_get_latency(const ma_hpf* pHPF); /************************************************************************************************************************************************************** Band-Pass Filtering **************************************************************************************************************************************************************/ typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; double cutoffFrequency; double q; } ma_bpf2_config; MA_API ma_bpf2_config ma_bpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q); typedef struct { ma_biquad bq; /* The second order band-pass filter is implemented as a biquad filter. */ } ma_bpf2; MA_API ma_result ma_bpf2_get_heap_size(const ma_bpf2_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_bpf2_init_preallocated(const ma_bpf2_config* pConfig, void* pHeap, ma_bpf2* pBPF); MA_API ma_result ma_bpf2_init(const ma_bpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf2* pBPF); MA_API void ma_bpf2_uninit(ma_bpf2* pBPF, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_bpf2_reinit(const ma_bpf2_config* pConfig, ma_bpf2* pBPF); MA_API ma_result ma_bpf2_process_pcm_frames(ma_bpf2* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_bpf2_get_latency(const ma_bpf2* pBPF); typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; double cutoffFrequency; ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */ } ma_bpf_config; MA_API ma_bpf_config ma_bpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order); typedef struct { ma_format format; ma_uint32 channels; ma_uint32 bpf2Count; ma_bpf2* pBPF2; /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_bpf; MA_API ma_result ma_bpf_get_heap_size(const ma_bpf_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_bpf_init_preallocated(const ma_bpf_config* pConfig, void* pHeap, ma_bpf* pBPF); MA_API ma_result ma_bpf_init(const ma_bpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf* pBPF); MA_API void ma_bpf_uninit(ma_bpf* pBPF, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_bpf_reinit(const ma_bpf_config* pConfig, ma_bpf* pBPF); MA_API ma_result ma_bpf_process_pcm_frames(ma_bpf* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_bpf_get_latency(const ma_bpf* pBPF); /************************************************************************************************************************************************************** Notching Filter **************************************************************************************************************************************************************/ typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; double q; double frequency; } ma_notch2_config, ma_notch_config; MA_API ma_notch2_config ma_notch2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double q, double frequency); typedef struct { ma_biquad bq; } ma_notch2; MA_API ma_result ma_notch2_get_heap_size(const ma_notch2_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_notch2_init_preallocated(const ma_notch2_config* pConfig, void* pHeap, ma_notch2* pFilter); MA_API ma_result ma_notch2_init(const ma_notch2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_notch2* pFilter); MA_API void ma_notch2_uninit(ma_notch2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_notch2_reinit(const ma_notch2_config* pConfig, ma_notch2* pFilter); MA_API ma_result ma_notch2_process_pcm_frames(ma_notch2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_notch2_get_latency(const ma_notch2* pFilter); /************************************************************************************************************************************************************** Peaking EQ Filter **************************************************************************************************************************************************************/ typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; double gainDB; double q; double frequency; } ma_peak2_config, ma_peak_config; MA_API ma_peak2_config ma_peak2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency); typedef struct { ma_biquad bq; } ma_peak2; MA_API ma_result ma_peak2_get_heap_size(const ma_peak2_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_peak2_init_preallocated(const ma_peak2_config* pConfig, void* pHeap, ma_peak2* pFilter); MA_API ma_result ma_peak2_init(const ma_peak2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_peak2* pFilter); MA_API void ma_peak2_uninit(ma_peak2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_peak2_reinit(const ma_peak2_config* pConfig, ma_peak2* pFilter); MA_API ma_result ma_peak2_process_pcm_frames(ma_peak2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_peak2_get_latency(const ma_peak2* pFilter); /************************************************************************************************************************************************************** Low Shelf Filter **************************************************************************************************************************************************************/ typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; double gainDB; double shelfSlope; double frequency; } ma_loshelf2_config, ma_loshelf_config; MA_API ma_loshelf2_config ma_loshelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency); typedef struct { ma_biquad bq; } ma_loshelf2; MA_API ma_result ma_loshelf2_get_heap_size(const ma_loshelf2_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_loshelf2_init_preallocated(const ma_loshelf2_config* pConfig, void* pHeap, ma_loshelf2* pFilter); MA_API ma_result ma_loshelf2_init(const ma_loshelf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_loshelf2* pFilter); MA_API void ma_loshelf2_uninit(ma_loshelf2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_loshelf2_reinit(const ma_loshelf2_config* pConfig, ma_loshelf2* pFilter); MA_API ma_result ma_loshelf2_process_pcm_frames(ma_loshelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_loshelf2_get_latency(const ma_loshelf2* pFilter); /************************************************************************************************************************************************************** High Shelf Filter **************************************************************************************************************************************************************/ typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; double gainDB; double shelfSlope; double frequency; } ma_hishelf2_config, ma_hishelf_config; MA_API ma_hishelf2_config ma_hishelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency); typedef struct { ma_biquad bq; } ma_hishelf2; MA_API ma_result ma_hishelf2_get_heap_size(const ma_hishelf2_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_hishelf2_init_preallocated(const ma_hishelf2_config* pConfig, void* pHeap, ma_hishelf2* pFilter); MA_API ma_result ma_hishelf2_init(const ma_hishelf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hishelf2* pFilter); MA_API void ma_hishelf2_uninit(ma_hishelf2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_hishelf2_reinit(const ma_hishelf2_config* pConfig, ma_hishelf2* pFilter); MA_API ma_result ma_hishelf2_process_pcm_frames(ma_hishelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_uint32 ma_hishelf2_get_latency(const ma_hishelf2* pFilter); /* Delay */ typedef struct { ma_uint32 channels; ma_uint32 sampleRate; ma_uint32 delayInFrames; ma_bool32 delayStart; /* Set to true to delay the start of the output; false otherwise. */ float wet; /* 0..1. Default = 1. */ float dry; /* 0..1. Default = 1. */ float decay; /* 0..1. Default = 0 (no feedback). Feedback decay. Use this for echo. */ } ma_delay_config; MA_API ma_delay_config ma_delay_config_init(ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 delayInFrames, float decay); typedef struct { ma_delay_config config; ma_uint32 cursor; /* Feedback is written to this cursor. Always equal or in front of the read cursor. */ ma_uint32 bufferSizeInFrames; float* pBuffer; } ma_delay; MA_API ma_result ma_delay_init(const ma_delay_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_delay* pDelay); MA_API void ma_delay_uninit(ma_delay* pDelay, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_delay_process_pcm_frames(ma_delay* pDelay, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount); MA_API void ma_delay_set_wet(ma_delay* pDelay, float value); MA_API float ma_delay_get_wet(const ma_delay* pDelay); MA_API void ma_delay_set_dry(ma_delay* pDelay, float value); MA_API float ma_delay_get_dry(const ma_delay* pDelay); MA_API void ma_delay_set_decay(ma_delay* pDelay, float value); MA_API float ma_delay_get_decay(const ma_delay* pDelay); /* Gainer for smooth volume changes. */ typedef struct { ma_uint32 channels; ma_uint32 smoothTimeInFrames; } ma_gainer_config; MA_API ma_gainer_config ma_gainer_config_init(ma_uint32 channels, ma_uint32 smoothTimeInFrames); typedef struct { ma_gainer_config config; ma_uint32 t; float masterVolume; float* pOldGains; float* pNewGains; /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_gainer; MA_API ma_result ma_gainer_get_heap_size(const ma_gainer_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_gainer_init_preallocated(const ma_gainer_config* pConfig, void* pHeap, ma_gainer* pGainer); MA_API ma_result ma_gainer_init(const ma_gainer_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_gainer* pGainer); MA_API void ma_gainer_uninit(ma_gainer* pGainer, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_gainer_process_pcm_frames(ma_gainer* pGainer, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_result ma_gainer_set_gain(ma_gainer* pGainer, float newGain); MA_API ma_result ma_gainer_set_gains(ma_gainer* pGainer, float* pNewGains); MA_API ma_result ma_gainer_set_master_volume(ma_gainer* pGainer, float volume); MA_API ma_result ma_gainer_get_master_volume(const ma_gainer* pGainer, float* pVolume); /* Stereo panner. */ typedef enum { ma_pan_mode_balance = 0, /* Does not blend one side with the other. Technically just a balance. Compatible with other popular audio engines and therefore the default. */ ma_pan_mode_pan /* A true pan. The sound from one side will "move" to the other side and blend with it. */ } ma_pan_mode; typedef struct { ma_format format; ma_uint32 channels; ma_pan_mode mode; float pan; } ma_panner_config; MA_API ma_panner_config ma_panner_config_init(ma_format format, ma_uint32 channels); typedef struct { ma_format format; ma_uint32 channels; ma_pan_mode mode; float pan; /* -1..1 where 0 is no pan, -1 is left side, +1 is right side. Defaults to 0. */ } ma_panner; MA_API ma_result ma_panner_init(const ma_panner_config* pConfig, ma_panner* pPanner); MA_API ma_result ma_panner_process_pcm_frames(ma_panner* pPanner, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API void ma_panner_set_mode(ma_panner* pPanner, ma_pan_mode mode); MA_API ma_pan_mode ma_panner_get_mode(const ma_panner* pPanner); MA_API void ma_panner_set_pan(ma_panner* pPanner, float pan); MA_API float ma_panner_get_pan(const ma_panner* pPanner); /* Fader. */ typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; } ma_fader_config; MA_API ma_fader_config ma_fader_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate); typedef struct { ma_fader_config config; float volumeBeg; /* If volumeBeg and volumeEnd is equal to 1, no fading happens (ma_fader_process_pcm_frames() will run as a passthrough). */ float volumeEnd; ma_uint64 lengthInFrames; /* The total length of the fade. */ ma_int64 cursorInFrames; /* The current time in frames. Incremented by ma_fader_process_pcm_frames(). Signed because it'll be offset by startOffsetInFrames in set_fade_ex(). */ } ma_fader; MA_API ma_result ma_fader_init(const ma_fader_config* pConfig, ma_fader* pFader); MA_API ma_result ma_fader_process_pcm_frames(ma_fader* pFader, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API void ma_fader_get_data_format(const ma_fader* pFader, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate); MA_API void ma_fader_set_fade(ma_fader* pFader, float volumeBeg, float volumeEnd, ma_uint64 lengthInFrames); MA_API void ma_fader_set_fade_ex(ma_fader* pFader, float volumeBeg, float volumeEnd, ma_uint64 lengthInFrames, ma_int64 startOffsetInFrames); MA_API float ma_fader_get_current_volume(const ma_fader* pFader); /* Spatializer. */ typedef struct { float x; float y; float z; } ma_vec3f; typedef struct { ma_vec3f v; ma_spinlock lock; } ma_atomic_vec3f; typedef enum { ma_attenuation_model_none, /* No distance attenuation and no spatialization. */ ma_attenuation_model_inverse, /* Equivalent to OpenAL's AL_INVERSE_DISTANCE_CLAMPED. */ ma_attenuation_model_linear, /* Linear attenuation. Equivalent to OpenAL's AL_LINEAR_DISTANCE_CLAMPED. */ ma_attenuation_model_exponential /* Exponential attenuation. Equivalent to OpenAL's AL_EXPONENT_DISTANCE_CLAMPED. */ } ma_attenuation_model; typedef enum { ma_positioning_absolute, ma_positioning_relative } ma_positioning; typedef enum { ma_handedness_right, ma_handedness_left } ma_handedness; typedef struct { ma_uint32 channelsOut; ma_channel* pChannelMapOut; ma_handedness handedness; /* Defaults to right. Forward is -1 on the Z axis. In a left handed system, forward is +1 on the Z axis. */ float coneInnerAngleInRadians; float coneOuterAngleInRadians; float coneOuterGain; float speedOfSound; ma_vec3f worldUp; } ma_spatializer_listener_config; MA_API ma_spatializer_listener_config ma_spatializer_listener_config_init(ma_uint32 channelsOut); typedef struct { ma_spatializer_listener_config config; ma_atomic_vec3f position; /* The absolute position of the listener. */ ma_atomic_vec3f direction; /* The direction the listener is facing. The world up vector is config.worldUp. */ ma_atomic_vec3f velocity; ma_bool32 isEnabled; /* Memory management. */ ma_bool32 _ownsHeap; void* _pHeap; } ma_spatializer_listener; MA_API ma_result ma_spatializer_listener_get_heap_size(const ma_spatializer_listener_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_spatializer_listener_init_preallocated(const ma_spatializer_listener_config* pConfig, void* pHeap, ma_spatializer_listener* pListener); MA_API ma_result ma_spatializer_listener_init(const ma_spatializer_listener_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_spatializer_listener* pListener); MA_API void ma_spatializer_listener_uninit(ma_spatializer_listener* pListener, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_channel* ma_spatializer_listener_get_channel_map(ma_spatializer_listener* pListener); MA_API void ma_spatializer_listener_set_cone(ma_spatializer_listener* pListener, float innerAngleInRadians, float outerAngleInRadians, float outerGain); MA_API void ma_spatializer_listener_get_cone(const ma_spatializer_listener* pListener, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain); MA_API void ma_spatializer_listener_set_position(ma_spatializer_listener* pListener, float x, float y, float z); MA_API ma_vec3f ma_spatializer_listener_get_position(const ma_spatializer_listener* pListener); MA_API void ma_spatializer_listener_set_direction(ma_spatializer_listener* pListener, float x, float y, float z); MA_API ma_vec3f ma_spatializer_listener_get_direction(const ma_spatializer_listener* pListener); MA_API void ma_spatializer_listener_set_velocity(ma_spatializer_listener* pListener, float x, float y, float z); MA_API ma_vec3f ma_spatializer_listener_get_velocity(const ma_spatializer_listener* pListener); MA_API void ma_spatializer_listener_set_speed_of_sound(ma_spatializer_listener* pListener, float speedOfSound); MA_API float ma_spatializer_listener_get_speed_of_sound(const ma_spatializer_listener* pListener); MA_API void ma_spatializer_listener_set_world_up(ma_spatializer_listener* pListener, float x, float y, float z); MA_API ma_vec3f ma_spatializer_listener_get_world_up(const ma_spatializer_listener* pListener); MA_API void ma_spatializer_listener_set_enabled(ma_spatializer_listener* pListener, ma_bool32 isEnabled); MA_API ma_bool32 ma_spatializer_listener_is_enabled(const ma_spatializer_listener* pListener); typedef struct { ma_uint32 channelsIn; ma_uint32 channelsOut; ma_channel* pChannelMapIn; ma_attenuation_model attenuationModel; ma_positioning positioning; ma_handedness handedness; /* Defaults to right. Forward is -1 on the Z axis. In a left handed system, forward is +1 on the Z axis. */ float minGain; float maxGain; float minDistance; float maxDistance; float rolloff; float coneInnerAngleInRadians; float coneOuterAngleInRadians; float coneOuterGain; float dopplerFactor; /* Set to 0 to disable doppler effect. */ float directionalAttenuationFactor; /* Set to 0 to disable directional attenuation. */ float minSpatializationChannelGain; /* The minimal scaling factor to apply to channel gains when accounting for the direction of the sound relative to the listener. Must be in the range of 0..1. Smaller values means more aggressive directional panning, larger values means more subtle directional panning. */ ma_uint32 gainSmoothTimeInFrames; /* When the gain of a channel changes during spatialization, the transition will be linearly interpolated over this number of frames. */ } ma_spatializer_config; MA_API ma_spatializer_config ma_spatializer_config_init(ma_uint32 channelsIn, ma_uint32 channelsOut); typedef struct { ma_uint32 channelsIn; ma_uint32 channelsOut; ma_channel* pChannelMapIn; ma_attenuation_model attenuationModel; ma_positioning positioning; ma_handedness handedness; /* Defaults to right. Forward is -1 on the Z axis. In a left handed system, forward is +1 on the Z axis. */ float minGain; float maxGain; float minDistance; float maxDistance; float rolloff; float coneInnerAngleInRadians; float coneOuterAngleInRadians; float coneOuterGain; float dopplerFactor; /* Set to 0 to disable doppler effect. */ float directionalAttenuationFactor; /* Set to 0 to disable directional attenuation. */ ma_uint32 gainSmoothTimeInFrames; /* When the gain of a channel changes during spatialization, the transition will be linearly interpolated over this number of frames. */ ma_atomic_vec3f position; ma_atomic_vec3f direction; ma_atomic_vec3f velocity; /* For doppler effect. */ float dopplerPitch; /* Will be updated by ma_spatializer_process_pcm_frames() and can be used by higher level functions to apply a pitch shift for doppler effect. */ float minSpatializationChannelGain; ma_gainer gainer; /* For smooth gain transitions. */ float* pNewChannelGainsOut; /* An offset of _pHeap. Used by ma_spatializer_process_pcm_frames() to store new channel gains. The number of elements in this array is equal to config.channelsOut. */ /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_spatializer; MA_API ma_result ma_spatializer_get_heap_size(const ma_spatializer_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_spatializer_init_preallocated(const ma_spatializer_config* pConfig, void* pHeap, ma_spatializer* pSpatializer); MA_API ma_result ma_spatializer_init(const ma_spatializer_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_spatializer* pSpatializer); MA_API void ma_spatializer_uninit(ma_spatializer* pSpatializer, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_spatializer_process_pcm_frames(ma_spatializer* pSpatializer, ma_spatializer_listener* pListener, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_result ma_spatializer_set_master_volume(ma_spatializer* pSpatializer, float volume); MA_API ma_result ma_spatializer_get_master_volume(const ma_spatializer* pSpatializer, float* pVolume); MA_API ma_uint32 ma_spatializer_get_input_channels(const ma_spatializer* pSpatializer); MA_API ma_uint32 ma_spatializer_get_output_channels(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_attenuation_model(ma_spatializer* pSpatializer, ma_attenuation_model attenuationModel); MA_API ma_attenuation_model ma_spatializer_get_attenuation_model(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_positioning(ma_spatializer* pSpatializer, ma_positioning positioning); MA_API ma_positioning ma_spatializer_get_positioning(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_rolloff(ma_spatializer* pSpatializer, float rolloff); MA_API float ma_spatializer_get_rolloff(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_min_gain(ma_spatializer* pSpatializer, float minGain); MA_API float ma_spatializer_get_min_gain(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_max_gain(ma_spatializer* pSpatializer, float maxGain); MA_API float ma_spatializer_get_max_gain(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_min_distance(ma_spatializer* pSpatializer, float minDistance); MA_API float ma_spatializer_get_min_distance(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_max_distance(ma_spatializer* pSpatializer, float maxDistance); MA_API float ma_spatializer_get_max_distance(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_cone(ma_spatializer* pSpatializer, float innerAngleInRadians, float outerAngleInRadians, float outerGain); MA_API void ma_spatializer_get_cone(const ma_spatializer* pSpatializer, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain); MA_API void ma_spatializer_set_doppler_factor(ma_spatializer* pSpatializer, float dopplerFactor); MA_API float ma_spatializer_get_doppler_factor(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_directional_attenuation_factor(ma_spatializer* pSpatializer, float directionalAttenuationFactor); MA_API float ma_spatializer_get_directional_attenuation_factor(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_position(ma_spatializer* pSpatializer, float x, float y, float z); MA_API ma_vec3f ma_spatializer_get_position(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_direction(ma_spatializer* pSpatializer, float x, float y, float z); MA_API ma_vec3f ma_spatializer_get_direction(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_set_velocity(ma_spatializer* pSpatializer, float x, float y, float z); MA_API ma_vec3f ma_spatializer_get_velocity(const ma_spatializer* pSpatializer); MA_API void ma_spatializer_get_relative_position_and_direction(const ma_spatializer* pSpatializer, const ma_spatializer_listener* pListener, ma_vec3f* pRelativePos, ma_vec3f* pRelativeDir); /************************************************************************************************************************************************************ ************************************************************************************************************************************************************* DATA CONVERSION =============== This section contains the APIs for data conversion. You will find everything here for channel mapping, sample format conversion, resampling, etc. ************************************************************************************************************************************************************* ************************************************************************************************************************************************************/ /************************************************************************************************************************************************************** Resampling **************************************************************************************************************************************************************/ typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRateIn; ma_uint32 sampleRateOut; ma_uint32 lpfOrder; /* The low-pass filter order. Setting this to 0 will disable low-pass filtering. */ double lpfNyquistFactor; /* 0..1. Defaults to 1. 1 = Half the sampling frequency (Nyquist Frequency), 0.5 = Quarter the sampling frequency (half Nyquest Frequency), etc. */ } ma_linear_resampler_config; MA_API ma_linear_resampler_config ma_linear_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); typedef struct { ma_linear_resampler_config config; ma_uint32 inAdvanceInt; ma_uint32 inAdvanceFrac; ma_uint32 inTimeInt; ma_uint32 inTimeFrac; union { float* f32; ma_int16* s16; } x0; /* The previous input frame. */ union { float* f32; ma_int16* s16; } x1; /* The next input frame. */ ma_lpf lpf; /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_linear_resampler; MA_API ma_result ma_linear_resampler_get_heap_size(const ma_linear_resampler_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_linear_resampler_init_preallocated(const ma_linear_resampler_config* pConfig, void* pHeap, ma_linear_resampler* pResampler); MA_API ma_result ma_linear_resampler_init(const ma_linear_resampler_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_linear_resampler* pResampler); MA_API void ma_linear_resampler_uninit(ma_linear_resampler* pResampler, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_linear_resampler_process_pcm_frames(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut); MA_API ma_result ma_linear_resampler_set_rate(ma_linear_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); MA_API ma_result ma_linear_resampler_set_rate_ratio(ma_linear_resampler* pResampler, float ratioInOut); MA_API ma_uint64 ma_linear_resampler_get_input_latency(const ma_linear_resampler* pResampler); MA_API ma_uint64 ma_linear_resampler_get_output_latency(const ma_linear_resampler* pResampler); MA_API ma_result ma_linear_resampler_get_required_input_frame_count(const ma_linear_resampler* pResampler, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount); MA_API ma_result ma_linear_resampler_get_expected_output_frame_count(const ma_linear_resampler* pResampler, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount); MA_API ma_result ma_linear_resampler_reset(ma_linear_resampler* pResampler); typedef struct ma_resampler_config ma_resampler_config; typedef void ma_resampling_backend; typedef struct { ma_result (* onGetHeapSize )(void* pUserData, const ma_resampler_config* pConfig, size_t* pHeapSizeInBytes); ma_result (* onInit )(void* pUserData, const ma_resampler_config* pConfig, void* pHeap, ma_resampling_backend** ppBackend); void (* onUninit )(void* pUserData, ma_resampling_backend* pBackend, const ma_allocation_callbacks* pAllocationCallbacks); ma_result (* onProcess )(void* pUserData, ma_resampling_backend* pBackend, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut); ma_result (* onSetRate )(void* pUserData, ma_resampling_backend* pBackend, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); /* Optional. Rate changes will be disabled. */ ma_uint64 (* onGetInputLatency )(void* pUserData, const ma_resampling_backend* pBackend); /* Optional. Latency will be reported as 0. */ ma_uint64 (* onGetOutputLatency )(void* pUserData, const ma_resampling_backend* pBackend); /* Optional. Latency will be reported as 0. */ ma_result (* onGetRequiredInputFrameCount )(void* pUserData, const ma_resampling_backend* pBackend, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount); /* Optional. Latency mitigation will be disabled. */ ma_result (* onGetExpectedOutputFrameCount)(void* pUserData, const ma_resampling_backend* pBackend, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount); /* Optional. Latency mitigation will be disabled. */ ma_result (* onReset )(void* pUserData, ma_resampling_backend* pBackend); } ma_resampling_backend_vtable; typedef enum { ma_resample_algorithm_linear = 0, /* Fastest, lowest quality. Optional low-pass filtering. Default. */ ma_resample_algorithm_custom, } ma_resample_algorithm; struct ma_resampler_config { ma_format format; /* Must be either ma_format_f32 or ma_format_s16. */ ma_uint32 channels; ma_uint32 sampleRateIn; ma_uint32 sampleRateOut; ma_resample_algorithm algorithm; /* When set to ma_resample_algorithm_custom, pBackendVTable will be used. */ ma_resampling_backend_vtable* pBackendVTable; void* pBackendUserData; struct { ma_uint32 lpfOrder; } linear; }; MA_API ma_resampler_config ma_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_resample_algorithm algorithm); typedef struct { ma_resampling_backend* pBackend; ma_resampling_backend_vtable* pBackendVTable; void* pBackendUserData; ma_format format; ma_uint32 channels; ma_uint32 sampleRateIn; ma_uint32 sampleRateOut; union { ma_linear_resampler linear; } state; /* State for stock resamplers so we can avoid a malloc. For stock resamplers, pBackend will point here. */ /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_resampler; MA_API ma_result ma_resampler_get_heap_size(const ma_resampler_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_resampler_init_preallocated(const ma_resampler_config* pConfig, void* pHeap, ma_resampler* pResampler); /* Initializes a new resampler object from a config. */ MA_API ma_result ma_resampler_init(const ma_resampler_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_resampler* pResampler); /* Uninitializes a resampler. */ MA_API void ma_resampler_uninit(ma_resampler* pResampler, const ma_allocation_callbacks* pAllocationCallbacks); /* Converts the given input data. Both the input and output frames must be in the format specified in the config when the resampler was initialized. On input, [pFrameCountOut] contains the number of output frames to process. On output it contains the number of output frames that were actually processed, which may be less than the requested amount which will happen if there's not enough input data. You can use ma_resampler_get_expected_output_frame_count() to know how many output frames will be processed for a given number of input frames. On input, [pFrameCountIn] contains the number of input frames contained in [pFramesIn]. On output it contains the number of whole input frames that were actually processed. You can use ma_resampler_get_required_input_frame_count() to know how many input frames you should provide for a given number of output frames. [pFramesIn] can be NULL, in which case zeroes will be used instead. If [pFramesOut] is NULL, a seek is performed. In this case, if [pFrameCountOut] is not NULL it will seek by the specified number of output frames. Otherwise, if [pFramesCountOut] is NULL and [pFrameCountIn] is not NULL, it will seek by the specified number of input frames. When seeking, [pFramesIn] is allowed to NULL, in which case the internal timing state will be updated, but no input will be processed. In this case, any internal filter state will be updated as if zeroes were passed in. It is an error for [pFramesOut] to be non-NULL and [pFrameCountOut] to be NULL. It is an error for both [pFrameCountOut] and [pFrameCountIn] to be NULL. */ MA_API ma_result ma_resampler_process_pcm_frames(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut); /* Sets the input and output sample rate. */ MA_API ma_result ma_resampler_set_rate(ma_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); /* Sets the input and output sample rate as a ratio. The ration is in/out. */ MA_API ma_result ma_resampler_set_rate_ratio(ma_resampler* pResampler, float ratio); /* Retrieves the latency introduced by the resampler in input frames. */ MA_API ma_uint64 ma_resampler_get_input_latency(const ma_resampler* pResampler); /* Retrieves the latency introduced by the resampler in output frames. */ MA_API ma_uint64 ma_resampler_get_output_latency(const ma_resampler* pResampler); /* Calculates the number of whole input frames that would need to be read from the client in order to output the specified number of output frames. The returned value does not include cached input frames. It only returns the number of extra frames that would need to be read from the input buffer in order to output the specified number of output frames. */ MA_API ma_result ma_resampler_get_required_input_frame_count(const ma_resampler* pResampler, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount); /* Calculates the number of whole output frames that would be output after fully reading and consuming the specified number of input frames. */ MA_API ma_result ma_resampler_get_expected_output_frame_count(const ma_resampler* pResampler, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount); /* Resets the resampler's timer and clears its internal cache. */ MA_API ma_result ma_resampler_reset(ma_resampler* pResampler); /************************************************************************************************************************************************************** Channel Conversion **************************************************************************************************************************************************************/ typedef enum { ma_channel_conversion_path_unknown, ma_channel_conversion_path_passthrough, ma_channel_conversion_path_mono_out, /* Converting to mono. */ ma_channel_conversion_path_mono_in, /* Converting from mono. */ ma_channel_conversion_path_shuffle, /* Simple shuffle. Will use this when all channels are present in both input and output channel maps, but just in a different order. */ ma_channel_conversion_path_weights /* Blended based on weights. */ } ma_channel_conversion_path; typedef enum { ma_mono_expansion_mode_duplicate = 0, /* The default. */ ma_mono_expansion_mode_average, /* Average the mono channel across all channels. */ ma_mono_expansion_mode_stereo_only, /* Duplicate to the left and right channels only and ignore the others. */ ma_mono_expansion_mode_default = ma_mono_expansion_mode_duplicate } ma_mono_expansion_mode; typedef struct { ma_format format; ma_uint32 channelsIn; ma_uint32 channelsOut; const ma_channel* pChannelMapIn; const ma_channel* pChannelMapOut; ma_channel_mix_mode mixingMode; ma_bool32 calculateLFEFromSpatialChannels; /* When an output LFE channel is present, but no input LFE, set to true to set the output LFE to the average of all spatial channels (LR, FR, etc.). Ignored when an input LFE is present. */ float** ppWeights; /* [in][out]. Only used when mixingMode is set to ma_channel_mix_mode_custom_weights. */ } ma_channel_converter_config; MA_API ma_channel_converter_config ma_channel_converter_config_init(ma_format format, ma_uint32 channelsIn, const ma_channel* pChannelMapIn, ma_uint32 channelsOut, const ma_channel* pChannelMapOut, ma_channel_mix_mode mixingMode); typedef struct { ma_format format; ma_uint32 channelsIn; ma_uint32 channelsOut; ma_channel_mix_mode mixingMode; ma_channel_conversion_path conversionPath; ma_channel* pChannelMapIn; ma_channel* pChannelMapOut; ma_uint8* pShuffleTable; /* Indexed by output channel index. */ union { float** f32; ma_int32** s16; } weights; /* [in][out] */ /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_channel_converter; MA_API ma_result ma_channel_converter_get_heap_size(const ma_channel_converter_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_channel_converter_init_preallocated(const ma_channel_converter_config* pConfig, void* pHeap, ma_channel_converter* pConverter); MA_API ma_result ma_channel_converter_init(const ma_channel_converter_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_channel_converter* pConverter); MA_API void ma_channel_converter_uninit(ma_channel_converter* pConverter, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_channel_converter_process_pcm_frames(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); MA_API ma_result ma_channel_converter_get_input_channel_map(const ma_channel_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap); MA_API ma_result ma_channel_converter_get_output_channel_map(const ma_channel_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap); /************************************************************************************************************************************************************** Data Conversion **************************************************************************************************************************************************************/ typedef struct { ma_format formatIn; ma_format formatOut; ma_uint32 channelsIn; ma_uint32 channelsOut; ma_uint32 sampleRateIn; ma_uint32 sampleRateOut; ma_channel* pChannelMapIn; ma_channel* pChannelMapOut; ma_dither_mode ditherMode; ma_channel_mix_mode channelMixMode; ma_bool32 calculateLFEFromSpatialChannels; /* When an output LFE channel is present, but no input LFE, set to true to set the output LFE to the average of all spatial channels (LR, FR, etc.). Ignored when an input LFE is present. */ float** ppChannelWeights; /* [in][out]. Only used when mixingMode is set to ma_channel_mix_mode_custom_weights. */ ma_bool32 allowDynamicSampleRate; ma_resampler_config resampling; } ma_data_converter_config; MA_API ma_data_converter_config ma_data_converter_config_init_default(void); MA_API ma_data_converter_config ma_data_converter_config_init(ma_format formatIn, ma_format formatOut, ma_uint32 channelsIn, ma_uint32 channelsOut, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); typedef enum { ma_data_converter_execution_path_passthrough, /* No conversion. */ ma_data_converter_execution_path_format_only, /* Only format conversion. */ ma_data_converter_execution_path_channels_only, /* Only channel conversion. */ ma_data_converter_execution_path_resample_only, /* Only resampling. */ ma_data_converter_execution_path_resample_first, /* All conversions, but resample as the first step. */ ma_data_converter_execution_path_channels_first /* All conversions, but channels as the first step. */ } ma_data_converter_execution_path; typedef struct { ma_format formatIn; ma_format formatOut; ma_uint32 channelsIn; ma_uint32 channelsOut; ma_uint32 sampleRateIn; ma_uint32 sampleRateOut; ma_dither_mode ditherMode; ma_data_converter_execution_path executionPath; /* The execution path the data converter will follow when processing. */ ma_channel_converter channelConverter; ma_resampler resampler; ma_bool8 hasPreFormatConversion; ma_bool8 hasPostFormatConversion; ma_bool8 hasChannelConverter; ma_bool8 hasResampler; ma_bool8 isPassthrough; /* Memory management. */ ma_bool8 _ownsHeap; void* _pHeap; } ma_data_converter; MA_API ma_result ma_data_converter_get_heap_size(const ma_data_converter_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_data_converter_init_preallocated(const ma_data_converter_config* pConfig, void* pHeap, ma_data_converter* pConverter); MA_API ma_result ma_data_converter_init(const ma_data_converter_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_converter* pConverter); MA_API void ma_data_converter_uninit(ma_data_converter* pConverter, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_data_converter_process_pcm_frames(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut); MA_API ma_result ma_data_converter_set_rate(ma_data_converter* pConverter, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); MA_API ma_result ma_data_converter_set_rate_ratio(ma_data_converter* pConverter, float ratioInOut); MA_API ma_uint64 ma_data_converter_get_input_latency(const ma_data_converter* pConverter); MA_API ma_uint64 ma_data_converter_get_output_latency(const ma_data_converter* pConverter); MA_API ma_result ma_data_converter_get_required_input_frame_count(const ma_data_converter* pConverter, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount); MA_API ma_result ma_data_converter_get_expected_output_frame_count(const ma_data_converter* pConverter, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount); MA_API ma_result ma_data_converter_get_input_channel_map(const ma_data_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap); MA_API ma_result ma_data_converter_get_output_channel_map(const ma_data_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap); MA_API ma_result ma_data_converter_reset(ma_data_converter* pConverter); /************************************************************************************************************************************************************ Format Conversion ************************************************************************************************************************************************************/ MA_API void ma_pcm_u8_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_u8_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_u8_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_u8_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s16_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s16_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s16_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s16_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s24_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s24_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s24_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s24_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_s32_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_f32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_f32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_f32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_f32_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); MA_API void ma_pcm_convert(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 sampleCount, ma_dither_mode ditherMode); MA_API void ma_convert_pcm_frames_format(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 frameCount, ma_uint32 channels, ma_dither_mode ditherMode); /* Deinterleaves an interleaved buffer. */ MA_API void ma_deinterleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void* pInterleavedPCMFrames, void** ppDeinterleavedPCMFrames); /* Interleaves a group of deinterleaved buffers. */ MA_API void ma_interleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void** ppDeinterleavedPCMFrames, void* pInterleavedPCMFrames); /************************************************************************************************************************************************************ Channel Maps ************************************************************************************************************************************************************/ /* This is used in the shuffle table to indicate that the channel index is undefined and should be ignored. */ #define MA_CHANNEL_INDEX_NULL 255 /* Retrieves the channel position of the specified channel in the given channel map. The pChannelMap parameter can be null, in which case miniaudio's default channel map will be assumed. */ MA_API ma_channel ma_channel_map_get_channel(const ma_channel* pChannelMap, ma_uint32 channelCount, ma_uint32 channelIndex); /* Initializes a blank channel map. When a blank channel map is specified anywhere it indicates that the native channel map should be used. */ MA_API void ma_channel_map_init_blank(ma_channel* pChannelMap, ma_uint32 channels); /* Helper for retrieving a standard channel map. The output channel map buffer must have a capacity of at least `channelMapCap`. */ MA_API void ma_channel_map_init_standard(ma_standard_channel_map standardChannelMap, ma_channel* pChannelMap, size_t channelMapCap, ma_uint32 channels); /* Copies a channel map. Both input and output channel map buffers must have a capacity of at least `channels`. */ MA_API void ma_channel_map_copy(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels); /* Copies a channel map if one is specified, otherwise copies the default channel map. The output buffer must have a capacity of at least `channels`. If not NULL, the input channel map must also have a capacity of at least `channels`. */ MA_API void ma_channel_map_copy_or_default(ma_channel* pOut, size_t channelMapCapOut, const ma_channel* pIn, ma_uint32 channels); /* Determines whether or not a channel map is valid. A blank channel map is valid (all channels set to MA_CHANNEL_NONE). The way a blank channel map is handled is context specific, but is usually treated as a passthrough. Invalid channel maps: - A channel map with no channels - A channel map with more than one channel and a mono channel The channel map buffer must have a capacity of at least `channels`. */ MA_API ma_bool32 ma_channel_map_is_valid(const ma_channel* pChannelMap, ma_uint32 channels); /* Helper for comparing two channel maps for equality. This assumes the channel count is the same between the two. Both channels map buffers must have a capacity of at least `channels`. */ MA_API ma_bool32 ma_channel_map_is_equal(const ma_channel* pChannelMapA, const ma_channel* pChannelMapB, ma_uint32 channels); /* Helper for determining if a channel map is blank (all channels set to MA_CHANNEL_NONE). The channel map buffer must have a capacity of at least `channels`. */ MA_API ma_bool32 ma_channel_map_is_blank(const ma_channel* pChannelMap, ma_uint32 channels); /* Helper for determining whether or not a channel is present in the given channel map. The channel map buffer must have a capacity of at least `channels`. */ MA_API ma_bool32 ma_channel_map_contains_channel_position(ma_uint32 channels, const ma_channel* pChannelMap, ma_channel channelPosition); /* Find a channel position in the given channel map. Returns MA_TRUE if the channel is found; MA_FALSE otherwise. The index of the channel is output to `pChannelIndex`. The channel map buffer must have a capacity of at least `channels`. */ MA_API ma_bool32 ma_channel_map_find_channel_position(ma_uint32 channels, const ma_channel* pChannelMap, ma_channel channelPosition, ma_uint32* pChannelIndex); /* Generates a string representing the given channel map. This is for printing and debugging purposes, not serialization/deserialization. Returns the length of the string, not including the null terminator. */ MA_API size_t ma_channel_map_to_string(const ma_channel* pChannelMap, ma_uint32 channels, char* pBufferOut, size_t bufferCap); /* Retrieves a human readable version of a channel position. */ MA_API const char* ma_channel_position_to_string(ma_channel channel); /************************************************************************************************************************************************************ Conversion Helpers ************************************************************************************************************************************************************/ /* High-level helper for doing a full format conversion in one go. Returns the number of output frames. Call this with pOut set to NULL to determine the required size of the output buffer. frameCountOut should be set to the capacity of pOut. If pOut is NULL, frameCountOut is ignored. A return value of 0 indicates an error. This function is useful for one-off bulk conversions, but if you're streaming data you should use the ma_data_converter APIs instead. */ MA_API ma_uint64 ma_convert_frames(void* pOut, ma_uint64 frameCountOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, const void* pIn, ma_uint64 frameCountIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn); MA_API ma_uint64 ma_convert_frames_ex(void* pOut, ma_uint64 frameCountOut, const void* pIn, ma_uint64 frameCountIn, const ma_data_converter_config* pConfig); /************************************************************************************************************************************************************ Data Source ************************************************************************************************************************************************************/ typedef void ma_data_source; #define MA_DATA_SOURCE_SELF_MANAGED_RANGE_AND_LOOP_POINT 0x00000001 typedef struct { ma_result (* onRead)(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); ma_result (* onSeek)(ma_data_source* pDataSource, ma_uint64 frameIndex); ma_result (* onGetDataFormat)(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap); ma_result (* onGetCursor)(ma_data_source* pDataSource, ma_uint64* pCursor); ma_result (* onGetLength)(ma_data_source* pDataSource, ma_uint64* pLength); ma_result (* onSetLooping)(ma_data_source* pDataSource, ma_bool32 isLooping); ma_uint32 flags; } ma_data_source_vtable; typedef ma_data_source* (* ma_data_source_get_next_proc)(ma_data_source* pDataSource); typedef struct { const ma_data_source_vtable* vtable; } ma_data_source_config; MA_API ma_data_source_config ma_data_source_config_init(void); typedef struct { const ma_data_source_vtable* vtable; ma_uint64 rangeBegInFrames; ma_uint64 rangeEndInFrames; /* Set to -1 for unranged (default). */ ma_uint64 loopBegInFrames; /* Relative to rangeBegInFrames. */ ma_uint64 loopEndInFrames; /* Relative to rangeBegInFrames. Set to -1 for the end of the range. */ ma_data_source* pCurrent; /* When non-NULL, the data source being initialized will act as a proxy and will route all operations to pCurrent. Used in conjunction with pNext/onGetNext for seamless chaining. */ ma_data_source* pNext; /* When set to NULL, onGetNext will be used. */ ma_data_source_get_next_proc onGetNext; /* Will be used when pNext is NULL. If both are NULL, no next will be used. */ MA_ATOMIC(4, ma_bool32) isLooping; } ma_data_source_base; MA_API ma_result ma_data_source_init(const ma_data_source_config* pConfig, ma_data_source* pDataSource); MA_API void ma_data_source_uninit(ma_data_source* pDataSource); MA_API ma_result ma_data_source_read_pcm_frames(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); /* Must support pFramesOut = NULL in which case a forward seek should be performed. */ MA_API ma_result ma_data_source_seek_pcm_frames(ma_data_source* pDataSource, ma_uint64 frameCount, ma_uint64* pFramesSeeked); /* Can only seek forward. Equivalent to ma_data_source_read_pcm_frames(pDataSource, NULL, frameCount, &framesRead); */ MA_API ma_result ma_data_source_seek_to_pcm_frame(ma_data_source* pDataSource, ma_uint64 frameIndex); MA_API ma_result ma_data_source_seek_seconds(ma_data_source* pDataSource, float secondCount, float* pSecondsSeeked); /* Can only seek forward. Abstraction to ma_data_source_seek_pcm_frames() */ MA_API ma_result ma_data_source_seek_to_second(ma_data_source* pDataSource, float seekPointInSeconds); /* Abstraction to ma_data_source_seek_to_pcm_frame() */ MA_API ma_result ma_data_source_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap); MA_API ma_result ma_data_source_get_cursor_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pCursor); MA_API ma_result ma_data_source_get_length_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLength); /* Returns MA_NOT_IMPLEMENTED if the length is unknown or cannot be determined. Decoders can return this. */ MA_API ma_result ma_data_source_get_cursor_in_seconds(ma_data_source* pDataSource, float* pCursor); MA_API ma_result ma_data_source_get_length_in_seconds(ma_data_source* pDataSource, float* pLength); MA_API ma_result ma_data_source_set_looping(ma_data_source* pDataSource, ma_bool32 isLooping); MA_API ma_bool32 ma_data_source_is_looping(const ma_data_source* pDataSource); MA_API ma_result ma_data_source_set_range_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 rangeBegInFrames, ma_uint64 rangeEndInFrames); MA_API void ma_data_source_get_range_in_pcm_frames(const ma_data_source* pDataSource, ma_uint64* pRangeBegInFrames, ma_uint64* pRangeEndInFrames); MA_API ma_result ma_data_source_set_loop_point_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 loopBegInFrames, ma_uint64 loopEndInFrames); MA_API void ma_data_source_get_loop_point_in_pcm_frames(const ma_data_source* pDataSource, ma_uint64* pLoopBegInFrames, ma_uint64* pLoopEndInFrames); MA_API ma_result ma_data_source_set_current(ma_data_source* pDataSource, ma_data_source* pCurrentDataSource); MA_API ma_data_source* ma_data_source_get_current(const ma_data_source* pDataSource); MA_API ma_result ma_data_source_set_next(ma_data_source* pDataSource, ma_data_source* pNextDataSource); MA_API ma_data_source* ma_data_source_get_next(const ma_data_source* pDataSource); MA_API ma_result ma_data_source_set_next_callback(ma_data_source* pDataSource, ma_data_source_get_next_proc onGetNext); MA_API ma_data_source_get_next_proc ma_data_source_get_next_callback(const ma_data_source* pDataSource); typedef struct { ma_data_source_base ds; ma_format format; ma_uint32 channels; ma_uint32 sampleRate; ma_uint64 cursor; ma_uint64 sizeInFrames; const void* pData; } ma_audio_buffer_ref; MA_API ma_result ma_audio_buffer_ref_init(ma_format format, ma_uint32 channels, const void* pData, ma_uint64 sizeInFrames, ma_audio_buffer_ref* pAudioBufferRef); MA_API void ma_audio_buffer_ref_uninit(ma_audio_buffer_ref* pAudioBufferRef); MA_API ma_result ma_audio_buffer_ref_set_data(ma_audio_buffer_ref* pAudioBufferRef, const void* pData, ma_uint64 sizeInFrames); MA_API ma_uint64 ma_audio_buffer_ref_read_pcm_frames(ma_audio_buffer_ref* pAudioBufferRef, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop); MA_API ma_result ma_audio_buffer_ref_seek_to_pcm_frame(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameIndex); MA_API ma_result ma_audio_buffer_ref_map(ma_audio_buffer_ref* pAudioBufferRef, void** ppFramesOut, ma_uint64* pFrameCount); MA_API ma_result ma_audio_buffer_ref_unmap(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameCount); /* Returns MA_AT_END if the end has been reached. This should be considered successful. */ MA_API ma_bool32 ma_audio_buffer_ref_at_end(const ma_audio_buffer_ref* pAudioBufferRef); MA_API ma_result ma_audio_buffer_ref_get_cursor_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pCursor); MA_API ma_result ma_audio_buffer_ref_get_length_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pLength); MA_API ma_result ma_audio_buffer_ref_get_available_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pAvailableFrames); typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; ma_uint64 sizeInFrames; const void* pData; /* If set to NULL, will allocate a block of memory for you. */ ma_allocation_callbacks allocationCallbacks; } ma_audio_buffer_config; MA_API ma_audio_buffer_config ma_audio_buffer_config_init(ma_format format, ma_uint32 channels, ma_uint64 sizeInFrames, const void* pData, const ma_allocation_callbacks* pAllocationCallbacks); typedef struct { ma_audio_buffer_ref ref; ma_allocation_callbacks allocationCallbacks; ma_bool32 ownsData; /* Used to control whether or not miniaudio owns the data buffer. If set to true, pData will be freed in ma_audio_buffer_uninit(). */ ma_uint8 _pExtraData[1]; /* For allocating a buffer with the memory located directly after the other memory of the structure. */ } ma_audio_buffer; MA_API ma_result ma_audio_buffer_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer); MA_API ma_result ma_audio_buffer_init_copy(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer); MA_API ma_result ma_audio_buffer_alloc_and_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer** ppAudioBuffer); /* Always copies the data. Doesn't make sense to use this otherwise. Use ma_audio_buffer_uninit_and_free() to uninit. */ MA_API void ma_audio_buffer_uninit(ma_audio_buffer* pAudioBuffer); MA_API void ma_audio_buffer_uninit_and_free(ma_audio_buffer* pAudioBuffer); MA_API ma_uint64 ma_audio_buffer_read_pcm_frames(ma_audio_buffer* pAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop); MA_API ma_result ma_audio_buffer_seek_to_pcm_frame(ma_audio_buffer* pAudioBuffer, ma_uint64 frameIndex); MA_API ma_result ma_audio_buffer_map(ma_audio_buffer* pAudioBuffer, void** ppFramesOut, ma_uint64* pFrameCount); MA_API ma_result ma_audio_buffer_unmap(ma_audio_buffer* pAudioBuffer, ma_uint64 frameCount); /* Returns MA_AT_END if the end has been reached. This should be considered successful. */ MA_API ma_bool32 ma_audio_buffer_at_end(const ma_audio_buffer* pAudioBuffer); MA_API ma_result ma_audio_buffer_get_cursor_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pCursor); MA_API ma_result ma_audio_buffer_get_length_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pLength); MA_API ma_result ma_audio_buffer_get_available_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pAvailableFrames); /* Paged Audio Buffer ================== A paged audio buffer is made up of a linked list of pages. It's expandable, but not shrinkable. It can be used for cases where audio data is streamed in asynchronously while allowing data to be read at the same time. This is lock-free, but not 100% thread safe. You can append a page and read from the buffer across simultaneously across different threads, however only one thread at a time can append, and only one thread at a time can read and seek. */ typedef struct ma_paged_audio_buffer_page ma_paged_audio_buffer_page; struct ma_paged_audio_buffer_page { MA_ATOMIC(MA_SIZEOF_PTR, ma_paged_audio_buffer_page*) pNext; ma_uint64 sizeInFrames; ma_uint8 pAudioData[1]; }; typedef struct { ma_format format; ma_uint32 channels; ma_paged_audio_buffer_page head; /* Dummy head for the lock-free algorithm. Always has a size of 0. */ MA_ATOMIC(MA_SIZEOF_PTR, ma_paged_audio_buffer_page*) pTail; /* Never null. Initially set to &head. */ } ma_paged_audio_buffer_data; MA_API ma_result ma_paged_audio_buffer_data_init(ma_format format, ma_uint32 channels, ma_paged_audio_buffer_data* pData); MA_API void ma_paged_audio_buffer_data_uninit(ma_paged_audio_buffer_data* pData, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_paged_audio_buffer_page* ma_paged_audio_buffer_data_get_head(ma_paged_audio_buffer_data* pData); MA_API ma_paged_audio_buffer_page* ma_paged_audio_buffer_data_get_tail(ma_paged_audio_buffer_data* pData); MA_API ma_result ma_paged_audio_buffer_data_get_length_in_pcm_frames(ma_paged_audio_buffer_data* pData, ma_uint64* pLength); MA_API ma_result ma_paged_audio_buffer_data_allocate_page(ma_paged_audio_buffer_data* pData, ma_uint64 pageSizeInFrames, const void* pInitialData, const ma_allocation_callbacks* pAllocationCallbacks, ma_paged_audio_buffer_page** ppPage); MA_API ma_result ma_paged_audio_buffer_data_free_page(ma_paged_audio_buffer_data* pData, ma_paged_audio_buffer_page* pPage, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_paged_audio_buffer_data_append_page(ma_paged_audio_buffer_data* pData, ma_paged_audio_buffer_page* pPage); MA_API ma_result ma_paged_audio_buffer_data_allocate_and_append_page(ma_paged_audio_buffer_data* pData, ma_uint32 pageSizeInFrames, const void* pInitialData, const ma_allocation_callbacks* pAllocationCallbacks); typedef struct { ma_paged_audio_buffer_data* pData; /* Must not be null. */ } ma_paged_audio_buffer_config; MA_API ma_paged_audio_buffer_config ma_paged_audio_buffer_config_init(ma_paged_audio_buffer_data* pData); typedef struct { ma_data_source_base ds; ma_paged_audio_buffer_data* pData; /* Audio data is read from here. Cannot be null. */ ma_paged_audio_buffer_page* pCurrent; ma_uint64 relativeCursor; /* Relative to the current page. */ ma_uint64 absoluteCursor; } ma_paged_audio_buffer; MA_API ma_result ma_paged_audio_buffer_init(const ma_paged_audio_buffer_config* pConfig, ma_paged_audio_buffer* pPagedAudioBuffer); MA_API void ma_paged_audio_buffer_uninit(ma_paged_audio_buffer* pPagedAudioBuffer); MA_API ma_result ma_paged_audio_buffer_read_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); /* Returns MA_AT_END if no more pages available. */ MA_API ma_result ma_paged_audio_buffer_seek_to_pcm_frame(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64 frameIndex); MA_API ma_result ma_paged_audio_buffer_get_cursor_in_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64* pCursor); MA_API ma_result ma_paged_audio_buffer_get_length_in_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64* pLength); /************************************************************************************************************************************************************ Ring Buffer ************************************************************************************************************************************************************/ typedef struct { void* pBuffer; ma_uint32 subbufferSizeInBytes; ma_uint32 subbufferCount; ma_uint32 subbufferStrideInBytes; MA_ATOMIC(4, ma_uint32) encodedReadOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. Must be used atomically. */ MA_ATOMIC(4, ma_uint32) encodedWriteOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. Must be used atomically. */ ma_bool8 ownsBuffer; /* Used to know whether or not miniaudio is responsible for free()-ing the buffer. */ ma_bool8 clearOnWriteAcquire; /* When set, clears the acquired write buffer before returning from ma_rb_acquire_write(). */ ma_allocation_callbacks allocationCallbacks; } ma_rb; MA_API ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size_t subbufferStrideInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB); MA_API ma_result ma_rb_init(size_t bufferSizeInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB); MA_API void ma_rb_uninit(ma_rb* pRB); MA_API void ma_rb_reset(ma_rb* pRB); MA_API ma_result ma_rb_acquire_read(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut); MA_API ma_result ma_rb_commit_read(ma_rb* pRB, size_t sizeInBytes); MA_API ma_result ma_rb_acquire_write(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut); MA_API ma_result ma_rb_commit_write(ma_rb* pRB, size_t sizeInBytes); MA_API ma_result ma_rb_seek_read(ma_rb* pRB, size_t offsetInBytes); MA_API ma_result ma_rb_seek_write(ma_rb* pRB, size_t offsetInBytes); MA_API ma_int32 ma_rb_pointer_distance(ma_rb* pRB); /* Returns the distance between the write pointer and the read pointer. Should never be negative for a correct program. Will return the number of bytes that can be read before the read pointer hits the write pointer. */ MA_API ma_uint32 ma_rb_available_read(ma_rb* pRB); MA_API ma_uint32 ma_rb_available_write(ma_rb* pRB); MA_API size_t ma_rb_get_subbuffer_size(ma_rb* pRB); MA_API size_t ma_rb_get_subbuffer_stride(ma_rb* pRB); MA_API size_t ma_rb_get_subbuffer_offset(ma_rb* pRB, size_t subbufferIndex); MA_API void* ma_rb_get_subbuffer_ptr(ma_rb* pRB, size_t subbufferIndex, void* pBuffer); typedef struct { ma_data_source_base ds; ma_rb rb; ma_format format; ma_uint32 channels; ma_uint32 sampleRate; /* Not required for the ring buffer itself, but useful for associating the data with some sample rate, particularly for data sources. */ } ma_pcm_rb; MA_API ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subbufferSizeInFrames, ma_uint32 subbufferCount, ma_uint32 subbufferStrideInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB); MA_API ma_result ma_pcm_rb_init(ma_format format, ma_uint32 channels, ma_uint32 bufferSizeInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB); MA_API void ma_pcm_rb_uninit(ma_pcm_rb* pRB); MA_API void ma_pcm_rb_reset(ma_pcm_rb* pRB); MA_API ma_result ma_pcm_rb_acquire_read(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut); MA_API ma_result ma_pcm_rb_commit_read(ma_pcm_rb* pRB, ma_uint32 sizeInFrames); MA_API ma_result ma_pcm_rb_acquire_write(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut); MA_API ma_result ma_pcm_rb_commit_write(ma_pcm_rb* pRB, ma_uint32 sizeInFrames); MA_API ma_result ma_pcm_rb_seek_read(ma_pcm_rb* pRB, ma_uint32 offsetInFrames); MA_API ma_result ma_pcm_rb_seek_write(ma_pcm_rb* pRB, ma_uint32 offsetInFrames); MA_API ma_int32 ma_pcm_rb_pointer_distance(ma_pcm_rb* pRB); /* Return value is in frames. */ MA_API ma_uint32 ma_pcm_rb_available_read(ma_pcm_rb* pRB); MA_API ma_uint32 ma_pcm_rb_available_write(ma_pcm_rb* pRB); MA_API ma_uint32 ma_pcm_rb_get_subbuffer_size(ma_pcm_rb* pRB); MA_API ma_uint32 ma_pcm_rb_get_subbuffer_stride(ma_pcm_rb* pRB); MA_API ma_uint32 ma_pcm_rb_get_subbuffer_offset(ma_pcm_rb* pRB, ma_uint32 subbufferIndex); MA_API void* ma_pcm_rb_get_subbuffer_ptr(ma_pcm_rb* pRB, ma_uint32 subbufferIndex, void* pBuffer); MA_API ma_format ma_pcm_rb_get_format(const ma_pcm_rb* pRB); MA_API ma_uint32 ma_pcm_rb_get_channels(const ma_pcm_rb* pRB); MA_API ma_uint32 ma_pcm_rb_get_sample_rate(const ma_pcm_rb* pRB); MA_API void ma_pcm_rb_set_sample_rate(ma_pcm_rb* pRB, ma_uint32 sampleRate); /* The idea of the duplex ring buffer is to act as the intermediary buffer when running two asynchronous devices in a duplex set up. The capture device writes to it, and then a playback device reads from it. At the moment this is just a simple naive implementation, but in the future I want to implement some dynamic resampling to seamlessly handle desyncs. Note that the API is work in progress and may change at any time in any version. The size of the buffer is based on the capture side since that's what'll be written to the buffer. It is based on the capture period size in frames. The internal sample rate of the capture device is also needed in order to calculate the size. */ typedef struct { ma_pcm_rb rb; } ma_duplex_rb; MA_API ma_result ma_duplex_rb_init(ma_format captureFormat, ma_uint32 captureChannels, ma_uint32 sampleRate, ma_uint32 captureInternalSampleRate, ma_uint32 captureInternalPeriodSizeInFrames, const ma_allocation_callbacks* pAllocationCallbacks, ma_duplex_rb* pRB); MA_API ma_result ma_duplex_rb_uninit(ma_duplex_rb* pRB); /************************************************************************************************************************************************************ Miscellaneous Helpers ************************************************************************************************************************************************************/ /* Retrieves a human readable description of the given result code. */ MA_API const char* ma_result_description(ma_result result); /* malloc() */ MA_API void* ma_malloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks); /* calloc() */ MA_API void* ma_calloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks); /* realloc() */ MA_API void* ma_realloc(void* p, size_t sz, const ma_allocation_callbacks* pAllocationCallbacks); /* free() */ MA_API void ma_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks); /* Performs an aligned malloc, with the assumption that the alignment is a power of 2. */ MA_API void* ma_aligned_malloc(size_t sz, size_t alignment, const ma_allocation_callbacks* pAllocationCallbacks); /* Free's an aligned malloc'd buffer. */ MA_API void ma_aligned_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks); /* Retrieves a friendly name for a format. */ MA_API const char* ma_get_format_name(ma_format format); /* Blends two frames in floating point format. */ MA_API void ma_blend_f32(float* pOut, float* pInA, float* pInB, float factor, ma_uint32 channels); /* Retrieves the size of a sample in bytes for the given format. This API is efficient and is implemented using a lookup table. Thread Safety: SAFE This API is pure. */ MA_API ma_uint32 ma_get_bytes_per_sample(ma_format format); static MA_INLINE ma_uint32 ma_get_bytes_per_frame(ma_format format, ma_uint32 channels) { return ma_get_bytes_per_sample(format) * channels; } /* Converts a log level to a string. */ MA_API const char* ma_log_level_to_string(ma_uint32 logLevel); /************************************************************************************************************************************************************ Synchronization ************************************************************************************************************************************************************/ /* Locks a spinlock. */ MA_API ma_result ma_spinlock_lock(volatile ma_spinlock* pSpinlock); /* Locks a spinlock, but does not yield() when looping. */ MA_API ma_result ma_spinlock_lock_noyield(volatile ma_spinlock* pSpinlock); /* Unlocks a spinlock. */ MA_API ma_result ma_spinlock_unlock(volatile ma_spinlock* pSpinlock); #ifndef MA_NO_THREADING /* Creates a mutex. A mutex must be created from a valid context. A mutex is initially unlocked. */ MA_API ma_result ma_mutex_init(ma_mutex* pMutex); /* Deletes a mutex. */ MA_API void ma_mutex_uninit(ma_mutex* pMutex); /* Locks a mutex with an infinite timeout. */ MA_API void ma_mutex_lock(ma_mutex* pMutex); /* Unlocks a mutex. */ MA_API void ma_mutex_unlock(ma_mutex* pMutex); /* Initializes an auto-reset event. */ MA_API ma_result ma_event_init(ma_event* pEvent); /* Uninitializes an auto-reset event. */ MA_API void ma_event_uninit(ma_event* pEvent); /* Waits for the specified auto-reset event to become signalled. */ MA_API ma_result ma_event_wait(ma_event* pEvent); /* Signals the specified auto-reset event. */ MA_API ma_result ma_event_signal(ma_event* pEvent); MA_API ma_result ma_semaphore_init(int initialValue, ma_semaphore* pSemaphore); MA_API void ma_semaphore_uninit(ma_semaphore* pSemaphore); MA_API ma_result ma_semaphore_wait(ma_semaphore* pSemaphore); MA_API ma_result ma_semaphore_release(ma_semaphore* pSemaphore); #endif /* MA_NO_THREADING */ /* Fence ===== This locks while the counter is larger than 0. Counter can be incremented and decremented by any thread, but care needs to be taken when waiting. It is possible for one thread to acquire the fence just as another thread returns from ma_fence_wait(). The idea behind a fence is to allow you to wait for a group of operations to complete. When an operation starts, the counter is incremented which locks the fence. When the operation completes, the fence will be released which decrements the counter. ma_fence_wait() will block until the counter hits zero. If threading is disabled, ma_fence_wait() will spin on the counter. */ typedef struct { #ifndef MA_NO_THREADING ma_event e; #endif ma_uint32 counter; } ma_fence; MA_API ma_result ma_fence_init(ma_fence* pFence); MA_API void ma_fence_uninit(ma_fence* pFence); MA_API ma_result ma_fence_acquire(ma_fence* pFence); /* Increment counter. */ MA_API ma_result ma_fence_release(ma_fence* pFence); /* Decrement counter. */ MA_API ma_result ma_fence_wait(ma_fence* pFence); /* Wait for counter to reach 0. */ /* Notification callback for asynchronous operations. */ typedef void ma_async_notification; typedef struct { void (* onSignal)(ma_async_notification* pNotification); } ma_async_notification_callbacks; MA_API ma_result ma_async_notification_signal(ma_async_notification* pNotification); /* Simple polling notification. This just sets a variable when the notification has been signalled which is then polled with ma_async_notification_poll_is_signalled() */ typedef struct { ma_async_notification_callbacks cb; ma_bool32 signalled; } ma_async_notification_poll; MA_API ma_result ma_async_notification_poll_init(ma_async_notification_poll* pNotificationPoll); MA_API ma_bool32 ma_async_notification_poll_is_signalled(const ma_async_notification_poll* pNotificationPoll); /* Event Notification This uses an ma_event. If threading is disabled (MA_NO_THREADING), initialization will fail. */ typedef struct { ma_async_notification_callbacks cb; #ifndef MA_NO_THREADING ma_event e; #endif } ma_async_notification_event; MA_API ma_result ma_async_notification_event_init(ma_async_notification_event* pNotificationEvent); MA_API ma_result ma_async_notification_event_uninit(ma_async_notification_event* pNotificationEvent); MA_API ma_result ma_async_notification_event_wait(ma_async_notification_event* pNotificationEvent); MA_API ma_result ma_async_notification_event_signal(ma_async_notification_event* pNotificationEvent); /************************************************************************************************************************************************************ Job Queue ************************************************************************************************************************************************************/ /* Slot Allocator -------------- The idea of the slot allocator is for it to be used in conjunction with a fixed sized buffer. You use the slot allocator to allocate an index that can be used as the insertion point for an object. Slots are reference counted to help mitigate the ABA problem in the lock-free queue we use for tracking jobs. The slot index is stored in the low 32 bits. The reference counter is stored in the high 32 bits: +-----------------+-----------------+ | 32 Bits | 32 Bits | +-----------------+-----------------+ | Reference Count | Slot Index | +-----------------+-----------------+ */ typedef struct { ma_uint32 capacity; /* The number of slots to make available. */ } ma_slot_allocator_config; MA_API ma_slot_allocator_config ma_slot_allocator_config_init(ma_uint32 capacity); typedef struct { MA_ATOMIC(4, ma_uint32) bitfield; /* Must be used atomically because the allocation and freeing routines need to make copies of this which must never be optimized away by the compiler. */ } ma_slot_allocator_group; typedef struct { ma_slot_allocator_group* pGroups; /* Slots are grouped in chunks of 32. */ ma_uint32* pSlots; /* 32 bits for reference counting for ABA mitigation. */ ma_uint32 count; /* Allocation count. */ ma_uint32 capacity; /* Memory management. */ ma_bool32 _ownsHeap; void* _pHeap; } ma_slot_allocator; MA_API ma_result ma_slot_allocator_get_heap_size(const ma_slot_allocator_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_slot_allocator_init_preallocated(const ma_slot_allocator_config* pConfig, void* pHeap, ma_slot_allocator* pAllocator); MA_API ma_result ma_slot_allocator_init(const ma_slot_allocator_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_slot_allocator* pAllocator); MA_API void ma_slot_allocator_uninit(ma_slot_allocator* pAllocator, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_slot_allocator_alloc(ma_slot_allocator* pAllocator, ma_uint64* pSlot); MA_API ma_result ma_slot_allocator_free(ma_slot_allocator* pAllocator, ma_uint64 slot); typedef struct ma_job ma_job; /* Callback for processing a job. Each job type will have their own processing callback which will be called by ma_job_process(). */ typedef ma_result (* ma_job_proc)(ma_job* pJob); /* When a job type is added here an callback needs to be added go "g_jobVTable" in the implementation section. */ typedef enum { /* Miscellaneous. */ MA_JOB_TYPE_QUIT = 0, MA_JOB_TYPE_CUSTOM, /* Resource Manager. */ MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER_NODE, MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER_NODE, MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE, MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER, MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER, MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_STREAM, MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_STREAM, MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_STREAM, MA_JOB_TYPE_RESOURCE_MANAGER_SEEK_DATA_STREAM, /* Device. */ MA_JOB_TYPE_DEVICE_AAUDIO_REROUTE, /* Count. Must always be last. */ MA_JOB_TYPE_COUNT } ma_job_type; struct ma_job { union { struct { ma_uint16 code; /* Job type. */ ma_uint16 slot; /* Index into a ma_slot_allocator. */ ma_uint32 refcount; } breakup; ma_uint64 allocation; } toc; /* 8 bytes. We encode the job code into the slot allocation data to save space. */ MA_ATOMIC(8, ma_uint64) next; /* refcount + slot for the next item. Does not include the job code. */ ma_uint32 order; /* Execution order. Used to create a data dependency and ensure a job is executed in order. Usage is contextual depending on the job type. */ union { /* Miscellaneous. */ struct { ma_job_proc proc; ma_uintptr data0; ma_uintptr data1; } custom; /* Resource Manager */ union { struct { /*ma_resource_manager**/ void* pResourceManager; /*ma_resource_manager_data_buffer_node**/ void* pDataBufferNode; char* pFilePath; wchar_t* pFilePathW; ma_uint32 flags; /* Resource manager data source flags that were used when initializing the data buffer. */ ma_async_notification* pInitNotification; /* Signalled when the data buffer has been initialized and the format/channels/rate can be retrieved. */ ma_async_notification* pDoneNotification; /* Signalled when the data buffer has been fully decoded. Will be passed through to MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE when decoding. */ ma_fence* pInitFence; /* Released when initialization of the decoder is complete. */ ma_fence* pDoneFence; /* Released if initialization of the decoder fails. Passed through to PAGE_DATA_BUFFER_NODE untouched if init is successful. */ } loadDataBufferNode; struct { /*ma_resource_manager**/ void* pResourceManager; /*ma_resource_manager_data_buffer_node**/ void* pDataBufferNode; ma_async_notification* pDoneNotification; ma_fence* pDoneFence; } freeDataBufferNode; struct { /*ma_resource_manager**/ void* pResourceManager; /*ma_resource_manager_data_buffer_node**/ void* pDataBufferNode; /*ma_decoder**/ void* pDecoder; ma_async_notification* pDoneNotification; /* Signalled when the data buffer has been fully decoded. */ ma_fence* pDoneFence; /* Passed through from LOAD_DATA_BUFFER_NODE and released when the data buffer completes decoding or an error occurs. */ } pageDataBufferNode; struct { /*ma_resource_manager_data_buffer**/ void* pDataBuffer; ma_async_notification* pInitNotification; /* Signalled when the data buffer has been initialized and the format/channels/rate can be retrieved. */ ma_async_notification* pDoneNotification; /* Signalled when the data buffer has been fully decoded. */ ma_fence* pInitFence; /* Released when the data buffer has been initialized and the format/channels/rate can be retrieved. */ ma_fence* pDoneFence; /* Released when the data buffer has been fully decoded. */ ma_uint64 rangeBegInPCMFrames; ma_uint64 rangeEndInPCMFrames; ma_uint64 loopPointBegInPCMFrames; ma_uint64 loopPointEndInPCMFrames; ma_uint32 isLooping; } loadDataBuffer; struct { /*ma_resource_manager_data_buffer**/ void* pDataBuffer; ma_async_notification* pDoneNotification; ma_fence* pDoneFence; } freeDataBuffer; struct { /*ma_resource_manager_data_stream**/ void* pDataStream; char* pFilePath; /* Allocated when the job is posted, freed by the job thread after loading. */ wchar_t* pFilePathW; /* ^ As above ^. Only used if pFilePath is NULL. */ ma_uint64 initialSeekPoint; ma_async_notification* pInitNotification; /* Signalled after the first two pages have been decoded and frames can be read from the stream. */ ma_fence* pInitFence; } loadDataStream; struct { /*ma_resource_manager_data_stream**/ void* pDataStream; ma_async_notification* pDoneNotification; ma_fence* pDoneFence; } freeDataStream; struct { /*ma_resource_manager_data_stream**/ void* pDataStream; ma_uint32 pageIndex; /* The index of the page to decode into. */ } pageDataStream; struct { /*ma_resource_manager_data_stream**/ void* pDataStream; ma_uint64 frameIndex; } seekDataStream; } resourceManager; /* Device. */ union { union { struct { /*ma_device**/ void* pDevice; /*ma_device_type*/ ma_uint32 deviceType; } reroute; } aaudio; } device; } data; }; MA_API ma_job ma_job_init(ma_uint16 code); MA_API ma_result ma_job_process(ma_job* pJob); /* When set, ma_job_queue_next() will not wait and no semaphore will be signaled in ma_job_queue_post(). ma_job_queue_next() will return MA_NO_DATA_AVAILABLE if nothing is available. This flag should always be used for platforms that do not support multithreading. */ typedef enum { MA_JOB_QUEUE_FLAG_NON_BLOCKING = 0x00000001 } ma_job_queue_flags; typedef struct { ma_uint32 flags; ma_uint32 capacity; /* The maximum number of jobs that can fit in the queue at a time. */ } ma_job_queue_config; MA_API ma_job_queue_config ma_job_queue_config_init(ma_uint32 flags, ma_uint32 capacity); typedef struct { ma_uint32 flags; /* Flags passed in at initialization time. */ ma_uint32 capacity; /* The maximum number of jobs that can fit in the queue at a time. Set by the config. */ MA_ATOMIC(8, ma_uint64) head; /* The first item in the list. Required for removing from the top of the list. */ MA_ATOMIC(8, ma_uint64) tail; /* The last item in the list. Required for appending to the end of the list. */ #ifndef MA_NO_THREADING ma_semaphore sem; /* Only used when MA_JOB_QUEUE_FLAG_NON_BLOCKING is unset. */ #endif ma_slot_allocator allocator; ma_job* pJobs; #ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE ma_spinlock lock; #endif /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_job_queue; MA_API ma_result ma_job_queue_get_heap_size(const ma_job_queue_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_job_queue_init_preallocated(const ma_job_queue_config* pConfig, void* pHeap, ma_job_queue* pQueue); MA_API ma_result ma_job_queue_init(const ma_job_queue_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_job_queue* pQueue); MA_API void ma_job_queue_uninit(ma_job_queue* pQueue, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_job_queue_post(ma_job_queue* pQueue, const ma_job* pJob); MA_API ma_result ma_job_queue_next(ma_job_queue* pQueue, ma_job* pJob); /* Returns MA_CANCELLED if the next job is a quit job. */ /************************************************************************************************************************************************************ ************************************************************************************************************************************************************* DEVICE I/O ========== This section contains the APIs for device playback and capture. Here is where you'll find ma_device_init(), etc. ************************************************************************************************************************************************************* ************************************************************************************************************************************************************/ #ifndef MA_NO_DEVICE_IO /* Some backends are only supported on certain platforms. */ #if defined(MA_WIN32) && !defined(MA_XBOX) #define MA_SUPPORT_WASAPI #if defined(MA_WIN32_DESKTOP) /* DirectSound and WinMM backends are only supported on desktops. */ #define MA_SUPPORT_DSOUND #define MA_SUPPORT_WINMM #define MA_SUPPORT_JACK /* JACK is technically supported on Windows, but I don't know how many people use it in practice... */ #endif #endif #if defined(MA_UNIX) && !defined(MA_ORBIS) && !defined(MA_PROSPERO) #if defined(MA_LINUX) #if !defined(MA_ANDROID) && !defined(MA_EMSCRIPTEN) /* ALSA is not supported on Android. */ #define MA_SUPPORT_ALSA #endif #endif #if !defined(MA_BSD) && !defined(MA_ANDROID) && !defined(MA_EMSCRIPTEN) #define MA_SUPPORT_PULSEAUDIO #define MA_SUPPORT_JACK #endif #if defined(__OpenBSD__) /* <-- Change this to "#if defined(MA_BSD)" to enable sndio on all BSD flavors. */ #define MA_SUPPORT_SNDIO /* sndio is only supported on OpenBSD for now. May be expanded later if there's demand. */ #endif #if defined(__NetBSD__) || defined(__OpenBSD__) #define MA_SUPPORT_AUDIO4 /* Only support audio(4) on platforms with known support. */ #endif #if defined(__FreeBSD__) || defined(__DragonFly__) #define MA_SUPPORT_OSS /* Only support OSS on specific platforms with known support. */ #endif #endif #if defined(MA_ANDROID) #define MA_SUPPORT_AAUDIO #define MA_SUPPORT_OPENSL #endif #if defined(MA_APPLE) #define MA_SUPPORT_COREAUDIO #endif #if defined(MA_EMSCRIPTEN) #define MA_SUPPORT_WEBAUDIO #endif /* All platforms should support custom backends. */ #define MA_SUPPORT_CUSTOM /* Explicitly disable the Null backend for Emscripten because it uses a background thread which is not properly supported right now. */ #if !defined(MA_EMSCRIPTEN) #define MA_SUPPORT_NULL #endif #if defined(MA_SUPPORT_WASAPI) && !defined(MA_NO_WASAPI) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_WASAPI)) #define MA_HAS_WASAPI #endif #if defined(MA_SUPPORT_DSOUND) && !defined(MA_NO_DSOUND) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_DSOUND)) #define MA_HAS_DSOUND #endif #if defined(MA_SUPPORT_WINMM) && !defined(MA_NO_WINMM) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_WINMM)) #define MA_HAS_WINMM #endif #if defined(MA_SUPPORT_ALSA) && !defined(MA_NO_ALSA) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_ALSA)) #define MA_HAS_ALSA #endif #if defined(MA_SUPPORT_PULSEAUDIO) && !defined(MA_NO_PULSEAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_PULSEAUDIO)) #define MA_HAS_PULSEAUDIO #endif #if defined(MA_SUPPORT_JACK) && !defined(MA_NO_JACK) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_JACK)) #define MA_HAS_JACK #endif #if defined(MA_SUPPORT_COREAUDIO) && !defined(MA_NO_COREAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_COREAUDIO)) #define MA_HAS_COREAUDIO #endif #if defined(MA_SUPPORT_SNDIO) && !defined(MA_NO_SNDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_SNDIO)) #define MA_HAS_SNDIO #endif #if defined(MA_SUPPORT_AUDIO4) && !defined(MA_NO_AUDIO4) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_AUDIO4)) #define MA_HAS_AUDIO4 #endif #if defined(MA_SUPPORT_OSS) && !defined(MA_NO_OSS) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_OSS)) #define MA_HAS_OSS #endif #if defined(MA_SUPPORT_AAUDIO) && !defined(MA_NO_AAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_AAUDIO)) #define MA_HAS_AAUDIO #endif #if defined(MA_SUPPORT_OPENSL) && !defined(MA_NO_OPENSL) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_OPENSL)) #define MA_HAS_OPENSL #endif #if defined(MA_SUPPORT_WEBAUDIO) && !defined(MA_NO_WEBAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_WEBAUDIO)) #define MA_HAS_WEBAUDIO #endif #if defined(MA_SUPPORT_CUSTOM) && !defined(MA_NO_CUSTOM) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_CUSTOM)) #define MA_HAS_CUSTOM #endif #if defined(MA_SUPPORT_NULL) && !defined(MA_NO_NULL) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_NULL)) #define MA_HAS_NULL #endif typedef enum { ma_device_state_uninitialized = 0, ma_device_state_stopped = 1, /* The device's default state after initialization. */ ma_device_state_started = 2, /* The device is started and is requesting and/or delivering audio data. */ ma_device_state_starting = 3, /* Transitioning from a stopped state to started. */ ma_device_state_stopping = 4 /* Transitioning from a started state to stopped. */ } ma_device_state; MA_ATOMIC_SAFE_TYPE_DECL(i32, 4, device_state) #ifdef MA_SUPPORT_WASAPI /* We need a IMMNotificationClient object for WASAPI. */ typedef struct { void* lpVtbl; ma_uint32 counter; ma_device* pDevice; } ma_IMMNotificationClient; #endif /* Backend enums must be in priority order. */ typedef enum { ma_backend_wasapi, ma_backend_dsound, ma_backend_winmm, ma_backend_coreaudio, ma_backend_sndio, ma_backend_audio4, ma_backend_oss, ma_backend_pulseaudio, ma_backend_alsa, ma_backend_jack, ma_backend_aaudio, ma_backend_opensl, ma_backend_webaudio, ma_backend_custom, /* <-- Custom backend, with callbacks defined by the context config. */ ma_backend_null /* <-- Must always be the last item. Lowest priority, and used as the terminator for backend enumeration. */ } ma_backend; #define MA_BACKEND_COUNT (ma_backend_null+1) /* Device job thread. This is used by backends that require asynchronous processing of certain operations. It is not used by all backends. The device job thread is made up of a thread and a job queue. You can post a job to the thread with ma_device_job_thread_post(). The thread will do the processing of the job. */ typedef struct { ma_bool32 noThread; /* Set this to true if you want to process jobs yourself. */ ma_uint32 jobQueueCapacity; ma_uint32 jobQueueFlags; } ma_device_job_thread_config; MA_API ma_device_job_thread_config ma_device_job_thread_config_init(void); typedef struct { ma_thread thread; ma_job_queue jobQueue; ma_bool32 _hasThread; } ma_device_job_thread; MA_API ma_result ma_device_job_thread_init(const ma_device_job_thread_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_device_job_thread* pJobThread); MA_API void ma_device_job_thread_uninit(ma_device_job_thread* pJobThread, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_device_job_thread_post(ma_device_job_thread* pJobThread, const ma_job* pJob); MA_API ma_result ma_device_job_thread_next(ma_device_job_thread* pJobThread, ma_job* pJob); /* Device notification types. */ typedef enum { ma_device_notification_type_started, ma_device_notification_type_stopped, ma_device_notification_type_rerouted, ma_device_notification_type_interruption_began, ma_device_notification_type_interruption_ended, ma_device_notification_type_unlocked } ma_device_notification_type; typedef struct { ma_device* pDevice; ma_device_notification_type type; union { struct { int _unused; } started; struct { int _unused; } stopped; struct { int _unused; } rerouted; struct { int _unused; } interruption; } data; } ma_device_notification; /* The notification callback for when the application should be notified of a change to the device. This callback is used for notifying the application of changes such as when the device has started, stopped, rerouted or an interruption has occurred. Note that not all backends will post all notification types. For example, some backends will perform automatic stream routing without any kind of notification to the host program which means miniaudio will never know about it and will never be able to fire the rerouted notification. You should keep this in mind when designing your program. The stopped notification will *not* get fired when a device is rerouted. Parameters ---------- pNotification (in) A pointer to a structure containing information about the event. Use the `pDevice` member of this object to retrieve the relevant device. The `type` member can be used to discriminate against each of the notification types. Remarks ------- Do not restart or uninitialize the device from the callback. Not all notifications will be triggered by all backends, however the started and stopped events should be reliable for all backends. Some backends do not have a good way to detect device stoppages due to unplugging the device which may result in the stopped callback not getting fired. This has been observed with at least one BSD variant. The rerouted notification is fired *after* the reroute has occurred. The stopped notification will *not* get fired when a device is rerouted. The following backends are known to do automatic stream rerouting, but do not have a way to be notified of the change: * DirectSound The interruption notifications are used on mobile platforms for detecting when audio is interrupted due to things like an incoming phone call. Currently this is only implemented on iOS. None of the Android backends will report this notification. */ typedef void (* ma_device_notification_proc)(const ma_device_notification* pNotification); /* The callback for processing audio data from the device. The data callback is fired by miniaudio whenever the device needs to have more data delivered to a playback device, or when a capture device has some data available. This is called as soon as the backend asks for more data which means it may be called with inconsistent frame counts. You cannot assume the callback will be fired with a consistent frame count. Parameters ---------- pDevice (in) A pointer to the relevant device. pOutput (out) A pointer to the output buffer that will receive audio data that will later be played back through the speakers. This will be non-null for a playback or full-duplex device and null for a capture and loopback device. pInput (in) A pointer to the buffer containing input data from a recording device. This will be non-null for a capture, full-duplex or loopback device and null for a playback device. frameCount (in) The number of PCM frames to process. Note that this will not necessarily be equal to what you requested when you initialized the device. The `periodSizeInFrames` and `periodSizeInMilliseconds` members of the device config are just hints, and are not necessarily exactly what you'll get. You must not assume this will always be the same value each time the callback is fired. Remarks ------- You cannot stop and start the device from inside the callback or else you'll get a deadlock. You must also not uninitialize the device from inside the callback. The following APIs cannot be called from inside the callback: ma_device_init() ma_device_init_ex() ma_device_uninit() ma_device_start() ma_device_stop() The proper way to stop the device is to call `ma_device_stop()` from a different thread, normally the main application thread. */ typedef void (* ma_device_data_proc)(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount); /* DEPRECATED. Use ma_device_notification_proc instead. The callback for when the device has been stopped. This will be called when the device is stopped explicitly with `ma_device_stop()` and also called implicitly when the device is stopped through external forces such as being unplugged or an internal error occurring. Parameters ---------- pDevice (in) A pointer to the device that has just stopped. Remarks ------- Do not restart or uninitialize the device from the callback. */ typedef void (* ma_stop_proc)(ma_device* pDevice); /* DEPRECATED. Use ma_device_notification_proc instead. */ typedef enum { ma_device_type_playback = 1, ma_device_type_capture = 2, ma_device_type_duplex = ma_device_type_playback | ma_device_type_capture, /* 3 */ ma_device_type_loopback = 4 } ma_device_type; typedef enum { ma_share_mode_shared = 0, ma_share_mode_exclusive } ma_share_mode; /* iOS/tvOS/watchOS session categories. */ typedef enum { ma_ios_session_category_default = 0, /* AVAudioSessionCategoryPlayAndRecord. */ ma_ios_session_category_none, /* Leave the session category unchanged. */ ma_ios_session_category_ambient, /* AVAudioSessionCategoryAmbient */ ma_ios_session_category_solo_ambient, /* AVAudioSessionCategorySoloAmbient */ ma_ios_session_category_playback, /* AVAudioSessionCategoryPlayback */ ma_ios_session_category_record, /* AVAudioSessionCategoryRecord */ ma_ios_session_category_play_and_record, /* AVAudioSessionCategoryPlayAndRecord */ ma_ios_session_category_multi_route /* AVAudioSessionCategoryMultiRoute */ } ma_ios_session_category; /* iOS/tvOS/watchOS session category options */ typedef enum { ma_ios_session_category_option_mix_with_others = 0x01, /* AVAudioSessionCategoryOptionMixWithOthers */ ma_ios_session_category_option_duck_others = 0x02, /* AVAudioSessionCategoryOptionDuckOthers */ ma_ios_session_category_option_allow_bluetooth = 0x04, /* AVAudioSessionCategoryOptionAllowBluetooth */ ma_ios_session_category_option_default_to_speaker = 0x08, /* AVAudioSessionCategoryOptionDefaultToSpeaker */ ma_ios_session_category_option_interrupt_spoken_audio_and_mix_with_others = 0x11, /* AVAudioSessionCategoryOptionInterruptSpokenAudioAndMixWithOthers */ ma_ios_session_category_option_allow_bluetooth_a2dp = 0x20, /* AVAudioSessionCategoryOptionAllowBluetoothA2DP */ ma_ios_session_category_option_allow_air_play = 0x40, /* AVAudioSessionCategoryOptionAllowAirPlay */ } ma_ios_session_category_option; /* OpenSL stream types. */ typedef enum { ma_opensl_stream_type_default = 0, /* Leaves the stream type unset. */ ma_opensl_stream_type_voice, /* SL_ANDROID_STREAM_VOICE */ ma_opensl_stream_type_system, /* SL_ANDROID_STREAM_SYSTEM */ ma_opensl_stream_type_ring, /* SL_ANDROID_STREAM_RING */ ma_opensl_stream_type_media, /* SL_ANDROID_STREAM_MEDIA */ ma_opensl_stream_type_alarm, /* SL_ANDROID_STREAM_ALARM */ ma_opensl_stream_type_notification /* SL_ANDROID_STREAM_NOTIFICATION */ } ma_opensl_stream_type; /* OpenSL recording presets. */ typedef enum { ma_opensl_recording_preset_default = 0, /* Leaves the input preset unset. */ ma_opensl_recording_preset_generic, /* SL_ANDROID_RECORDING_PRESET_GENERIC */ ma_opensl_recording_preset_camcorder, /* SL_ANDROID_RECORDING_PRESET_CAMCORDER */ ma_opensl_recording_preset_voice_recognition, /* SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION */ ma_opensl_recording_preset_voice_communication, /* SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION */ ma_opensl_recording_preset_voice_unprocessed /* SL_ANDROID_RECORDING_PRESET_UNPROCESSED */ } ma_opensl_recording_preset; /* WASAPI audio thread priority characteristics. */ typedef enum { ma_wasapi_usage_default = 0, ma_wasapi_usage_games, ma_wasapi_usage_pro_audio, } ma_wasapi_usage; /* AAudio usage types. */ typedef enum { ma_aaudio_usage_default = 0, /* Leaves the usage type unset. */ ma_aaudio_usage_media, /* AAUDIO_USAGE_MEDIA */ ma_aaudio_usage_voice_communication, /* AAUDIO_USAGE_VOICE_COMMUNICATION */ ma_aaudio_usage_voice_communication_signalling, /* AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING */ ma_aaudio_usage_alarm, /* AAUDIO_USAGE_ALARM */ ma_aaudio_usage_notification, /* AAUDIO_USAGE_NOTIFICATION */ ma_aaudio_usage_notification_ringtone, /* AAUDIO_USAGE_NOTIFICATION_RINGTONE */ ma_aaudio_usage_notification_event, /* AAUDIO_USAGE_NOTIFICATION_EVENT */ ma_aaudio_usage_assistance_accessibility, /* AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY */ ma_aaudio_usage_assistance_navigation_guidance, /* AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE */ ma_aaudio_usage_assistance_sonification, /* AAUDIO_USAGE_ASSISTANCE_SONIFICATION */ ma_aaudio_usage_game, /* AAUDIO_USAGE_GAME */ ma_aaudio_usage_assitant, /* AAUDIO_USAGE_ASSISTANT */ ma_aaudio_usage_emergency, /* AAUDIO_SYSTEM_USAGE_EMERGENCY */ ma_aaudio_usage_safety, /* AAUDIO_SYSTEM_USAGE_SAFETY */ ma_aaudio_usage_vehicle_status, /* AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS */ ma_aaudio_usage_announcement /* AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT */ } ma_aaudio_usage; /* AAudio content types. */ typedef enum { ma_aaudio_content_type_default = 0, /* Leaves the content type unset. */ ma_aaudio_content_type_speech, /* AAUDIO_CONTENT_TYPE_SPEECH */ ma_aaudio_content_type_music, /* AAUDIO_CONTENT_TYPE_MUSIC */ ma_aaudio_content_type_movie, /* AAUDIO_CONTENT_TYPE_MOVIE */ ma_aaudio_content_type_sonification /* AAUDIO_CONTENT_TYPE_SONIFICATION */ } ma_aaudio_content_type; /* AAudio input presets. */ typedef enum { ma_aaudio_input_preset_default = 0, /* Leaves the input preset unset. */ ma_aaudio_input_preset_generic, /* AAUDIO_INPUT_PRESET_GENERIC */ ma_aaudio_input_preset_camcorder, /* AAUDIO_INPUT_PRESET_CAMCORDER */ ma_aaudio_input_preset_voice_recognition, /* AAUDIO_INPUT_PRESET_VOICE_RECOGNITION */ ma_aaudio_input_preset_voice_communication, /* AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION */ ma_aaudio_input_preset_unprocessed, /* AAUDIO_INPUT_PRESET_UNPROCESSED */ ma_aaudio_input_preset_voice_performance /* AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE */ } ma_aaudio_input_preset; typedef enum { ma_aaudio_allow_capture_default = 0, /* Leaves the allowed capture policy unset. */ ma_aaudio_allow_capture_by_all, /* AAUDIO_ALLOW_CAPTURE_BY_ALL */ ma_aaudio_allow_capture_by_system, /* AAUDIO_ALLOW_CAPTURE_BY_SYSTEM */ ma_aaudio_allow_capture_by_none /* AAUDIO_ALLOW_CAPTURE_BY_NONE */ } ma_aaudio_allowed_capture_policy; typedef union { ma_int64 counter; double counterD; } ma_timer; typedef union { ma_wchar_win32 wasapi[64]; /* WASAPI uses a wchar_t string for identification. */ ma_uint8 dsound[16]; /* DirectSound uses a GUID for identification. */ /*UINT_PTR*/ ma_uint32 winmm; /* When creating a device, WinMM expects a Win32 UINT_PTR for device identification. In practice it's actually just a UINT. */ char alsa[256]; /* ALSA uses a name string for identification. */ char pulse[256]; /* PulseAudio uses a name string for identification. */ int jack; /* JACK always uses default devices. */ char coreaudio[256]; /* Core Audio uses a string for identification. */ char sndio[256]; /* "snd/0", etc. */ char audio4[256]; /* "/dev/audio", etc. */ char oss[64]; /* "dev/dsp0", etc. "dev/dsp" for the default device. */ ma_int32 aaudio; /* AAudio uses a 32-bit integer for identification. */ ma_uint32 opensl; /* OpenSL|ES uses a 32-bit unsigned integer for identification. */ char webaudio[32]; /* Web Audio always uses default devices for now, but if this changes it'll be a GUID. */ union { int i; char s[256]; void* p; } custom; /* The custom backend could be anything. Give them a few options. */ int nullbackend; /* The null backend uses an integer for device IDs. */ } ma_device_id; MA_API ma_bool32 ma_device_id_equal(const ma_device_id* pA, const ma_device_id* pB); typedef struct ma_context_config ma_context_config; typedef struct ma_device_config ma_device_config; typedef struct ma_backend_callbacks ma_backend_callbacks; #define MA_DATA_FORMAT_FLAG_EXCLUSIVE_MODE (1U << 1) /* If set, this is supported in exclusive mode. Otherwise not natively supported by exclusive mode. */ #ifndef MA_MAX_DEVICE_NAME_LENGTH #define MA_MAX_DEVICE_NAME_LENGTH 255 #endif typedef struct { /* Basic info. This is the only information guaranteed to be filled in during device enumeration. */ ma_device_id id; char name[MA_MAX_DEVICE_NAME_LENGTH + 1]; /* +1 for null terminator. */ ma_bool32 isDefault; ma_uint32 nativeDataFormatCount; struct { ma_format format; /* Sample format. If set to ma_format_unknown, all sample formats are supported. */ ma_uint32 channels; /* If set to 0, all channels are supported. */ ma_uint32 sampleRate; /* If set to 0, all sample rates are supported. */ ma_uint32 flags; /* A combination of MA_DATA_FORMAT_FLAG_* flags. */ } nativeDataFormats[/*ma_format_count * ma_standard_sample_rate_count * MA_MAX_CHANNELS*/ 64]; /* Not sure how big to make this. There can be *many* permutations for virtual devices which can support anything. */ } ma_device_info; struct ma_device_config { ma_device_type deviceType; ma_uint32 sampleRate; ma_uint32 periodSizeInFrames; ma_uint32 periodSizeInMilliseconds; ma_uint32 periods; ma_performance_profile performanceProfile; ma_bool8 noPreSilencedOutputBuffer; /* When set to true, the contents of the output buffer passed into the data callback will be left undefined rather than initialized to silence. */ ma_bool8 noClip; /* When set to true, the contents of the output buffer passed into the data callback will not be clipped after returning. Only applies when the playback sample format is f32. */ ma_bool8 noDisableDenormals; /* Do not disable denormals when firing the data callback. */ ma_bool8 noFixedSizedCallback; /* Disables strict fixed-sized data callbacks. Setting this to true will result in the period size being treated only as a hint to the backend. This is an optimization for those who don't need fixed sized callbacks. */ ma_device_data_proc dataCallback; ma_device_notification_proc notificationCallback; ma_stop_proc stopCallback; void* pUserData; ma_resampler_config resampling; struct { const ma_device_id* pDeviceID; ma_format format; ma_uint32 channels; ma_channel* pChannelMap; ma_channel_mix_mode channelMixMode; ma_bool32 calculateLFEFromSpatialChannels; /* When an output LFE channel is present, but no input LFE, set to true to set the output LFE to the average of all spatial channels (LR, FR, etc.). Ignored when an input LFE is present. */ ma_share_mode shareMode; } playback; struct { const ma_device_id* pDeviceID; ma_format format; ma_uint32 channels; ma_channel* pChannelMap; ma_channel_mix_mode channelMixMode; ma_bool32 calculateLFEFromSpatialChannels; /* When an output LFE channel is present, but no input LFE, set to true to set the output LFE to the average of all spatial channels (LR, FR, etc.). Ignored when an input LFE is present. */ ma_share_mode shareMode; } capture; struct { ma_wasapi_usage usage; /* When configured, uses Avrt APIs to set the thread characteristics. */ ma_bool8 noAutoConvertSRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */ ma_bool8 noDefaultQualitySRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY. */ ma_bool8 noAutoStreamRouting; /* Disables automatic stream routing. */ ma_bool8 noHardwareOffloading; /* Disables WASAPI's hardware offloading feature. */ ma_uint32 loopbackProcessID; /* The process ID to include or exclude for loopback mode. Set to 0 to capture audio from all processes. Ignored when an explicit device ID is specified. */ ma_bool8 loopbackProcessExclude; /* When set to true, excludes the process specified by loopbackProcessID. By default, the process will be included. */ } wasapi; struct { ma_bool32 noMMap; /* Disables MMap mode. */ ma_bool32 noAutoFormat; /* Opens the ALSA device with SND_PCM_NO_AUTO_FORMAT. */ ma_bool32 noAutoChannels; /* Opens the ALSA device with SND_PCM_NO_AUTO_CHANNELS. */ ma_bool32 noAutoResample; /* Opens the ALSA device with SND_PCM_NO_AUTO_RESAMPLE. */ } alsa; struct { const char* pStreamNamePlayback; const char* pStreamNameCapture; int channelMap; } pulse; struct { ma_bool32 allowNominalSampleRateChange; /* Desktop only. When enabled, allows changing of the sample rate at the operating system level. */ } coreaudio; struct { ma_opensl_stream_type streamType; ma_opensl_recording_preset recordingPreset; ma_bool32 enableCompatibilityWorkarounds; } opensl; struct { ma_aaudio_usage usage; ma_aaudio_content_type contentType; ma_aaudio_input_preset inputPreset; ma_aaudio_allowed_capture_policy allowedCapturePolicy; ma_bool32 noAutoStartAfterReroute; ma_bool32 enableCompatibilityWorkarounds; ma_bool32 allowSetBufferCapacity; } aaudio; }; /* The callback for handling device enumeration. This is fired from `ma_context_enumerate_devices()`. Parameters ---------- pContext (in) A pointer to the context performing the enumeration. deviceType (in) The type of the device being enumerated. This will always be either `ma_device_type_playback` or `ma_device_type_capture`. pInfo (in) A pointer to a `ma_device_info` containing the ID and name of the enumerated device. Note that this will not include detailed information about the device, only basic information (ID and name). The reason for this is that it would otherwise require opening the backend device to probe for the information which is too inefficient. pUserData (in) The user data pointer passed into `ma_context_enumerate_devices()`. */ typedef ma_bool32 (* ma_enum_devices_callback_proc)(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData); /* Describes some basic details about a playback or capture device. */ typedef struct { const ma_device_id* pDeviceID; ma_share_mode shareMode; ma_format format; ma_uint32 channels; ma_uint32 sampleRate; ma_channel channelMap[MA_MAX_CHANNELS]; ma_uint32 periodSizeInFrames; ma_uint32 periodSizeInMilliseconds; ma_uint32 periodCount; } ma_device_descriptor; /* These are the callbacks required to be implemented for a backend. These callbacks are grouped into two parts: context and device. There is one context to many devices. A device is created from a context. The general flow goes like this: 1) A context is created with `onContextInit()` 1a) Available devices can be enumerated with `onContextEnumerateDevices()` if required. 1b) Detailed information about a device can be queried with `onContextGetDeviceInfo()` if required. 2) A device is created from the context that was created in the first step using `onDeviceInit()`, and optionally a device ID that was selected from device enumeration via `onContextEnumerateDevices()`. 3) A device is started or stopped with `onDeviceStart()` / `onDeviceStop()` 4) Data is delivered to and from the device by the backend. This is always done based on the native format returned by the prior call to `onDeviceInit()`. Conversion between the device's native format and the format requested by the application will be handled by miniaudio internally. Initialization of the context is quite simple. You need to do any necessary initialization of internal objects and then output the callbacks defined in this structure. Once the context has been initialized you can initialize a device. Before doing so, however, the application may want to know which physical devices are available. This is where `onContextEnumerateDevices()` comes in. This is fairly simple. For each device, fire the given callback with, at a minimum, the basic information filled out in `ma_device_info`. When the callback returns `MA_FALSE`, enumeration needs to stop and the `onContextEnumerateDevices()` function returns with a success code. Detailed device information can be retrieved from a device ID using `onContextGetDeviceInfo()`. This takes as input the device type and ID, and on output returns detailed information about the device in `ma_device_info`. The `onContextGetDeviceInfo()` callback must handle the case when the device ID is NULL, in which case information about the default device needs to be retrieved. Once the context has been created and the device ID retrieved (if using anything other than the default device), the device can be created. This is a little bit more complicated than initialization of the context due to its more complicated configuration. When initializing a device, a duplex device may be requested. This means a separate data format needs to be specified for both playback and capture. On input, the data format is set to what the application wants. On output it's set to the native format which should match as closely as possible to the requested format. The conversion between the format requested by the application and the device's native format will be handled internally by miniaudio. On input, if the sample format is set to `ma_format_unknown`, the backend is free to use whatever sample format it desires, so long as it's supported by miniaudio. When the channel count is set to 0, the backend should use the device's native channel count. The same applies for sample rate. For the channel map, the default should be used when `ma_channel_map_is_blank()` returns true (all channels set to `MA_CHANNEL_NONE`). On input, the `periodSizeInFrames` or `periodSizeInMilliseconds` option should always be set. The backend should inspect both of these variables. If `periodSizeInFrames` is set, it should take priority, otherwise it needs to be derived from the period size in milliseconds (`periodSizeInMilliseconds`) and the sample rate, keeping in mind that the sample rate may be 0, in which case the sample rate will need to be determined before calculating the period size in frames. On output, all members of the `ma_device_descriptor` object should be set to a valid value, except for `periodSizeInMilliseconds` which is optional (`periodSizeInFrames` *must* be set). Starting and stopping of the device is done with `onDeviceStart()` and `onDeviceStop()` and should be self-explanatory. If the backend uses asynchronous reading and writing, `onDeviceStart()` and `onDeviceStop()` should always be implemented. The handling of data delivery between the application and the device is the most complicated part of the process. To make this a bit easier, some helper callbacks are available. If the backend uses a blocking read/write style of API, the `onDeviceRead()` and `onDeviceWrite()` callbacks can optionally be implemented. These are blocking and work just like reading and writing from a file. If the backend uses a callback for data delivery, that callback must call `ma_device_handle_backend_data_callback()` from within its callback. This allows miniaudio to then process any necessary data conversion and then pass it to the miniaudio data callback. If the backend requires absolute flexibility with its data delivery, it can optionally implement the `onDeviceDataLoop()` callback which will allow it to implement the logic that will run on the audio thread. This is much more advanced and is completely optional. The audio thread should run data delivery logic in a loop while `ma_device_get_state() == ma_device_state_started` and no errors have been encountered. Do not start or stop the device here. That will be handled from outside the `onDeviceDataLoop()` callback. The invocation of the `onDeviceDataLoop()` callback will be handled by miniaudio. When you start the device, miniaudio will fire this callback. When the device is stopped, the `ma_device_get_state() == ma_device_state_started` condition will fail and the loop will be terminated which will then fall through to the part that stops the device. For an example on how to implement the `onDeviceDataLoop()` callback, look at `ma_device_audio_thread__default_read_write()`. Implement the `onDeviceDataLoopWakeup()` callback if you need a mechanism to wake up the audio thread. If the backend supports an optimized retrieval of device information from an initialized `ma_device` object, it should implement the `onDeviceGetInfo()` callback. This is optional, in which case it will fall back to `onContextGetDeviceInfo()` which is less efficient. */ struct ma_backend_callbacks { ma_result (* onContextInit)(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks); ma_result (* onContextUninit)(ma_context* pContext); ma_result (* onContextEnumerateDevices)(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData); ma_result (* onContextGetDeviceInfo)(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo); ma_result (* onDeviceInit)(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture); ma_result (* onDeviceUninit)(ma_device* pDevice); ma_result (* onDeviceStart)(ma_device* pDevice); ma_result (* onDeviceStop)(ma_device* pDevice); ma_result (* onDeviceRead)(ma_device* pDevice, void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesRead); ma_result (* onDeviceWrite)(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten); ma_result (* onDeviceDataLoop)(ma_device* pDevice); ma_result (* onDeviceDataLoopWakeup)(ma_device* pDevice); ma_result (* onDeviceGetInfo)(ma_device* pDevice, ma_device_type type, ma_device_info* pDeviceInfo); }; struct ma_context_config { ma_log* pLog; ma_thread_priority threadPriority; size_t threadStackSize; void* pUserData; ma_allocation_callbacks allocationCallbacks; struct { ma_handle hWnd; /* HWND. Optional window handle to pass into SetCooperativeLevel(). Will default to the foreground window, and if that fails, the desktop window. */ } dsound; struct { ma_bool32 useVerboseDeviceEnumeration; } alsa; struct { const char* pApplicationName; const char* pServerName; ma_bool32 tryAutoSpawn; /* Enables autospawning of the PulseAudio daemon if necessary. */ } pulse; struct { ma_ios_session_category sessionCategory; ma_uint32 sessionCategoryOptions; ma_bool32 noAudioSessionActivate; /* iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:true] on initialization. */ ma_bool32 noAudioSessionDeactivate; /* iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:false] on uninitialization. */ } coreaudio; struct { const char* pClientName; ma_bool32 tryStartServer; } jack; ma_backend_callbacks custom; }; /* WASAPI specific structure for some commands which must run on a common thread due to bugs in WASAPI. */ typedef struct { int code; ma_event* pEvent; /* This will be signalled when the event is complete. */ union { struct { int _unused; } quit; struct { ma_device_type deviceType; void* pAudioClient; void** ppAudioClientService; ma_result* pResult; /* The result from creating the audio client service. */ } createAudioClient; struct { ma_device* pDevice; ma_device_type deviceType; } releaseAudioClient; } data; } ma_context_command__wasapi; struct ma_context { ma_backend_callbacks callbacks; ma_backend backend; /* DirectSound, ALSA, etc. */ ma_log* pLog; ma_log log; /* Only used if the log is owned by the context. The pLog member will be set to &log in this case. */ ma_thread_priority threadPriority; size_t threadStackSize; void* pUserData; ma_allocation_callbacks allocationCallbacks; ma_mutex deviceEnumLock; /* Used to make ma_context_get_devices() thread safe. */ ma_mutex deviceInfoLock; /* Used to make ma_context_get_device_info() thread safe. */ ma_uint32 deviceInfoCapacity; /* Total capacity of pDeviceInfos. */ ma_uint32 playbackDeviceInfoCount; ma_uint32 captureDeviceInfoCount; ma_device_info* pDeviceInfos; /* Playback devices first, then capture. */ union { #ifdef MA_SUPPORT_WASAPI struct { ma_thread commandThread; ma_mutex commandLock; ma_semaphore commandSem; ma_uint32 commandIndex; ma_uint32 commandCount; ma_context_command__wasapi commands[4]; ma_handle hAvrt; ma_proc AvSetMmThreadCharacteristicsA; ma_proc AvRevertMmThreadcharacteristics; ma_handle hMMDevapi; ma_proc ActivateAudioInterfaceAsync; } wasapi; #endif #ifdef MA_SUPPORT_DSOUND struct { ma_handle hWnd; /* Can be null. */ ma_handle hDSoundDLL; ma_proc DirectSoundCreate; ma_proc DirectSoundEnumerateA; ma_proc DirectSoundCaptureCreate; ma_proc DirectSoundCaptureEnumerateA; } dsound; #endif #ifdef MA_SUPPORT_WINMM struct { ma_handle hWinMM; ma_proc waveOutGetNumDevs; ma_proc waveOutGetDevCapsA; ma_proc waveOutOpen; ma_proc waveOutClose; ma_proc waveOutPrepareHeader; ma_proc waveOutUnprepareHeader; ma_proc waveOutWrite; ma_proc waveOutReset; ma_proc waveInGetNumDevs; ma_proc waveInGetDevCapsA; ma_proc waveInOpen; ma_proc waveInClose; ma_proc waveInPrepareHeader; ma_proc waveInUnprepareHeader; ma_proc waveInAddBuffer; ma_proc waveInStart; ma_proc waveInReset; } winmm; #endif #ifdef MA_SUPPORT_ALSA struct { ma_handle asoundSO; ma_proc snd_pcm_open; ma_proc snd_pcm_close; ma_proc snd_pcm_hw_params_sizeof; ma_proc snd_pcm_hw_params_any; ma_proc snd_pcm_hw_params_set_format; ma_proc snd_pcm_hw_params_set_format_first; ma_proc snd_pcm_hw_params_get_format_mask; ma_proc snd_pcm_hw_params_set_channels; ma_proc snd_pcm_hw_params_set_channels_near; ma_proc snd_pcm_hw_params_set_channels_minmax; ma_proc snd_pcm_hw_params_set_rate_resample; ma_proc snd_pcm_hw_params_set_rate; ma_proc snd_pcm_hw_params_set_rate_near; ma_proc snd_pcm_hw_params_set_rate_minmax; ma_proc snd_pcm_hw_params_set_buffer_size_near; ma_proc snd_pcm_hw_params_set_periods_near; ma_proc snd_pcm_hw_params_set_access; ma_proc snd_pcm_hw_params_get_format; ma_proc snd_pcm_hw_params_get_channels; ma_proc snd_pcm_hw_params_get_channels_min; ma_proc snd_pcm_hw_params_get_channels_max; ma_proc snd_pcm_hw_params_get_rate; ma_proc snd_pcm_hw_params_get_rate_min; ma_proc snd_pcm_hw_params_get_rate_max; ma_proc snd_pcm_hw_params_get_buffer_size; ma_proc snd_pcm_hw_params_get_periods; ma_proc snd_pcm_hw_params_get_access; ma_proc snd_pcm_hw_params_test_format; ma_proc snd_pcm_hw_params_test_channels; ma_proc snd_pcm_hw_params_test_rate; ma_proc snd_pcm_hw_params; ma_proc snd_pcm_sw_params_sizeof; ma_proc snd_pcm_sw_params_current; ma_proc snd_pcm_sw_params_get_boundary; ma_proc snd_pcm_sw_params_set_avail_min; ma_proc snd_pcm_sw_params_set_start_threshold; ma_proc snd_pcm_sw_params_set_stop_threshold; ma_proc snd_pcm_sw_params; ma_proc snd_pcm_format_mask_sizeof; ma_proc snd_pcm_format_mask_test; ma_proc snd_pcm_get_chmap; ma_proc snd_pcm_state; ma_proc snd_pcm_prepare; ma_proc snd_pcm_start; ma_proc snd_pcm_drop; ma_proc snd_pcm_drain; ma_proc snd_pcm_reset; ma_proc snd_device_name_hint; ma_proc snd_device_name_get_hint; ma_proc snd_card_get_index; ma_proc snd_device_name_free_hint; ma_proc snd_pcm_mmap_begin; ma_proc snd_pcm_mmap_commit; ma_proc snd_pcm_recover; ma_proc snd_pcm_readi; ma_proc snd_pcm_writei; ma_proc snd_pcm_avail; ma_proc snd_pcm_avail_update; ma_proc snd_pcm_wait; ma_proc snd_pcm_nonblock; ma_proc snd_pcm_info; ma_proc snd_pcm_info_sizeof; ma_proc snd_pcm_info_get_name; ma_proc snd_pcm_poll_descriptors; ma_proc snd_pcm_poll_descriptors_count; ma_proc snd_pcm_poll_descriptors_revents; ma_proc snd_config_update_free_global; ma_mutex internalDeviceEnumLock; ma_bool32 useVerboseDeviceEnumeration; } alsa; #endif #ifdef MA_SUPPORT_PULSEAUDIO struct { ma_handle pulseSO; ma_proc pa_mainloop_new; ma_proc pa_mainloop_free; ma_proc pa_mainloop_quit; ma_proc pa_mainloop_get_api; ma_proc pa_mainloop_iterate; ma_proc pa_mainloop_wakeup; ma_proc pa_threaded_mainloop_new; ma_proc pa_threaded_mainloop_free; ma_proc pa_threaded_mainloop_start; ma_proc pa_threaded_mainloop_stop; ma_proc pa_threaded_mainloop_lock; ma_proc pa_threaded_mainloop_unlock; ma_proc pa_threaded_mainloop_wait; ma_proc pa_threaded_mainloop_signal; ma_proc pa_threaded_mainloop_accept; ma_proc pa_threaded_mainloop_get_retval; ma_proc pa_threaded_mainloop_get_api; ma_proc pa_threaded_mainloop_in_thread; ma_proc pa_threaded_mainloop_set_name; ma_proc pa_context_new; ma_proc pa_context_unref; ma_proc pa_context_connect; ma_proc pa_context_disconnect; ma_proc pa_context_set_state_callback; ma_proc pa_context_get_state; ma_proc pa_context_get_sink_info_list; ma_proc pa_context_get_source_info_list; ma_proc pa_context_get_sink_info_by_name; ma_proc pa_context_get_source_info_by_name; ma_proc pa_operation_unref; ma_proc pa_operation_get_state; ma_proc pa_channel_map_init_extend; ma_proc pa_channel_map_valid; ma_proc pa_channel_map_compatible; ma_proc pa_stream_new; ma_proc pa_stream_unref; ma_proc pa_stream_connect_playback; ma_proc pa_stream_connect_record; ma_proc pa_stream_disconnect; ma_proc pa_stream_get_state; ma_proc pa_stream_get_sample_spec; ma_proc pa_stream_get_channel_map; ma_proc pa_stream_get_buffer_attr; ma_proc pa_stream_set_buffer_attr; ma_proc pa_stream_get_device_name; ma_proc pa_stream_set_write_callback; ma_proc pa_stream_set_read_callback; ma_proc pa_stream_set_suspended_callback; ma_proc pa_stream_set_moved_callback; ma_proc pa_stream_is_suspended; ma_proc pa_stream_flush; ma_proc pa_stream_drain; ma_proc pa_stream_is_corked; ma_proc pa_stream_cork; ma_proc pa_stream_trigger; ma_proc pa_stream_begin_write; ma_proc pa_stream_write; ma_proc pa_stream_peek; ma_proc pa_stream_drop; ma_proc pa_stream_writable_size; ma_proc pa_stream_readable_size; /*pa_mainloop**/ ma_ptr pMainLoop; /*pa_context**/ ma_ptr pPulseContext; char* pApplicationName; /* Set when the context is initialized. Used by devices for their local pa_context objects. */ char* pServerName; /* Set when the context is initialized. Used by devices for their local pa_context objects. */ } pulse; #endif #ifdef MA_SUPPORT_JACK struct { ma_handle jackSO; ma_proc jack_client_open; ma_proc jack_client_close; ma_proc jack_client_name_size; ma_proc jack_set_process_callback; ma_proc jack_set_buffer_size_callback; ma_proc jack_on_shutdown; ma_proc jack_get_sample_rate; ma_proc jack_get_buffer_size; ma_proc jack_get_ports; ma_proc jack_activate; ma_proc jack_deactivate; ma_proc jack_connect; ma_proc jack_port_register; ma_proc jack_port_name; ma_proc jack_port_get_buffer; ma_proc jack_free; char* pClientName; ma_bool32 tryStartServer; } jack; #endif #ifdef MA_SUPPORT_COREAUDIO struct { ma_handle hCoreFoundation; ma_proc CFStringGetCString; ma_proc CFRelease; ma_handle hCoreAudio; ma_proc AudioObjectGetPropertyData; ma_proc AudioObjectGetPropertyDataSize; ma_proc AudioObjectSetPropertyData; ma_proc AudioObjectAddPropertyListener; ma_proc AudioObjectRemovePropertyListener; ma_handle hAudioUnit; /* Could possibly be set to AudioToolbox on later versions of macOS. */ ma_proc AudioComponentFindNext; ma_proc AudioComponentInstanceDispose; ma_proc AudioComponentInstanceNew; ma_proc AudioOutputUnitStart; ma_proc AudioOutputUnitStop; ma_proc AudioUnitAddPropertyListener; ma_proc AudioUnitGetPropertyInfo; ma_proc AudioUnitGetProperty; ma_proc AudioUnitSetProperty; ma_proc AudioUnitInitialize; ma_proc AudioUnitRender; /*AudioComponent*/ ma_ptr component; ma_bool32 noAudioSessionDeactivate; /* For tracking whether or not the iOS audio session should be explicitly deactivated. Set from the config in ma_context_init__coreaudio(). */ } coreaudio; #endif #ifdef MA_SUPPORT_SNDIO struct { ma_handle sndioSO; ma_proc sio_open; ma_proc sio_close; ma_proc sio_setpar; ma_proc sio_getpar; ma_proc sio_getcap; ma_proc sio_start; ma_proc sio_stop; ma_proc sio_read; ma_proc sio_write; ma_proc sio_onmove; ma_proc sio_nfds; ma_proc sio_pollfd; ma_proc sio_revents; ma_proc sio_eof; ma_proc sio_setvol; ma_proc sio_onvol; ma_proc sio_initpar; } sndio; #endif #ifdef MA_SUPPORT_AUDIO4 struct { int _unused; } audio4; #endif #ifdef MA_SUPPORT_OSS struct { int versionMajor; int versionMinor; } oss; #endif #ifdef MA_SUPPORT_AAUDIO struct { ma_handle hAAudio; /* libaaudio.so */ ma_proc AAudio_createStreamBuilder; ma_proc AAudioStreamBuilder_delete; ma_proc AAudioStreamBuilder_setDeviceId; ma_proc AAudioStreamBuilder_setDirection; ma_proc AAudioStreamBuilder_setSharingMode; ma_proc AAudioStreamBuilder_setFormat; ma_proc AAudioStreamBuilder_setChannelCount; ma_proc AAudioStreamBuilder_setSampleRate; ma_proc AAudioStreamBuilder_setBufferCapacityInFrames; ma_proc AAudioStreamBuilder_setFramesPerDataCallback; ma_proc AAudioStreamBuilder_setDataCallback; ma_proc AAudioStreamBuilder_setErrorCallback; ma_proc AAudioStreamBuilder_setPerformanceMode; ma_proc AAudioStreamBuilder_setUsage; ma_proc AAudioStreamBuilder_setContentType; ma_proc AAudioStreamBuilder_setInputPreset; ma_proc AAudioStreamBuilder_setAllowedCapturePolicy; ma_proc AAudioStreamBuilder_openStream; ma_proc AAudioStream_close; ma_proc AAudioStream_getState; ma_proc AAudioStream_waitForStateChange; ma_proc AAudioStream_getFormat; ma_proc AAudioStream_getChannelCount; ma_proc AAudioStream_getSampleRate; ma_proc AAudioStream_getBufferCapacityInFrames; ma_proc AAudioStream_getFramesPerDataCallback; ma_proc AAudioStream_getFramesPerBurst; ma_proc AAudioStream_requestStart; ma_proc AAudioStream_requestStop; ma_device_job_thread jobThread; /* For processing operations outside of the error callback, specifically device disconnections and rerouting. */ } aaudio; #endif #ifdef MA_SUPPORT_OPENSL struct { ma_handle libOpenSLES; ma_handle SL_IID_ENGINE; ma_handle SL_IID_AUDIOIODEVICECAPABILITIES; ma_handle SL_IID_ANDROIDSIMPLEBUFFERQUEUE; ma_handle SL_IID_RECORD; ma_handle SL_IID_PLAY; ma_handle SL_IID_OUTPUTMIX; ma_handle SL_IID_ANDROIDCONFIGURATION; ma_proc slCreateEngine; } opensl; #endif #ifdef MA_SUPPORT_WEBAUDIO struct { int _unused; } webaudio; #endif #ifdef MA_SUPPORT_NULL struct { int _unused; } null_backend; #endif }; union { #if defined(MA_WIN32) struct { /*HMODULE*/ ma_handle hOle32DLL; ma_proc CoInitialize; ma_proc CoInitializeEx; ma_proc CoUninitialize; ma_proc CoCreateInstance; ma_proc CoTaskMemFree; ma_proc PropVariantClear; ma_proc StringFromGUID2; /*HMODULE*/ ma_handle hUser32DLL; ma_proc GetForegroundWindow; ma_proc GetDesktopWindow; /*HMODULE*/ ma_handle hAdvapi32DLL; ma_proc RegOpenKeyExA; ma_proc RegCloseKey; ma_proc RegQueryValueExA; /*HRESULT*/ long CoInitializeResult; } win32; #endif #ifdef MA_POSIX struct { int _unused; } posix; #endif int _unused; }; }; struct ma_device { ma_context* pContext; ma_device_type type; ma_uint32 sampleRate; ma_atomic_device_state state; /* The state of the device is variable and can change at any time on any thread. Must be used atomically. */ ma_device_data_proc onData; /* Set once at initialization time and should not be changed after. */ ma_device_notification_proc onNotification; /* Set once at initialization time and should not be changed after. */ ma_stop_proc onStop; /* DEPRECATED. Use the notification callback instead. Set once at initialization time and should not be changed after. */ void* pUserData; /* Application defined data. */ ma_mutex startStopLock; ma_event wakeupEvent; ma_event startEvent; ma_event stopEvent; ma_thread thread; ma_result workResult; /* This is set by the worker thread after it's finished doing a job. */ ma_bool8 isOwnerOfContext; /* When set to true, uninitializing the device will also uninitialize the context. Set to true when NULL is passed into ma_device_init(). */ ma_bool8 noPreSilencedOutputBuffer; ma_bool8 noClip; ma_bool8 noDisableDenormals; ma_bool8 noFixedSizedCallback; ma_atomic_float masterVolumeFactor; /* Linear 0..1. Can be read and written simultaneously by different threads. Must be used atomically. */ ma_duplex_rb duplexRB; /* Intermediary buffer for duplex device on asynchronous backends. */ struct { ma_resample_algorithm algorithm; ma_resampling_backend_vtable* pBackendVTable; void* pBackendUserData; struct { ma_uint32 lpfOrder; } linear; } resampling; struct { ma_device_id* pID; /* Set to NULL if using default ID, otherwise set to the address of "id". */ ma_device_id id; /* If using an explicit device, will be set to a copy of the ID used for initialization. Otherwise cleared to 0. */ char name[MA_MAX_DEVICE_NAME_LENGTH + 1]; /* Maybe temporary. Likely to be replaced with a query API. */ ma_share_mode shareMode; /* Set to whatever was passed in when the device was initialized. */ ma_format format; ma_uint32 channels; ma_channel channelMap[MA_MAX_CHANNELS]; ma_format internalFormat; ma_uint32 internalChannels; ma_uint32 internalSampleRate; ma_channel internalChannelMap[MA_MAX_CHANNELS]; ma_uint32 internalPeriodSizeInFrames; ma_uint32 internalPeriods; ma_channel_mix_mode channelMixMode; ma_bool32 calculateLFEFromSpatialChannels; ma_data_converter converter; void* pIntermediaryBuffer; /* For implementing fixed sized buffer callbacks. Will be null if using variable sized callbacks. */ ma_uint32 intermediaryBufferCap; ma_uint32 intermediaryBufferLen; /* How many valid frames are sitting in the intermediary buffer. */ void* pInputCache; /* In external format. Can be null. */ ma_uint64 inputCacheCap; ma_uint64 inputCacheConsumed; ma_uint64 inputCacheRemaining; } playback; struct { ma_device_id* pID; /* Set to NULL if using default ID, otherwise set to the address of "id". */ ma_device_id id; /* If using an explicit device, will be set to a copy of the ID used for initialization. Otherwise cleared to 0. */ char name[MA_MAX_DEVICE_NAME_LENGTH + 1]; /* Maybe temporary. Likely to be replaced with a query API. */ ma_share_mode shareMode; /* Set to whatever was passed in when the device was initialized. */ ma_format format; ma_uint32 channels; ma_channel channelMap[MA_MAX_CHANNELS]; ma_format internalFormat; ma_uint32 internalChannels; ma_uint32 internalSampleRate; ma_channel internalChannelMap[MA_MAX_CHANNELS]; ma_uint32 internalPeriodSizeInFrames; ma_uint32 internalPeriods; ma_channel_mix_mode channelMixMode; ma_bool32 calculateLFEFromSpatialChannels; ma_data_converter converter; void* pIntermediaryBuffer; /* For implementing fixed sized buffer callbacks. Will be null if using variable sized callbacks. */ ma_uint32 intermediaryBufferCap; ma_uint32 intermediaryBufferLen; /* How many valid frames are sitting in the intermediary buffer. */ } capture; union { #ifdef MA_SUPPORT_WASAPI struct { /*IAudioClient**/ ma_ptr pAudioClientPlayback; /*IAudioClient**/ ma_ptr pAudioClientCapture; /*IAudioRenderClient**/ ma_ptr pRenderClient; /*IAudioCaptureClient**/ ma_ptr pCaptureClient; /*IMMDeviceEnumerator**/ ma_ptr pDeviceEnumerator; /* Used for IMMNotificationClient notifications. Required for detecting default device changes. */ ma_IMMNotificationClient notificationClient; /*HANDLE*/ ma_handle hEventPlayback; /* Auto reset. Initialized to signaled. */ /*HANDLE*/ ma_handle hEventCapture; /* Auto reset. Initialized to unsignaled. */ ma_uint32 actualBufferSizeInFramesPlayback; /* Value from GetBufferSize(). internalPeriodSizeInFrames is not set to the _actual_ buffer size when low-latency shared mode is being used due to the way the IAudioClient3 API works. */ ma_uint32 actualBufferSizeInFramesCapture; ma_uint32 originalPeriodSizeInFrames; ma_uint32 originalPeriodSizeInMilliseconds; ma_uint32 originalPeriods; ma_performance_profile originalPerformanceProfile; ma_uint32 periodSizeInFramesPlayback; ma_uint32 periodSizeInFramesCapture; void* pMappedBufferCapture; ma_uint32 mappedBufferCaptureCap; ma_uint32 mappedBufferCaptureLen; void* pMappedBufferPlayback; ma_uint32 mappedBufferPlaybackCap; ma_uint32 mappedBufferPlaybackLen; ma_atomic_bool32 isStartedCapture; /* Can be read and written simultaneously across different threads. Must be used atomically, and must be 32-bit. */ ma_atomic_bool32 isStartedPlayback; /* Can be read and written simultaneously across different threads. Must be used atomically, and must be 32-bit. */ ma_uint32 loopbackProcessID; ma_bool8 loopbackProcessExclude; ma_bool8 noAutoConvertSRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */ ma_bool8 noDefaultQualitySRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY. */ ma_bool8 noHardwareOffloading; ma_bool8 allowCaptureAutoStreamRouting; ma_bool8 allowPlaybackAutoStreamRouting; ma_bool8 isDetachedPlayback; ma_bool8 isDetachedCapture; ma_wasapi_usage usage; void* hAvrtHandle; ma_mutex rerouteLock; } wasapi; #endif #ifdef MA_SUPPORT_DSOUND struct { /*LPDIRECTSOUND*/ ma_ptr pPlayback; /*LPDIRECTSOUNDBUFFER*/ ma_ptr pPlaybackPrimaryBuffer; /*LPDIRECTSOUNDBUFFER*/ ma_ptr pPlaybackBuffer; /*LPDIRECTSOUNDCAPTURE*/ ma_ptr pCapture; /*LPDIRECTSOUNDCAPTUREBUFFER*/ ma_ptr pCaptureBuffer; } dsound; #endif #ifdef MA_SUPPORT_WINMM struct { /*HWAVEOUT*/ ma_handle hDevicePlayback; /*HWAVEIN*/ ma_handle hDeviceCapture; /*HANDLE*/ ma_handle hEventPlayback; /*HANDLE*/ ma_handle hEventCapture; ma_uint32 fragmentSizeInFrames; ma_uint32 iNextHeaderPlayback; /* [0,periods). Used as an index into pWAVEHDRPlayback. */ ma_uint32 iNextHeaderCapture; /* [0,periods). Used as an index into pWAVEHDRCapture. */ ma_uint32 headerFramesConsumedPlayback; /* The number of PCM frames consumed in the buffer in pWAVEHEADER[iNextHeader]. */ ma_uint32 headerFramesConsumedCapture; /* ^^^ */ /*WAVEHDR**/ ma_uint8* pWAVEHDRPlayback; /* One instantiation for each period. */ /*WAVEHDR**/ ma_uint8* pWAVEHDRCapture; /* One instantiation for each period. */ ma_uint8* pIntermediaryBufferPlayback; ma_uint8* pIntermediaryBufferCapture; ma_uint8* _pHeapData; /* Used internally and is used for the heap allocated data for the intermediary buffer and the WAVEHDR structures. */ } winmm; #endif #ifdef MA_SUPPORT_ALSA struct { /*snd_pcm_t**/ ma_ptr pPCMPlayback; /*snd_pcm_t**/ ma_ptr pPCMCapture; /*struct pollfd**/ void* pPollDescriptorsPlayback; /*struct pollfd**/ void* pPollDescriptorsCapture; int pollDescriptorCountPlayback; int pollDescriptorCountCapture; int wakeupfdPlayback; /* eventfd for waking up from poll() when the playback device is stopped. */ int wakeupfdCapture; /* eventfd for waking up from poll() when the capture device is stopped. */ ma_bool8 isUsingMMapPlayback; ma_bool8 isUsingMMapCapture; } alsa; #endif #ifdef MA_SUPPORT_PULSEAUDIO struct { /*pa_mainloop**/ ma_ptr pMainLoop; /*pa_context**/ ma_ptr pPulseContext; /*pa_stream**/ ma_ptr pStreamPlayback; /*pa_stream**/ ma_ptr pStreamCapture; } pulse; #endif #ifdef MA_SUPPORT_JACK struct { /*jack_client_t**/ ma_ptr pClient; /*jack_port_t**/ ma_ptr* ppPortsPlayback; /*jack_port_t**/ ma_ptr* ppPortsCapture; float* pIntermediaryBufferPlayback; /* Typed as a float because JACK is always floating point. */ float* pIntermediaryBufferCapture; } jack; #endif #ifdef MA_SUPPORT_COREAUDIO struct { ma_uint32 deviceObjectIDPlayback; ma_uint32 deviceObjectIDCapture; /*AudioUnit*/ ma_ptr audioUnitPlayback; /*AudioUnit*/ ma_ptr audioUnitCapture; /*AudioBufferList**/ ma_ptr pAudioBufferList; /* Only used for input devices. */ ma_uint32 audioBufferCapInFrames; /* Only used for input devices. The capacity in frames of each buffer in pAudioBufferList. */ ma_event stopEvent; ma_uint32 originalPeriodSizeInFrames; ma_uint32 originalPeriodSizeInMilliseconds; ma_uint32 originalPeriods; ma_performance_profile originalPerformanceProfile; ma_bool32 isDefaultPlaybackDevice; ma_bool32 isDefaultCaptureDevice; ma_bool32 isSwitchingPlaybackDevice; /* <-- Set to true when the default device has changed and miniaudio is in the process of switching. */ ma_bool32 isSwitchingCaptureDevice; /* <-- Set to true when the default device has changed and miniaudio is in the process of switching. */ void* pNotificationHandler; /* Only used on mobile platforms. Obj-C object for handling route changes. */ } coreaudio; #endif #ifdef MA_SUPPORT_SNDIO struct { ma_ptr handlePlayback; ma_ptr handleCapture; ma_bool32 isStartedPlayback; ma_bool32 isStartedCapture; } sndio; #endif #ifdef MA_SUPPORT_AUDIO4 struct { int fdPlayback; int fdCapture; } audio4; #endif #ifdef MA_SUPPORT_OSS struct { int fdPlayback; int fdCapture; } oss; #endif #ifdef MA_SUPPORT_AAUDIO struct { /*AAudioStream**/ ma_ptr pStreamPlayback; /*AAudioStream**/ ma_ptr pStreamCapture; ma_mutex rerouteLock; ma_atomic_bool32 isTearingDown; ma_aaudio_usage usage; ma_aaudio_content_type contentType; ma_aaudio_input_preset inputPreset; ma_aaudio_allowed_capture_policy allowedCapturePolicy; ma_bool32 noAutoStartAfterReroute; } aaudio; #endif #ifdef MA_SUPPORT_OPENSL struct { /*SLObjectItf*/ ma_ptr pOutputMixObj; /*SLOutputMixItf*/ ma_ptr pOutputMix; /*SLObjectItf*/ ma_ptr pAudioPlayerObj; /*SLPlayItf*/ ma_ptr pAudioPlayer; /*SLObjectItf*/ ma_ptr pAudioRecorderObj; /*SLRecordItf*/ ma_ptr pAudioRecorder; /*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueuePlayback; /*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueueCapture; ma_bool32 isDrainingCapture; ma_bool32 isDrainingPlayback; ma_uint32 currentBufferIndexPlayback; ma_uint32 currentBufferIndexCapture; ma_uint8* pBufferPlayback; /* This is malloc()'d and is used for storing audio data. Typed as ma_uint8 for easy offsetting. */ ma_uint8* pBufferCapture; } opensl; #endif #ifdef MA_SUPPORT_WEBAUDIO struct { /* AudioWorklets path. */ /* EMSCRIPTEN_WEBAUDIO_T */ int audioContext; /* EMSCRIPTEN_WEBAUDIO_T */ int audioWorklet; float* pIntermediaryBuffer; void* pStackBuffer; ma_result initResult; /* Set to MA_BUSY while initialization is in progress. */ int deviceIndex; /* We store the device in a list on the JavaScript side. This is used to map our C object to the JS object. */ } webaudio; #endif #ifdef MA_SUPPORT_NULL struct { ma_thread deviceThread; ma_event operationEvent; ma_event operationCompletionEvent; ma_semaphore operationSemaphore; ma_uint32 operation; ma_result operationResult; ma_timer timer; double priorRunTime; ma_uint32 currentPeriodFramesRemainingPlayback; ma_uint32 currentPeriodFramesRemainingCapture; ma_uint64 lastProcessedFramePlayback; ma_uint64 lastProcessedFrameCapture; ma_atomic_bool32 isStarted; /* Read and written by multiple threads. Must be used atomically, and must be 32-bit for compiler compatibility. */ } null_device; #endif }; }; #if defined(_MSC_VER) && !defined(__clang__) #pragma warning(pop) #elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))) #pragma GCC diagnostic pop /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */ #endif /* Initializes a `ma_context_config` object. Return Value ------------ A `ma_context_config` initialized to defaults. Remarks ------- You must always use this to initialize the default state of the `ma_context_config` object. Not using this will result in your program breaking when miniaudio is updated and new members are added to `ma_context_config`. It also sets logical defaults. You can override members of the returned object by changing it's members directly. See Also -------- ma_context_init() */ MA_API ma_context_config ma_context_config_init(void); /* Initializes a context. The context is used for selecting and initializing an appropriate backend and to represent the backend at a more global level than that of an individual device. There is one context to many devices, and a device is created from a context. A context is required to enumerate devices. Parameters ---------- backends (in, optional) A list of backends to try initializing, in priority order. Can be NULL, in which case it uses default priority order. backendCount (in, optional) The number of items in `backend`. Ignored if `backend` is NULL. pConfig (in, optional) The context configuration. pContext (in) A pointer to the context object being initialized. Return Value ------------ MA_SUCCESS if successful; any other error code otherwise. Thread Safety ------------- Unsafe. Do not call this function across multiple threads as some backends read and write to global state. Remarks ------- When `backends` is NULL, the default priority order will be used. Below is a list of backends in priority order: |-------------|-----------------------|--------------------------------------------------------| | Name | Enum Name | Supported Operating Systems | |-------------|-----------------------|--------------------------------------------------------| | WASAPI | ma_backend_wasapi | Windows Vista+ | | DirectSound | ma_backend_dsound | Windows XP+ | | WinMM | ma_backend_winmm | Windows XP+ (may work on older versions, but untested) | | Core Audio | ma_backend_coreaudio | macOS, iOS | | ALSA | ma_backend_alsa | Linux | | PulseAudio | ma_backend_pulseaudio | Cross Platform (disabled on Windows, BSD and Android) | | JACK | ma_backend_jack | Cross Platform (disabled on BSD and Android) | | sndio | ma_backend_sndio | OpenBSD | | audio(4) | ma_backend_audio4 | NetBSD, OpenBSD | | OSS | ma_backend_oss | FreeBSD | | AAudio | ma_backend_aaudio | Android 8+ | | OpenSL|ES | ma_backend_opensl | Android (API level 16+) | | Web Audio | ma_backend_webaudio | Web (via Emscripten) | | Null | ma_backend_null | Cross Platform (not used on Web) | |-------------|-----------------------|--------------------------------------------------------| The context can be configured via the `pConfig` argument. The config object is initialized with `ma_context_config_init()`. Individual configuration settings can then be set directly on the structure. Below are the members of the `ma_context_config` object. pLog A pointer to the `ma_log` to post log messages to. Can be NULL if the application does not require logging. See the `ma_log` API for details on how to use the logging system. threadPriority The desired priority to use for the audio thread. Allowable values include the following: |--------------------------------------| | Thread Priority | |--------------------------------------| | ma_thread_priority_idle | | ma_thread_priority_lowest | | ma_thread_priority_low | | ma_thread_priority_normal | | ma_thread_priority_high | | ma_thread_priority_highest (default) | | ma_thread_priority_realtime | | ma_thread_priority_default | |--------------------------------------| threadStackSize The desired size of the stack for the audio thread. Defaults to the operating system's default. pUserData A pointer to application-defined data. This can be accessed from the context object directly such as `context.pUserData`. allocationCallbacks Structure containing custom allocation callbacks. Leaving this at defaults will cause it to use MA_MALLOC, MA_REALLOC and MA_FREE. These allocation callbacks will be used for anything tied to the context, including devices. alsa.useVerboseDeviceEnumeration ALSA will typically enumerate many different devices which can be intrusive and not user-friendly. To combat this, miniaudio will enumerate only unique card/device pairs by default. The problem with this is that you lose a bit of flexibility and control. Setting alsa.useVerboseDeviceEnumeration makes it so the ALSA backend includes all devices. Defaults to false. pulse.pApplicationName PulseAudio only. The application name to use when initializing the PulseAudio context with `pa_context_new()`. pulse.pServerName PulseAudio only. The name of the server to connect to with `pa_context_connect()`. pulse.tryAutoSpawn PulseAudio only. Whether or not to try automatically starting the PulseAudio daemon. Defaults to false. If you set this to true, keep in mind that miniaudio uses a trial and error method to find the most appropriate backend, and this will result in the PulseAudio daemon starting which may be intrusive for the end user. coreaudio.sessionCategory iOS only. The session category to use for the shared AudioSession instance. Below is a list of allowable values and their Core Audio equivalents. |-----------------------------------------|-------------------------------------| | miniaudio Token | Core Audio Token | |-----------------------------------------|-------------------------------------| | ma_ios_session_category_ambient | AVAudioSessionCategoryAmbient | | ma_ios_session_category_solo_ambient | AVAudioSessionCategorySoloAmbient | | ma_ios_session_category_playback | AVAudioSessionCategoryPlayback | | ma_ios_session_category_record | AVAudioSessionCategoryRecord | | ma_ios_session_category_play_and_record | AVAudioSessionCategoryPlayAndRecord | | ma_ios_session_category_multi_route | AVAudioSessionCategoryMultiRoute | | ma_ios_session_category_none | AVAudioSessionCategoryAmbient | | ma_ios_session_category_default | AVAudioSessionCategoryAmbient | |-----------------------------------------|-------------------------------------| coreaudio.sessionCategoryOptions iOS only. Session category options to use with the shared AudioSession instance. Below is a list of allowable values and their Core Audio equivalents. |---------------------------------------------------------------------------|------------------------------------------------------------------| | miniaudio Token | Core Audio Token | |---------------------------------------------------------------------------|------------------------------------------------------------------| | ma_ios_session_category_option_mix_with_others | AVAudioSessionCategoryOptionMixWithOthers | | ma_ios_session_category_option_duck_others | AVAudioSessionCategoryOptionDuckOthers | | ma_ios_session_category_option_allow_bluetooth | AVAudioSessionCategoryOptionAllowBluetooth | | ma_ios_session_category_option_default_to_speaker | AVAudioSessionCategoryOptionDefaultToSpeaker | | ma_ios_session_category_option_interrupt_spoken_audio_and_mix_with_others | AVAudioSessionCategoryOptionInterruptSpokenAudioAndMixWithOthers | | ma_ios_session_category_option_allow_bluetooth_a2dp | AVAudioSessionCategoryOptionAllowBluetoothA2DP | | ma_ios_session_category_option_allow_air_play | AVAudioSessionCategoryOptionAllowAirPlay | |---------------------------------------------------------------------------|------------------------------------------------------------------| coreaudio.noAudioSessionActivate iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:true] on initialization. coreaudio.noAudioSessionDeactivate iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:false] on uninitialization. jack.pClientName The name of the client to pass to `jack_client_open()`. jack.tryStartServer Whether or not to try auto-starting the JACK server. Defaults to false. It is recommended that only a single context is active at any given time because it's a bulky data structure which performs run-time linking for the relevant backends every time it's initialized. The location of the context cannot change throughout it's lifetime. Consider allocating the `ma_context` object with `malloc()` if this is an issue. The reason for this is that a pointer to the context is stored in the `ma_device` structure. Example 1 - Default Initialization ---------------------------------- The example below shows how to initialize the context using the default configuration. ```c ma_context context; ma_result result = ma_context_init(NULL, 0, NULL, &context); if (result != MA_SUCCESS) { // Error. } ``` Example 2 - Custom Configuration -------------------------------- The example below shows how to initialize the context using custom backend priorities and a custom configuration. In this hypothetical example, the program wants to prioritize ALSA over PulseAudio on Linux. They also want to avoid using the WinMM backend on Windows because it's latency is too high. They also want an error to be returned if no valid backend is available which they achieve by excluding the Null backend. For the configuration, the program wants to capture any log messages so they can, for example, route it to a log file and user interface. ```c ma_backend backends[] = { ma_backend_alsa, ma_backend_pulseaudio, ma_backend_wasapi, ma_backend_dsound }; ma_log log; ma_log_init(&log); ma_log_register_callback(&log, ma_log_callback_init(my_log_callbac, pMyLogUserData)); ma_context_config config = ma_context_config_init(); config.pLog = &log; // Specify a custom log object in the config so any logs that are posted from ma_context_init() are captured. ma_context context; ma_result result = ma_context_init(backends, sizeof(backends)/sizeof(backends[0]), &config, &context); if (result != MA_SUCCESS) { // Error. if (result == MA_NO_BACKEND) { // Couldn't find an appropriate backend. } } // You could also attach a log callback post-initialization: ma_log_register_callback(ma_context_get_log(&context), ma_log_callback_init(my_log_callback, pMyLogUserData)); ``` See Also -------- ma_context_config_init() ma_context_uninit() */ MA_API ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pConfig, ma_context* pContext); /* Uninitializes a context. Return Value ------------ MA_SUCCESS if successful; any other error code otherwise. Thread Safety ------------- Unsafe. Do not call this function across multiple threads as some backends read and write to global state. Remarks ------- Results are undefined if you call this while any device created by this context is still active. See Also -------- ma_context_init() */ MA_API ma_result ma_context_uninit(ma_context* pContext); /* Retrieves the size of the ma_context object. This is mainly for the purpose of bindings to know how much memory to allocate. */ MA_API size_t ma_context_sizeof(void); /* Retrieves a pointer to the log object associated with this context. Remarks ------- Pass the returned pointer to `ma_log_post()`, `ma_log_postv()` or `ma_log_postf()` to post a log message. You can attach your own logging callback to the log with `ma_log_register_callback()` Return Value ------------ A pointer to the `ma_log` object that the context uses to post log messages. If some error occurs, NULL will be returned. */ MA_API ma_log* ma_context_get_log(ma_context* pContext); /* Enumerates over every device (both playback and capture). This is a lower-level enumeration function to the easier to use `ma_context_get_devices()`. Use `ma_context_enumerate_devices()` if you would rather not incur an internal heap allocation, or it simply suits your code better. Note that this only retrieves the ID and name/description of the device. The reason for only retrieving basic information is that it would otherwise require opening the backend device in order to probe it for more detailed information which can be inefficient. Consider using `ma_context_get_device_info()` for this, but don't call it from within the enumeration callback. Returning false from the callback will stop enumeration. Returning true will continue enumeration. Parameters ---------- pContext (in) A pointer to the context performing the enumeration. callback (in) The callback to fire for each enumerated device. pUserData (in) A pointer to application-defined data passed to the callback. Return Value ------------ MA_SUCCESS if successful; any other error code otherwise. Thread Safety ------------- Safe. This is guarded using a simple mutex lock. Remarks ------- Do _not_ assume the first enumerated device of a given type is the default device. Some backends and platforms may only support default playback and capture devices. In general, you should not do anything complicated from within the callback. In particular, do not try initializing a device from within the callback. Also, do not try to call `ma_context_get_device_info()` from within the callback. Consider using `ma_context_get_devices()` for a simpler and safer API, albeit at the expense of an internal heap allocation. Example 1 - Simple Enumeration ------------------------------ ma_bool32 ma_device_enum_callback(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData) { printf("Device Name: %s\n", pInfo->name); return MA_TRUE; } ma_result result = ma_context_enumerate_devices(&context, my_device_enum_callback, pMyUserData); if (result != MA_SUCCESS) { // Error. } See Also -------- ma_context_get_devices() */ MA_API ma_result ma_context_enumerate_devices(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData); /* Retrieves basic information about every active playback and/or capture device. This function will allocate memory internally for the device lists and return a pointer to them through the `ppPlaybackDeviceInfos` and `ppCaptureDeviceInfos` parameters. If you do not want to incur the overhead of these allocations consider using `ma_context_enumerate_devices()` which will instead use a callback. Note that this only retrieves the ID and name/description of the device. The reason for only retrieving basic information is that it would otherwise require opening the backend device in order to probe it for more detailed information which can be inefficient. Consider using `ma_context_get_device_info()` for this, but don't call it from within the enumeration callback. Parameters ---------- pContext (in) A pointer to the context performing the enumeration. ppPlaybackDeviceInfos (out) A pointer to a pointer that will receive the address of a buffer containing the list of `ma_device_info` structures for playback devices. pPlaybackDeviceCount (out) A pointer to an unsigned integer that will receive the number of playback devices. ppCaptureDeviceInfos (out) A pointer to a pointer that will receive the address of a buffer containing the list of `ma_device_info` structures for capture devices. pCaptureDeviceCount (out) A pointer to an unsigned integer that will receive the number of capture devices. Return Value ------------ MA_SUCCESS if successful; any other error code otherwise. Thread Safety ------------- Unsafe. Since each call to this function invalidates the pointers from the previous call, you should not be calling this simultaneously across multiple threads. Instead, you need to make a copy of the returned data with your own higher level synchronization. Remarks ------- It is _not_ safe to assume the first device in the list is the default device. You can pass in NULL for the playback or capture lists in which case they'll be ignored. The returned pointers will become invalid upon the next call this this function, or when the context is uninitialized. Do not free the returned pointers. See Also -------- ma_context_enumerate_devices() */ MA_API ma_result ma_context_get_devices(ma_context* pContext, ma_device_info** ppPlaybackDeviceInfos, ma_uint32* pPlaybackDeviceCount, ma_device_info** ppCaptureDeviceInfos, ma_uint32* pCaptureDeviceCount); /* Retrieves information about a device of the given type, with the specified ID and share mode. Parameters ---------- pContext (in) A pointer to the context performing the query. deviceType (in) The type of the device being queried. Must be either `ma_device_type_playback` or `ma_device_type_capture`. pDeviceID (in) The ID of the device being queried. pDeviceInfo (out) A pointer to the `ma_device_info` structure that will receive the device information. Return Value ------------ MA_SUCCESS if successful; any other error code otherwise. Thread Safety ------------- Safe. This is guarded using a simple mutex lock. Remarks ------- Do _not_ call this from within the `ma_context_enumerate_devices()` callback. It's possible for a device to have different information and capabilities depending on whether or not it's opened in shared or exclusive mode. For example, in shared mode, WASAPI always uses floating point samples for mixing, but in exclusive mode it can be anything. Therefore, this function allows you to specify which share mode you want information for. Note that not all backends and devices support shared or exclusive mode, in which case this function will fail if the requested share mode is unsupported. This leaves pDeviceInfo unmodified in the result of an error. */ MA_API ma_result ma_context_get_device_info(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo); /* Determines if the given context supports loopback mode. Parameters ---------- pContext (in) A pointer to the context getting queried. Return Value ------------ MA_TRUE if the context supports loopback mode; MA_FALSE otherwise. */ MA_API ma_bool32 ma_context_is_loopback_supported(ma_context* pContext); /* Initializes a device config with default settings. Parameters ---------- deviceType (in) The type of the device this config is being initialized for. This must set to one of the following: |-------------------------| | Device Type | |-------------------------| | ma_device_type_playback | | ma_device_type_capture | | ma_device_type_duplex | | ma_device_type_loopback | |-------------------------| Return Value ------------ A new device config object with default settings. You will typically want to adjust the config after this function returns. See remarks. Thread Safety ------------- Safe. Callback Safety --------------- Safe, but don't try initializing a device in a callback. Remarks ------- The returned config will be initialized to defaults. You will normally want to customize a few variables before initializing the device. See Example 1 for a typical configuration which sets the sample format, channel count, sample rate, data callback and user data. These are usually things you will want to change before initializing the device. See `ma_device_init()` for details on specific configuration options. Example 1 - Simple Configuration -------------------------------- The example below is what a program will typically want to configure for each device at a minimum. Notice how `ma_device_config_init()` is called first, and then the returned object is modified directly. This is important because it ensures that your program continues to work as new configuration options are added to the `ma_device_config` structure. ```c ma_device_config config = ma_device_config_init(ma_device_type_playback); config.playback.format = ma_format_f32; config.playback.channels = 2; config.sampleRate = 48000; config.dataCallback = ma_data_callback; config.pUserData = pMyUserData; ``` See Also -------- ma_device_init() ma_device_init_ex() */ MA_API ma_device_config ma_device_config_init(ma_device_type deviceType); /* Initializes a device. A device represents a physical audio device. The idea is you send or receive audio data from the device to either play it back through a speaker, or capture it from a microphone. Whether or not you should send or receive data from the device (or both) depends on the type of device you are initializing which can be playback, capture, full-duplex or loopback. (Note that loopback mode is only supported on select backends.) Sending and receiving audio data to and from the device is done via a callback which is fired by miniaudio at periodic time intervals. The frequency at which data is delivered to and from a device depends on the size of its period. The size of the period can be defined in terms of PCM frames or milliseconds, whichever is more convenient. Generally speaking, the smaller the period, the lower the latency at the expense of higher CPU usage and increased risk of glitching due to the more frequent and granular data deliver intervals. The size of a period will depend on your requirements, but miniaudio's defaults should work fine for most scenarios. If you're building a game you should leave this fairly small, whereas if you're building a simple media player you can make it larger. Note that the period size you request is actually just a hint - miniaudio will tell the backend what you want, but the backend is ultimately responsible for what it gives you. You cannot assume you will get exactly what you ask for. When delivering data to and from a device you need to make sure it's in the correct format which you can set through the device configuration. You just set the format that you want to use and miniaudio will perform all of the necessary conversion for you internally. When delivering data to and from the callback you can assume the format is the same as what you requested when you initialized the device. See Remarks for more details on miniaudio's data conversion pipeline. Parameters ---------- pContext (in, optional) A pointer to the context that owns the device. This can be null, in which case it creates a default context internally. pConfig (in) A pointer to the device configuration. Cannot be null. See remarks for details. pDevice (out) A pointer to the device object being initialized. Return Value ------------ MA_SUCCESS if successful; any other error code otherwise. Thread Safety ------------- Unsafe. It is not safe to call this function simultaneously for different devices because some backends depend on and mutate global state. The same applies to calling this at the same time as `ma_device_uninit()`. Callback Safety --------------- Unsafe. It is not safe to call this inside any callback. Remarks ------- Setting `pContext` to NULL will result in miniaudio creating a default context internally and is equivalent to passing in a context initialized like so: ```c ma_context_init(NULL, 0, NULL, &context); ``` Do not set `pContext` to NULL if you are needing to open multiple devices. You can, however, use NULL when initializing the first device, and then use device.pContext for the initialization of other devices. The device can be configured via the `pConfig` argument. The config object is initialized with `ma_device_config_init()`. Individual configuration settings can then be set directly on the structure. Below are the members of the `ma_device_config` object. deviceType Must be `ma_device_type_playback`, `ma_device_type_capture`, `ma_device_type_duplex` of `ma_device_type_loopback`. sampleRate The sample rate, in hertz. The most common sample rates are 48000 and 44100. Setting this to 0 will use the device's native sample rate. periodSizeInFrames The desired size of a period in PCM frames. If this is 0, `periodSizeInMilliseconds` will be used instead. If both are 0 the default buffer size will be used depending on the selected performance profile. This value affects latency. See below for details. periodSizeInMilliseconds The desired size of a period in milliseconds. If this is 0, `periodSizeInFrames` will be used instead. If both are 0 the default buffer size will be used depending on the selected performance profile. The value affects latency. See below for details. periods The number of periods making up the device's entire buffer. The total buffer size is `periodSizeInFrames` or `periodSizeInMilliseconds` multiplied by this value. This is just a hint as backends will be the ones who ultimately decide how your periods will be configured. performanceProfile A hint to miniaudio as to the performance requirements of your program. Can be either `ma_performance_profile_low_latency` (default) or `ma_performance_profile_conservative`. This mainly affects the size of default buffers and can usually be left at its default value. noPreSilencedOutputBuffer When set to true, the contents of the output buffer passed into the data callback will be left undefined. When set to false (default), the contents of the output buffer will be cleared the zero. You can use this to avoid the overhead of zeroing out the buffer if you can guarantee that your data callback will write to every sample in the output buffer, or if you are doing your own clearing. noClip When set to true, the contents of the output buffer are left alone after returning and it will be left up to the backend itself to decide whether or not to clip. When set to false (default), the contents of the output buffer passed into the data callback will be clipped after returning. This only applies when the playback sample format is f32. noDisableDenormals By default, miniaudio will disable denormals when the data callback is called. Setting this to true will prevent the disabling of denormals. noFixedSizedCallback Allows miniaudio to fire the data callback with any frame count. When this is set to false (the default), the data callback will be fired with a consistent frame count as specified by `periodSizeInFrames` or `periodSizeInMilliseconds`. When set to true, miniaudio will fire the callback with whatever the backend requests, which could be anything. dataCallback The callback to fire whenever data is ready to be delivered to or from the device. notificationCallback The callback to fire when something has changed with the device, such as whether or not it has been started or stopped. pUserData The user data pointer to use with the device. You can access this directly from the device object like `device.pUserData`. resampling.algorithm The resampling algorithm to use when miniaudio needs to perform resampling between the rate specified by `sampleRate` and the device's native rate. The default value is `ma_resample_algorithm_linear`, and the quality can be configured with `resampling.linear.lpfOrder`. resampling.pBackendVTable A pointer to an optional vtable that can be used for plugging in a custom resampler. resampling.pBackendUserData A pointer that will passed to callbacks in pBackendVTable. resampling.linear.lpfOrder The linear resampler applies a low-pass filter as part of its processing for anti-aliasing. This setting controls the order of the filter. The higher the value, the better the quality, in general. Setting this to 0 will disable low-pass filtering altogether. The maximum value is `MA_MAX_FILTER_ORDER`. The default value is `min(4, MA_MAX_FILTER_ORDER)`. playback.pDeviceID A pointer to a `ma_device_id` structure containing the ID of the playback device to initialize. Setting this NULL (default) will use the system's default playback device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration. playback.format The sample format to use for playback. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after initialization from the device object directly with `device.playback.format`. playback.channels The number of channels to use for playback. When set to 0 the device's native channel count will be used. This can be retrieved after initialization from the device object directly with `device.playback.channels`. playback.pChannelMap The channel map to use for playback. When left empty, the device's native channel map will be used. This can be retrieved after initialization from the device object direct with `device.playback.pChannelMap`. When set, the buffer should contain `channels` items. playback.shareMode The preferred share mode to use for playback. Can be either `ma_share_mode_shared` (default) or `ma_share_mode_exclusive`. Note that if you specify exclusive mode, but it's not supported by the backend, initialization will fail. You can then fall back to shared mode if desired by changing this to ma_share_mode_shared and reinitializing. capture.pDeviceID A pointer to a `ma_device_id` structure containing the ID of the capture device to initialize. Setting this NULL (default) will use the system's default capture device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration. capture.format The sample format to use for capture. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after initialization from the device object directly with `device.capture.format`. capture.channels The number of channels to use for capture. When set to 0 the device's native channel count will be used. This can be retrieved after initialization from the device object directly with `device.capture.channels`. capture.pChannelMap The channel map to use for capture. When left empty, the device's native channel map will be used. This can be retrieved after initialization from the device object direct with `device.capture.pChannelMap`. When set, the buffer should contain `channels` items. capture.shareMode The preferred share mode to use for capture. Can be either `ma_share_mode_shared` (default) or `ma_share_mode_exclusive`. Note that if you specify exclusive mode, but it's not supported by the backend, initialization will fail. You can then fall back to shared mode if desired by changing this to ma_share_mode_shared and reinitializing. wasapi.noAutoConvertSRC WASAPI only. When set to true, disables WASAPI's automatic resampling and forces the use of miniaudio's resampler. Defaults to false. wasapi.noDefaultQualitySRC WASAPI only. Only used when `wasapi.noAutoConvertSRC` is set to false. When set to true, disables the use of `AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY`. You should usually leave this set to false, which is the default. wasapi.noAutoStreamRouting WASAPI only. When set to true, disables automatic stream routing on the WASAPI backend. Defaults to false. wasapi.noHardwareOffloading WASAPI only. When set to true, disables the use of WASAPI's hardware offloading feature. Defaults to false. alsa.noMMap ALSA only. When set to true, disables MMap mode. Defaults to false. alsa.noAutoFormat ALSA only. When set to true, disables ALSA's automatic format conversion by including the SND_PCM_NO_AUTO_FORMAT flag. Defaults to false. alsa.noAutoChannels ALSA only. When set to true, disables ALSA's automatic channel conversion by including the SND_PCM_NO_AUTO_CHANNELS flag. Defaults to false. alsa.noAutoResample ALSA only. When set to true, disables ALSA's automatic resampling by including the SND_PCM_NO_AUTO_RESAMPLE flag. Defaults to false. pulse.pStreamNamePlayback PulseAudio only. Sets the stream name for playback. pulse.pStreamNameCapture PulseAudio only. Sets the stream name for capture. pulse.channelMap PulseAudio only. Sets the channel map that is requested from PulseAudio. See MA_PA_CHANNEL_MAP_* constants. Defaults to MA_PA_CHANNEL_MAP_AIFF. coreaudio.allowNominalSampleRateChange Core Audio only. Desktop only. When enabled, allows the sample rate of the device to be changed at the operating system level. This is disabled by default in order to prevent intrusive changes to the user's system. This is useful if you want to use a sample rate that is known to be natively supported by the hardware thereby avoiding the cost of resampling. When set to true, miniaudio will find the closest match between the sample rate requested in the device config and the sample rates natively supported by the hardware. When set to false, the sample rate currently set by the operating system will always be used. opensl.streamType OpenSL only. Explicitly sets the stream type. If left unset (`ma_opensl_stream_type_default`), the stream type will be left unset. Think of this as the type of audio you're playing. opensl.recordingPreset OpenSL only. Explicitly sets the type of recording your program will be doing. When left unset, the recording preset will be left unchanged. aaudio.usage AAudio only. Explicitly sets the nature of the audio the program will be consuming. When left unset, the usage will be left unchanged. aaudio.contentType AAudio only. Sets the content type. When left unset, the content type will be left unchanged. aaudio.inputPreset AAudio only. Explicitly sets the type of recording your program will be doing. When left unset, the input preset will be left unchanged. aaudio.noAutoStartAfterReroute AAudio only. Controls whether or not the device should be automatically restarted after a stream reroute. When set to false (default) the device will be restarted automatically; otherwise the device will be stopped. Once initialized, the device's config is immutable. If you need to change the config you will need to initialize a new device. After initializing the device it will be in a stopped state. To start it, use `ma_device_start()`. If both `periodSizeInFrames` and `periodSizeInMilliseconds` are set to zero, it will default to `MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY` or `MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE`, depending on whether or not `performanceProfile` is set to `ma_performance_profile_low_latency` or `ma_performance_profile_conservative`. If you request exclusive mode and the backend does not support it an error will be returned. For robustness, you may want to first try initializing the device in exclusive mode, and then fall back to shared mode if required. Alternatively you can just request shared mode (the default if you leave it unset in the config) which is the most reliable option. Some backends do not have a practical way of choosing whether or not the device should be exclusive or not (ALSA, for example) in which case it just acts as a hint. Unless you have special requirements you should try avoiding exclusive mode as it's intrusive to the user. Starting with Windows 10, miniaudio will use low-latency shared mode where possible which may make exclusive mode unnecessary. When sending or receiving data to/from a device, miniaudio will internally perform a format conversion to convert between the format specified by the config and the format used internally by the backend. If you pass in 0 for the sample format, channel count, sample rate _and_ channel map, data transmission will run on an optimized pass-through fast path. You can retrieve the format, channel count and sample rate by inspecting the `playback/capture.format`, `playback/capture.channels` and `sampleRate` members of the device object. When compiling for UWP you must ensure you call this function on the main UI thread because the operating system may need to present the user with a message asking for permissions. Please refer to the official documentation for ActivateAudioInterfaceAsync() for more information. ALSA Specific: When initializing the default device, requesting shared mode will try using the "dmix" device for playback and the "dsnoop" device for capture. If these fail it will try falling back to the "hw" device. Example 1 - Simple Initialization --------------------------------- This example shows how to initialize a simple playback device using a standard configuration. If you are just needing to do simple playback from the default playback device this is usually all you need. ```c ma_device_config config = ma_device_config_init(ma_device_type_playback); config.playback.format = ma_format_f32; config.playback.channels = 2; config.sampleRate = 48000; config.dataCallback = ma_data_callback; config.pMyUserData = pMyUserData; ma_device device; ma_result result = ma_device_init(NULL, &config, &device); if (result != MA_SUCCESS) { // Error } ``` Example 2 - Advanced Initialization ----------------------------------- This example shows how you might do some more advanced initialization. In this hypothetical example we want to control the latency by setting the buffer size and period count. We also want to allow the user to be able to choose which device to output from which means we need a context so we can perform device enumeration. ```c ma_context context; ma_result result = ma_context_init(NULL, 0, NULL, &context); if (result != MA_SUCCESS) { // Error } ma_device_info* pPlaybackDeviceInfos; ma_uint32 playbackDeviceCount; result = ma_context_get_devices(&context, &pPlaybackDeviceInfos, &playbackDeviceCount, NULL, NULL); if (result != MA_SUCCESS) { // Error } // ... choose a device from pPlaybackDeviceInfos ... ma_device_config config = ma_device_config_init(ma_device_type_playback); config.playback.pDeviceID = pMyChosenDeviceID; // <-- Get this from the `id` member of one of the `ma_device_info` objects returned by ma_context_get_devices(). config.playback.format = ma_format_f32; config.playback.channels = 2; config.sampleRate = 48000; config.dataCallback = ma_data_callback; config.pUserData = pMyUserData; config.periodSizeInMilliseconds = 10; config.periods = 3; ma_device device; result = ma_device_init(&context, &config, &device); if (result != MA_SUCCESS) { // Error } ``` See Also -------- ma_device_config_init() ma_device_uninit() ma_device_start() ma_context_init() ma_context_get_devices() ma_context_enumerate_devices() */ MA_API ma_result ma_device_init(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice); /* Initializes a device without a context, with extra parameters for controlling the configuration of the internal self-managed context. This is the same as `ma_device_init()`, only instead of a context being passed in, the parameters from `ma_context_init()` are passed in instead. This function allows you to configure the internally created context. Parameters ---------- backends (in, optional) A list of backends to try initializing, in priority order. Can be NULL, in which case it uses default priority order. backendCount (in, optional) The number of items in `backend`. Ignored if `backend` is NULL. pContextConfig (in, optional) The context configuration. pConfig (in) A pointer to the device configuration. Cannot be null. See remarks for details. pDevice (out) A pointer to the device object being initialized. Return Value ------------ MA_SUCCESS if successful; any other error code otherwise. Thread Safety ------------- Unsafe. It is not safe to call this function simultaneously for different devices because some backends depend on and mutate global state. The same applies to calling this at the same time as `ma_device_uninit()`. Callback Safety --------------- Unsafe. It is not safe to call this inside any callback. Remarks ------- You only need to use this function if you want to configure the context differently to its defaults. You should never use this function if you want to manage your own context. See the documentation for `ma_context_init()` for information on the different context configuration options. See Also -------- ma_device_init() ma_device_uninit() ma_device_config_init() ma_context_init() */ MA_API ma_result ma_device_init_ex(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pContextConfig, const ma_device_config* pConfig, ma_device* pDevice); /* Uninitializes a device. This will explicitly stop the device. You do not need to call `ma_device_stop()` beforehand, but it's harmless if you do. Parameters ---------- pDevice (in) A pointer to the device to stop. Return Value ------------ Nothing Thread Safety ------------- Unsafe. As soon as this API is called the device should be considered undefined. Callback Safety --------------- Unsafe. It is not safe to call this inside any callback. Doing this will result in a deadlock. See Also -------- ma_device_init() ma_device_stop() */ MA_API void ma_device_uninit(ma_device* pDevice); /* Retrieves a pointer to the context that owns the given device. */ MA_API ma_context* ma_device_get_context(ma_device* pDevice); /* Helper function for retrieving the log object associated with the context that owns this device. */ MA_API ma_log* ma_device_get_log(ma_device* pDevice); /* Retrieves information about the device. Parameters ---------- pDevice (in) A pointer to the device whose information is being retrieved. type (in) The device type. This parameter is required for duplex devices. When retrieving device information, you are doing so for an individual playback or capture device. pDeviceInfo (out) A pointer to the `ma_device_info` that will receive the device information. Return Value ------------ MA_SUCCESS if successful; any other error code otherwise. Thread Safety ------------- Unsafe. This should be considered unsafe because it may be calling into the backend which may or may not be safe. Callback Safety --------------- Unsafe. You should avoid calling this in the data callback because it may call into the backend which may or may not be safe. */ MA_API ma_result ma_device_get_info(ma_device* pDevice, ma_device_type type, ma_device_info* pDeviceInfo); /* Retrieves the name of the device. Parameters ---------- pDevice (in) A pointer to the device whose information is being retrieved. type (in) The device type. This parameter is required for duplex devices. When retrieving device information, you are doing so for an individual playback or capture device. pName (out) A pointer to the buffer that will receive the name. nameCap (in) The capacity of the output buffer, including space for the null terminator. pLengthNotIncludingNullTerminator (out, optional) A pointer to the variable that will receive the length of the name, not including the null terminator. Return Value ------------ MA_SUCCESS if successful; any other error code otherwise. Thread Safety ------------- Unsafe. This should be considered unsafe because it may be calling into the backend which may or may not be safe. Callback Safety --------------- Unsafe. You should avoid calling this in the data callback because it may call into the backend which may or may not be safe. Remarks ------- If the name does not fully fit into the output buffer, it'll be truncated. You can pass in NULL to `pName` if you want to first get the length of the name for the purpose of memory allocation of the output buffer. Allocating a buffer of size `MA_MAX_DEVICE_NAME_LENGTH + 1` should be enough for most cases and will avoid the need for the inefficiency of calling this function twice. This is implemented in terms of `ma_device_get_info()`. */ MA_API ma_result ma_device_get_name(ma_device* pDevice, ma_device_type type, char* pName, size_t nameCap, size_t* pLengthNotIncludingNullTerminator); /* Starts the device. For playback devices this begins playback. For capture devices it begins recording. Use `ma_device_stop()` to stop the device. Parameters ---------- pDevice (in) A pointer to the device to start. Return Value ------------ MA_SUCCESS if successful; any other error code otherwise. Thread Safety ------------- Safe. It's safe to call this from any thread with the exception of the callback thread. Callback Safety --------------- Unsafe. It is not safe to call this inside any callback. Remarks ------- For a playback device, this will retrieve an initial chunk of audio data from the client before returning. The reason for this is to ensure there is valid audio data in the buffer, which needs to be done before the device begins playback. This API waits until the backend device has been started for real by the worker thread. It also waits on a mutex for thread-safety. Do not call this in any callback. See Also -------- ma_device_stop() */ MA_API ma_result ma_device_start(ma_device* pDevice); /* Stops the device. For playback devices this stops playback. For capture devices it stops recording. Use `ma_device_start()` to start the device again. Parameters ---------- pDevice (in) A pointer to the device to stop. Return Value ------------ MA_SUCCESS if successful; any other error code otherwise. Thread Safety ------------- Safe. It's safe to call this from any thread with the exception of the callback thread. Callback Safety --------------- Unsafe. It is not safe to call this inside any callback. Doing this will result in a deadlock. Remarks ------- This API needs to wait on the worker thread to stop the backend device properly before returning. It also waits on a mutex for thread-safety. In addition, some backends need to wait for the device to finish playback/recording of the current fragment which can take some time (usually proportionate to the buffer size that was specified at initialization time). Backends are required to either pause the stream in-place or drain the buffer if pausing is not possible. The reason for this is that stopping the device and the resuming it with ma_device_start() (which you might do when your program loses focus) may result in a situation where those samples are never output to the speakers or received from the microphone which can in turn result in de-syncs. Do not call this in any callback. See Also -------- ma_device_start() */ MA_API ma_result ma_device_stop(ma_device* pDevice); /* Determines whether or not the device is started. Parameters ---------- pDevice (in) A pointer to the device whose start state is being retrieved. Return Value ------------ True if the device is started, false otherwise. Thread Safety ------------- Safe. If another thread calls `ma_device_start()` or `ma_device_stop()` at this same time as this function is called, there's a very small chance the return value will be out of sync. Callback Safety --------------- Safe. This is implemented as a simple accessor. See Also -------- ma_device_start() ma_device_stop() */ MA_API ma_bool32 ma_device_is_started(const ma_device* pDevice); /* Retrieves the state of the device. Parameters ---------- pDevice (in) A pointer to the device whose state is being retrieved. Return Value ------------ The current state of the device. The return value will be one of the following: +-------------------------------+------------------------------------------------------------------------------+ | ma_device_state_uninitialized | Will only be returned if the device is in the middle of initialization. | +-------------------------------+------------------------------------------------------------------------------+ | ma_device_state_stopped | The device is stopped. The initial state of the device after initialization. | +-------------------------------+------------------------------------------------------------------------------+ | ma_device_state_started | The device started and requesting and/or delivering audio data. | +-------------------------------+------------------------------------------------------------------------------+ | ma_device_state_starting | The device is in the process of starting. | +-------------------------------+------------------------------------------------------------------------------+ | ma_device_state_stopping | The device is in the process of stopping. | +-------------------------------+------------------------------------------------------------------------------+ Thread Safety ------------- Safe. This is implemented as a simple accessor. Note that if the device is started or stopped at the same time as this function is called, there's a possibility the return value could be out of sync. See remarks. Callback Safety --------------- Safe. This is implemented as a simple accessor. Remarks ------- The general flow of a devices state goes like this: ``` ma_device_init() -> ma_device_state_uninitialized -> ma_device_state_stopped ma_device_start() -> ma_device_state_starting -> ma_device_state_started ma_device_stop() -> ma_device_state_stopping -> ma_device_state_stopped ``` When the state of the device is changed with `ma_device_start()` or `ma_device_stop()` at this same time as this function is called, the value returned by this function could potentially be out of sync. If this is significant to your program you need to implement your own synchronization. */ MA_API ma_device_state ma_device_get_state(const ma_device* pDevice); /* Performs post backend initialization routines for setting up internal data conversion. This should be called whenever the backend is initialized. The only time this should be called from outside of miniaudio is if you're implementing a custom backend, and you would only do it if you are reinitializing the backend due to rerouting or reinitializing for some reason. Parameters ---------- pDevice [in] A pointer to the device. deviceType [in] The type of the device that was just reinitialized. pPlaybackDescriptor [in] The descriptor of the playback device containing the internal data format and buffer sizes. pPlaybackDescriptor [in] The descriptor of the capture device containing the internal data format and buffer sizes. Return Value ------------ MA_SUCCESS if successful; any other error otherwise. Thread Safety ------------- Unsafe. This will be reinitializing internal data converters which may be in use by another thread. Callback Safety --------------- Unsafe. This will be reinitializing internal data converters which may be in use by the callback. Remarks ------- For a duplex device, you can call this for only one side of the system. This is why the deviceType is specified as a parameter rather than deriving it from the device. You do not need to call this manually unless you are doing a custom backend, in which case you need only do it if you're manually performing rerouting or reinitialization. */ MA_API ma_result ma_device_post_init(ma_device* pDevice, ma_device_type deviceType, const ma_device_descriptor* pPlaybackDescriptor, const ma_device_descriptor* pCaptureDescriptor); /* Sets the master volume factor for the device. The volume factor must be between 0 (silence) and 1 (full volume). Use `ma_device_set_master_volume_db()` to use decibel notation, where 0 is full volume and values less than 0 decreases the volume. Parameters ---------- pDevice (in) A pointer to the device whose volume is being set. volume (in) The new volume factor. Must be >= 0. Return Value ------------ MA_SUCCESS if the volume was set successfully. MA_INVALID_ARGS if pDevice is NULL. MA_INVALID_ARGS if volume is negative. Thread Safety ------------- Safe. This just sets a local member of the device object. Callback Safety --------------- Safe. If you set the volume in the data callback, that data written to the output buffer will have the new volume applied. Remarks ------- This applies the volume factor across all channels. This does not change the operating system's volume. It only affects the volume for the given `ma_device` object's audio stream. See Also -------- ma_device_get_master_volume() ma_device_set_master_volume_db() ma_device_get_master_volume_db() */ MA_API ma_result ma_device_set_master_volume(ma_device* pDevice, float volume); /* Retrieves the master volume factor for the device. Parameters ---------- pDevice (in) A pointer to the device whose volume factor is being retrieved. pVolume (in) A pointer to the variable that will receive the volume factor. The returned value will be in the range of [0, 1]. Return Value ------------ MA_SUCCESS if successful. MA_INVALID_ARGS if pDevice is NULL. MA_INVALID_ARGS if pVolume is NULL. Thread Safety ------------- Safe. This just a simple member retrieval. Callback Safety --------------- Safe. Remarks ------- If an error occurs, `*pVolume` will be set to 0. See Also -------- ma_device_set_master_volume() ma_device_set_master_volume_gain_db() ma_device_get_master_volume_gain_db() */ MA_API ma_result ma_device_get_master_volume(ma_device* pDevice, float* pVolume); /* Sets the master volume for the device as gain in decibels. A gain of 0 is full volume, whereas a gain of < 0 will decrease the volume. Parameters ---------- pDevice (in) A pointer to the device whose gain is being set. gainDB (in) The new volume as gain in decibels. Must be less than or equal to 0, where 0 is full volume and anything less than 0 decreases the volume. Return Value ------------ MA_SUCCESS if the volume was set successfully. MA_INVALID_ARGS if pDevice is NULL. MA_INVALID_ARGS if the gain is > 0. Thread Safety ------------- Safe. This just sets a local member of the device object. Callback Safety --------------- Safe. If you set the volume in the data callback, that data written to the output buffer will have the new volume applied. Remarks ------- This applies the gain across all channels. This does not change the operating system's volume. It only affects the volume for the given `ma_device` object's audio stream. See Also -------- ma_device_get_master_volume_gain_db() ma_device_set_master_volume() ma_device_get_master_volume() */ MA_API ma_result ma_device_set_master_volume_db(ma_device* pDevice, float gainDB); /* Retrieves the master gain in decibels. Parameters ---------- pDevice (in) A pointer to the device whose gain is being retrieved. pGainDB (in) A pointer to the variable that will receive the gain in decibels. The returned value will be <= 0. Return Value ------------ MA_SUCCESS if successful. MA_INVALID_ARGS if pDevice is NULL. MA_INVALID_ARGS if pGainDB is NULL. Thread Safety ------------- Safe. This just a simple member retrieval. Callback Safety --------------- Safe. Remarks ------- If an error occurs, `*pGainDB` will be set to 0. See Also -------- ma_device_set_master_volume_db() ma_device_set_master_volume() ma_device_get_master_volume() */ MA_API ma_result ma_device_get_master_volume_db(ma_device* pDevice, float* pGainDB); /* Called from the data callback of asynchronous backends to allow miniaudio to process the data and fire the miniaudio data callback. Parameters ---------- pDevice (in) A pointer to device whose processing the data callback. pOutput (out) A pointer to the buffer that will receive the output PCM frame data. On a playback device this must not be NULL. On a duplex device this can be NULL, in which case pInput must not be NULL. pInput (in) A pointer to the buffer containing input PCM frame data. On a capture device this must not be NULL. On a duplex device this can be NULL, in which case `pOutput` must not be NULL. frameCount (in) The number of frames being processed. Return Value ------------ MA_SUCCESS if successful; any other result code otherwise. Thread Safety ------------- This function should only ever be called from the internal data callback of the backend. It is safe to call this simultaneously between a playback and capture device in duplex setups. Callback Safety --------------- Do not call this from the miniaudio data callback. It should only ever be called from the internal data callback of the backend. Remarks ------- If both `pOutput` and `pInput` are NULL, and error will be returned. In duplex scenarios, both `pOutput` and `pInput` can be non-NULL, in which case `pInput` will be processed first, followed by `pOutput`. If you are implementing a custom backend, and that backend uses a callback for data delivery, you'll need to call this from inside that callback. */ MA_API ma_result ma_device_handle_backend_data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount); /* Calculates an appropriate buffer size from a descriptor, native sample rate and performance profile. This function is used by backends for helping determine an appropriately sized buffer to use with the device depending on the values of `periodSizeInFrames` and `periodSizeInMilliseconds` in the `pDescriptor` object. Since buffer size calculations based on time depends on the sample rate, a best guess at the device's native sample rate is also required which is where `nativeSampleRate` comes in. In addition, the performance profile is also needed for cases where both the period size in frames and milliseconds are both zero. Parameters ---------- pDescriptor (in) A pointer to device descriptor whose `periodSizeInFrames` and `periodSizeInMilliseconds` members will be used for the calculation of the buffer size. nativeSampleRate (in) The device's native sample rate. This is only ever used when the `periodSizeInFrames` member of `pDescriptor` is zero. In this case, `periodSizeInMilliseconds` will be used instead, in which case a sample rate is required to convert to a size in frames. performanceProfile (in) When both the `periodSizeInFrames` and `periodSizeInMilliseconds` members of `pDescriptor` are zero, miniaudio will fall back to a buffer size based on the performance profile. The profile to use for this calculation is determine by this parameter. Return Value ------------ The calculated buffer size in frames. Thread Safety ------------- This is safe so long as nothing modifies `pDescriptor` at the same time. However, this function should only ever be called from within the backend's device initialization routine and therefore shouldn't have any multithreading concerns. Callback Safety --------------- This is safe to call within the data callback, but there is no reason to ever do this. Remarks ------- If `nativeSampleRate` is zero, this function will fall back to `pDescriptor->sampleRate`. If that is also zero, `MA_DEFAULT_SAMPLE_RATE` will be used instead. */ MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_descriptor(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile); /* Retrieves a friendly name for a backend. */ MA_API const char* ma_get_backend_name(ma_backend backend); /* Retrieves the backend enum from the given name. */ MA_API ma_result ma_get_backend_from_name(const char* pBackendName, ma_backend* pBackend); /* Determines whether or not the given backend is available by the compilation environment. */ MA_API ma_bool32 ma_is_backend_enabled(ma_backend backend); /* Retrieves compile-time enabled backends. Parameters ---------- pBackends (out, optional) A pointer to the buffer that will receive the enabled backends. Set to NULL to retrieve the backend count. Setting the capacity of the buffer to `MA_BACKEND_COUNT` will guarantee it's large enough for all backends. backendCap (in) The capacity of the `pBackends` buffer. pBackendCount (out) A pointer to the variable that will receive the enabled backend count. Return Value ------------ MA_SUCCESS if successful. MA_INVALID_ARGS if `pBackendCount` is NULL. MA_NO_SPACE if the capacity of `pBackends` is not large enough. If `MA_NO_SPACE` is returned, the `pBackends` buffer will be filled with `*pBackendCount` values. Thread Safety ------------- Safe. Callback Safety --------------- Safe. Remarks ------- If you want to retrieve the number of backends so you can determine the capacity of `pBackends` buffer, you can call this function with `pBackends` set to NULL. This will also enumerate the null backend. If you don't want to include this you need to check for `ma_backend_null` when you enumerate over the returned backends and handle it appropriately. Alternatively, you can disable it at compile time with `MA_NO_NULL`. The returned backends are determined based on compile time settings, not the platform it's currently running on. For example, PulseAudio will be returned if it was enabled at compile time, even when the user doesn't actually have PulseAudio installed. Example 1 --------- The example below retrieves the enabled backend count using a fixed sized buffer allocated on the stack. The buffer is given a capacity of `MA_BACKEND_COUNT` which will guarantee it'll be large enough to store all available backends. Since `MA_BACKEND_COUNT` is always a relatively small value, this should be suitable for most scenarios. ``` ma_backend enabledBackends[MA_BACKEND_COUNT]; size_t enabledBackendCount; result = ma_get_enabled_backends(enabledBackends, MA_BACKEND_COUNT, &enabledBackendCount); if (result != MA_SUCCESS) { // Failed to retrieve enabled backends. Should never happen in this example since all inputs are valid. } ``` See Also -------- ma_is_backend_enabled() */ MA_API ma_result ma_get_enabled_backends(ma_backend* pBackends, size_t backendCap, size_t* pBackendCount); /* Determines whether or not loopback mode is support by a backend. */ MA_API ma_bool32 ma_is_loopback_supported(ma_backend backend); #endif /* MA_NO_DEVICE_IO */ /************************************************************************************************************************************************************ Utilities ************************************************************************************************************************************************************/ /* Calculates a buffer size in milliseconds (rounded up) from the specified number of frames and sample rate. */ MA_API ma_uint32 ma_calculate_buffer_size_in_milliseconds_from_frames(ma_uint32 bufferSizeInFrames, ma_uint32 sampleRate); /* Calculates a buffer size in frames from the specified number of milliseconds and sample rate. */ MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_milliseconds(ma_uint32 bufferSizeInMilliseconds, ma_uint32 sampleRate); /* Copies PCM frames from one buffer to another. */ MA_API void ma_copy_pcm_frames(void* dst, const void* src, ma_uint64 frameCount, ma_format format, ma_uint32 channels); /* Copies silent frames into the given buffer. Remarks ------- For all formats except `ma_format_u8`, the output buffer will be filled with 0. For `ma_format_u8` it will be filled with 128. The reason for this is that it makes more sense for the purpose of mixing to initialize it to the center point. */ MA_API void ma_silence_pcm_frames(void* p, ma_uint64 frameCount, ma_format format, ma_uint32 channels); /* Offsets a pointer by the specified number of PCM frames. */ MA_API void* ma_offset_pcm_frames_ptr(void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels); MA_API const void* ma_offset_pcm_frames_const_ptr(const void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels); static MA_INLINE float* ma_offset_pcm_frames_ptr_f32(float* p, ma_uint64 offsetInFrames, ma_uint32 channels) { return (float*)ma_offset_pcm_frames_ptr((void*)p, offsetInFrames, ma_format_f32, channels); } static MA_INLINE const float* ma_offset_pcm_frames_const_ptr_f32(const float* p, ma_uint64 offsetInFrames, ma_uint32 channels) { return (const float*)ma_offset_pcm_frames_const_ptr((const void*)p, offsetInFrames, ma_format_f32, channels); } /* Clips samples. */ MA_API void ma_clip_samples_u8(ma_uint8* pDst, const ma_int16* pSrc, ma_uint64 count); MA_API void ma_clip_samples_s16(ma_int16* pDst, const ma_int32* pSrc, ma_uint64 count); MA_API void ma_clip_samples_s24(ma_uint8* pDst, const ma_int64* pSrc, ma_uint64 count); MA_API void ma_clip_samples_s32(ma_int32* pDst, const ma_int64* pSrc, ma_uint64 count); MA_API void ma_clip_samples_f32(float* pDst, const float* pSrc, ma_uint64 count); MA_API void ma_clip_pcm_frames(void* pDst, const void* pSrc, ma_uint64 frameCount, ma_format format, ma_uint32 channels); /* Helper for applying a volume factor to samples. Note that the source and destination buffers can be the same, in which case it'll perform the operation in-place. */ MA_API void ma_copy_and_apply_volume_factor_u8(ma_uint8* pSamplesOut, const ma_uint8* pSamplesIn, ma_uint64 sampleCount, float factor); MA_API void ma_copy_and_apply_volume_factor_s16(ma_int16* pSamplesOut, const ma_int16* pSamplesIn, ma_uint64 sampleCount, float factor); MA_API void ma_copy_and_apply_volume_factor_s24(void* pSamplesOut, const void* pSamplesIn, ma_uint64 sampleCount, float factor); MA_API void ma_copy_and_apply_volume_factor_s32(ma_int32* pSamplesOut, const ma_int32* pSamplesIn, ma_uint64 sampleCount, float factor); MA_API void ma_copy_and_apply_volume_factor_f32(float* pSamplesOut, const float* pSamplesIn, ma_uint64 sampleCount, float factor); MA_API void ma_apply_volume_factor_u8(ma_uint8* pSamples, ma_uint64 sampleCount, float factor); MA_API void ma_apply_volume_factor_s16(ma_int16* pSamples, ma_uint64 sampleCount, float factor); MA_API void ma_apply_volume_factor_s24(void* pSamples, ma_uint64 sampleCount, float factor); MA_API void ma_apply_volume_factor_s32(ma_int32* pSamples, ma_uint64 sampleCount, float factor); MA_API void ma_apply_volume_factor_f32(float* pSamples, ma_uint64 sampleCount, float factor); MA_API void ma_copy_and_apply_volume_factor_pcm_frames_u8(ma_uint8* pFramesOut, const ma_uint8* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor); MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s16(ma_int16* pFramesOut, const ma_int16* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor); MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s24(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor); MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s32(ma_int32* pFramesOut, const ma_int32* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor); MA_API void ma_copy_and_apply_volume_factor_pcm_frames_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor); MA_API void ma_copy_and_apply_volume_factor_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor); MA_API void ma_apply_volume_factor_pcm_frames_u8(ma_uint8* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor); MA_API void ma_apply_volume_factor_pcm_frames_s16(ma_int16* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor); MA_API void ma_apply_volume_factor_pcm_frames_s24(void* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor); MA_API void ma_apply_volume_factor_pcm_frames_s32(ma_int32* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor); MA_API void ma_apply_volume_factor_pcm_frames_f32(float* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor); MA_API void ma_apply_volume_factor_pcm_frames(void* pFrames, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor); MA_API void ma_copy_and_apply_volume_factor_per_channel_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float* pChannelGains); MA_API void ma_copy_and_apply_volume_and_clip_samples_u8(ma_uint8* pDst, const ma_int16* pSrc, ma_uint64 count, float volume); MA_API void ma_copy_and_apply_volume_and_clip_samples_s16(ma_int16* pDst, const ma_int32* pSrc, ma_uint64 count, float volume); MA_API void ma_copy_and_apply_volume_and_clip_samples_s24(ma_uint8* pDst, const ma_int64* pSrc, ma_uint64 count, float volume); MA_API void ma_copy_and_apply_volume_and_clip_samples_s32(ma_int32* pDst, const ma_int64* pSrc, ma_uint64 count, float volume); MA_API void ma_copy_and_apply_volume_and_clip_samples_f32(float* pDst, const float* pSrc, ma_uint64 count, float volume); MA_API void ma_copy_and_apply_volume_and_clip_pcm_frames(void* pDst, const void* pSrc, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float volume); /* Helper for converting a linear factor to gain in decibels. */ MA_API float ma_volume_linear_to_db(float factor); /* Helper for converting gain in decibels to a linear factor. */ MA_API float ma_volume_db_to_linear(float gain); /* Mixes the specified number of frames in floating point format with a volume factor. This will run on an optimized path when the volume is equal to 1. */ MA_API ma_result ma_mix_pcm_frames_f32(float* pDst, const float* pSrc, ma_uint64 frameCount, ma_uint32 channels, float volume); /************************************************************************************************************************************************************ VFS === The VFS object (virtual file system) is what's used to customize file access. This is useful in cases where stdio FILE* based APIs may not be entirely appropriate for a given situation. ************************************************************************************************************************************************************/ typedef void ma_vfs; typedef ma_handle ma_vfs_file; typedef enum { MA_OPEN_MODE_READ = 0x00000001, MA_OPEN_MODE_WRITE = 0x00000002 } ma_open_mode_flags; typedef enum { ma_seek_origin_start, ma_seek_origin_current, ma_seek_origin_end /* Not used by decoders. */ } ma_seek_origin; typedef struct { ma_uint64 sizeInBytes; } ma_file_info; typedef struct { ma_result (* onOpen) (ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile); ma_result (* onOpenW)(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile); ma_result (* onClose)(ma_vfs* pVFS, ma_vfs_file file); ma_result (* onRead) (ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead); ma_result (* onWrite)(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten); ma_result (* onSeek) (ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin); ma_result (* onTell) (ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor); ma_result (* onInfo) (ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo); } ma_vfs_callbacks; MA_API ma_result ma_vfs_open(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile); MA_API ma_result ma_vfs_open_w(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile); MA_API ma_result ma_vfs_close(ma_vfs* pVFS, ma_vfs_file file); MA_API ma_result ma_vfs_read(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead); MA_API ma_result ma_vfs_write(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten); MA_API ma_result ma_vfs_seek(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin); MA_API ma_result ma_vfs_tell(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor); MA_API ma_result ma_vfs_info(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo); MA_API ma_result ma_vfs_open_and_read_file(ma_vfs* pVFS, const char* pFilePath, void** ppData, size_t* pSize, const ma_allocation_callbacks* pAllocationCallbacks); typedef struct { ma_vfs_callbacks cb; ma_allocation_callbacks allocationCallbacks; /* Only used for the wchar_t version of open() on non-Windows platforms. */ } ma_default_vfs; MA_API ma_result ma_default_vfs_init(ma_default_vfs* pVFS, const ma_allocation_callbacks* pAllocationCallbacks); typedef ma_result (* ma_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead); typedef ma_result (* ma_seek_proc)(void* pUserData, ma_int64 offset, ma_seek_origin origin); typedef ma_result (* ma_tell_proc)(void* pUserData, ma_int64* pCursor); #if !defined(MA_NO_DECODING) || !defined(MA_NO_ENCODING) typedef enum { ma_encoding_format_unknown = 0, ma_encoding_format_wav, ma_encoding_format_flac, ma_encoding_format_mp3, ma_encoding_format_vorbis } ma_encoding_format; #endif /************************************************************************************************************************************************************ Decoding ======== Decoders are independent of the main device API. Decoding APIs can be called freely inside the device's data callback, but they are not thread safe unless you do your own synchronization. ************************************************************************************************************************************************************/ #ifndef MA_NO_DECODING typedef struct ma_decoder ma_decoder; typedef struct { ma_format preferredFormat; ma_uint32 seekPointCount; /* Set to > 0 to generate a seektable if the decoding backend supports it. */ } ma_decoding_backend_config; MA_API ma_decoding_backend_config ma_decoding_backend_config_init(ma_format preferredFormat, ma_uint32 seekPointCount); typedef struct { ma_result (* onInit )(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); ma_result (* onInitFile )(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */ ma_result (* onInitFileW )(void* pUserData, const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */ ma_result (* onInitMemory)(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */ void (* onUninit )(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks); } ma_decoding_backend_vtable; typedef ma_result (* ma_decoder_read_proc)(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead); /* Returns the number of bytes read. */ typedef ma_result (* ma_decoder_seek_proc)(ma_decoder* pDecoder, ma_int64 byteOffset, ma_seek_origin origin); typedef ma_result (* ma_decoder_tell_proc)(ma_decoder* pDecoder, ma_int64* pCursor); typedef struct { ma_format format; /* Set to 0 or ma_format_unknown to use the stream's internal format. */ ma_uint32 channels; /* Set to 0 to use the stream's internal channels. */ ma_uint32 sampleRate; /* Set to 0 to use the stream's internal sample rate. */ ma_channel* pChannelMap; ma_channel_mix_mode channelMixMode; ma_dither_mode ditherMode; ma_resampler_config resampling; ma_allocation_callbacks allocationCallbacks; ma_encoding_format encodingFormat; ma_uint32 seekPointCount; /* When set to > 0, specifies the number of seek points to use for the generation of a seek table. Not all decoding backends support this. */ ma_decoding_backend_vtable** ppCustomBackendVTables; ma_uint32 customBackendCount; void* pCustomBackendUserData; } ma_decoder_config; struct ma_decoder { ma_data_source_base ds; ma_data_source* pBackend; /* The decoding backend we'll be pulling data from. */ const ma_decoding_backend_vtable* pBackendVTable; /* The vtable for the decoding backend. This needs to be stored so we can access the onUninit() callback. */ void* pBackendUserData; ma_decoder_read_proc onRead; ma_decoder_seek_proc onSeek; ma_decoder_tell_proc onTell; void* pUserData; ma_uint64 readPointerInPCMFrames; /* In output sample rate. Used for keeping track of how many frames are available for decoding. */ ma_format outputFormat; ma_uint32 outputChannels; ma_uint32 outputSampleRate; ma_data_converter converter; /* Data conversion is achieved by running frames through this. */ void* pInputCache; /* In input format. Can be null if it's not needed. */ ma_uint64 inputCacheCap; /* The capacity of the input cache. */ ma_uint64 inputCacheConsumed; /* The number of frames that have been consumed in the cache. Used for determining the next valid frame. */ ma_uint64 inputCacheRemaining; /* The number of valid frames remaining in the cache. */ ma_allocation_callbacks allocationCallbacks; union { struct { ma_vfs* pVFS; ma_vfs_file file; } vfs; struct { const ma_uint8* pData; size_t dataSize; size_t currentReadPos; } memory; /* Only used for decoders that were opened against a block of memory. */ } data; }; MA_API ma_decoder_config ma_decoder_config_init(ma_format outputFormat, ma_uint32 outputChannels, ma_uint32 outputSampleRate); MA_API ma_decoder_config ma_decoder_config_init_default(void); MA_API ma_result ma_decoder_init(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder); MA_API ma_result ma_decoder_init_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder); MA_API ma_result ma_decoder_init_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); MA_API ma_result ma_decoder_init_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); MA_API ma_result ma_decoder_init_file(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); MA_API ma_result ma_decoder_init_file_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); /* Uninitializes a decoder. */ MA_API ma_result ma_decoder_uninit(ma_decoder* pDecoder); /* Reads PCM frames from the given decoder. This is not thread safe without your own synchronization. */ MA_API ma_result ma_decoder_read_pcm_frames(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); /* Seeks to a PCM frame based on its absolute index. This is not thread safe without your own synchronization. */ MA_API ma_result ma_decoder_seek_to_pcm_frame(ma_decoder* pDecoder, ma_uint64 frameIndex); /* Retrieves the decoder's output data format. */ MA_API ma_result ma_decoder_get_data_format(ma_decoder* pDecoder, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap); /* Retrieves the current position of the read cursor in PCM frames. */ MA_API ma_result ma_decoder_get_cursor_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pCursor); /* Retrieves the length of the decoder in PCM frames. Do not call this on streams of an undefined length, such as internet radio. If the length is unknown or an error occurs, 0 will be returned. This will always return 0 for Vorbis decoders. This is due to a limitation with stb_vorbis in push mode which is what miniaudio uses internally. For MP3's, this will decode the entire file. Do not call this in time critical scenarios. This function is not thread safe without your own synchronization. */ MA_API ma_result ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pLength); /* Retrieves the number of frames that can be read before reaching the end. This calls `ma_decoder_get_length_in_pcm_frames()` so you need to be aware of the rules for that function, in particular ensuring you do not call it on streams of an undefined length, such as internet radio. If the total length of the decoder cannot be retrieved, such as with Vorbis decoders, `MA_NOT_IMPLEMENTED` will be returned. */ MA_API ma_result ma_decoder_get_available_frames(ma_decoder* pDecoder, ma_uint64* pAvailableFrames); /* Helper for opening and decoding a file into a heap allocated block of memory. Free the returned pointer with ma_free(). On input, pConfig should be set to what you want. On output it will be set to what you got. */ MA_API ma_result ma_decode_from_vfs(ma_vfs* pVFS, const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut); MA_API ma_result ma_decode_file(const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut); MA_API ma_result ma_decode_memory(const void* pData, size_t dataSize, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut); #endif /* MA_NO_DECODING */ /************************************************************************************************************************************************************ Encoding ======== Encoders do not perform any format conversion for you. If your target format does not support the format, and error will be returned. ************************************************************************************************************************************************************/ #ifndef MA_NO_ENCODING typedef struct ma_encoder ma_encoder; typedef ma_result (* ma_encoder_write_proc) (ma_encoder* pEncoder, const void* pBufferIn, size_t bytesToWrite, size_t* pBytesWritten); typedef ma_result (* ma_encoder_seek_proc) (ma_encoder* pEncoder, ma_int64 offset, ma_seek_origin origin); typedef ma_result (* ma_encoder_init_proc) (ma_encoder* pEncoder); typedef void (* ma_encoder_uninit_proc) (ma_encoder* pEncoder); typedef ma_result (* ma_encoder_write_pcm_frames_proc)(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount, ma_uint64* pFramesWritten); typedef struct { ma_encoding_format encodingFormat; ma_format format; ma_uint32 channels; ma_uint32 sampleRate; ma_allocation_callbacks allocationCallbacks; } ma_encoder_config; MA_API ma_encoder_config ma_encoder_config_init(ma_encoding_format encodingFormat, ma_format format, ma_uint32 channels, ma_uint32 sampleRate); struct ma_encoder { ma_encoder_config config; ma_encoder_write_proc onWrite; ma_encoder_seek_proc onSeek; ma_encoder_init_proc onInit; ma_encoder_uninit_proc onUninit; ma_encoder_write_pcm_frames_proc onWritePCMFrames; void* pUserData; void* pInternalEncoder; union { struct { ma_vfs* pVFS; ma_vfs_file file; } vfs; } data; }; MA_API ma_result ma_encoder_init(ma_encoder_write_proc onWrite, ma_encoder_seek_proc onSeek, void* pUserData, const ma_encoder_config* pConfig, ma_encoder* pEncoder); MA_API ma_result ma_encoder_init_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder); MA_API ma_result ma_encoder_init_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder); MA_API ma_result ma_encoder_init_file(const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder); MA_API ma_result ma_encoder_init_file_w(const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder); MA_API void ma_encoder_uninit(ma_encoder* pEncoder); MA_API ma_result ma_encoder_write_pcm_frames(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount, ma_uint64* pFramesWritten); #endif /* MA_NO_ENCODING */ /************************************************************************************************************************************************************ Generation ************************************************************************************************************************************************************/ #ifndef MA_NO_GENERATION typedef enum { ma_waveform_type_sine, ma_waveform_type_square, ma_waveform_type_triangle, ma_waveform_type_sawtooth } ma_waveform_type; typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; ma_waveform_type type; double amplitude; double frequency; } ma_waveform_config; MA_API ma_waveform_config ma_waveform_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_waveform_type type, double amplitude, double frequency); typedef struct { ma_data_source_base ds; ma_waveform_config config; double advance; double time; } ma_waveform; MA_API ma_result ma_waveform_init(const ma_waveform_config* pConfig, ma_waveform* pWaveform); MA_API void ma_waveform_uninit(ma_waveform* pWaveform); MA_API ma_result ma_waveform_read_pcm_frames(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); MA_API ma_result ma_waveform_seek_to_pcm_frame(ma_waveform* pWaveform, ma_uint64 frameIndex); MA_API ma_result ma_waveform_set_amplitude(ma_waveform* pWaveform, double amplitude); MA_API ma_result ma_waveform_set_frequency(ma_waveform* pWaveform, double frequency); MA_API ma_result ma_waveform_set_type(ma_waveform* pWaveform, ma_waveform_type type); MA_API ma_result ma_waveform_set_sample_rate(ma_waveform* pWaveform, ma_uint32 sampleRate); typedef struct { ma_format format; ma_uint32 channels; ma_uint32 sampleRate; double dutyCycle; double amplitude; double frequency; } ma_pulsewave_config; MA_API ma_pulsewave_config ma_pulsewave_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double dutyCycle, double amplitude, double frequency); typedef struct { ma_waveform waveform; ma_pulsewave_config config; } ma_pulsewave; MA_API ma_result ma_pulsewave_init(const ma_pulsewave_config* pConfig, ma_pulsewave* pWaveform); MA_API void ma_pulsewave_uninit(ma_pulsewave* pWaveform); MA_API ma_result ma_pulsewave_read_pcm_frames(ma_pulsewave* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); MA_API ma_result ma_pulsewave_seek_to_pcm_frame(ma_pulsewave* pWaveform, ma_uint64 frameIndex); MA_API ma_result ma_pulsewave_set_amplitude(ma_pulsewave* pWaveform, double amplitude); MA_API ma_result ma_pulsewave_set_frequency(ma_pulsewave* pWaveform, double frequency); MA_API ma_result ma_pulsewave_set_sample_rate(ma_pulsewave* pWaveform, ma_uint32 sampleRate); MA_API ma_result ma_pulsewave_set_duty_cycle(ma_pulsewave* pWaveform, double dutyCycle); typedef enum { ma_noise_type_white, ma_noise_type_pink, ma_noise_type_brownian } ma_noise_type; typedef struct { ma_format format; ma_uint32 channels; ma_noise_type type; ma_int32 seed; double amplitude; ma_bool32 duplicateChannels; } ma_noise_config; MA_API ma_noise_config ma_noise_config_init(ma_format format, ma_uint32 channels, ma_noise_type type, ma_int32 seed, double amplitude); typedef struct { ma_data_source_base ds; ma_noise_config config; ma_lcg lcg; union { struct { double** bin; double* accumulation; ma_uint32* counter; } pink; struct { double* accumulation; } brownian; } state; /* Memory management. */ void* _pHeap; ma_bool32 _ownsHeap; } ma_noise; MA_API ma_result ma_noise_get_heap_size(const ma_noise_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_noise_init_preallocated(const ma_noise_config* pConfig, void* pHeap, ma_noise* pNoise); MA_API ma_result ma_noise_init(const ma_noise_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_noise* pNoise); MA_API void ma_noise_uninit(ma_noise* pNoise, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_noise_read_pcm_frames(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); MA_API ma_result ma_noise_set_amplitude(ma_noise* pNoise, double amplitude); MA_API ma_result ma_noise_set_seed(ma_noise* pNoise, ma_int32 seed); MA_API ma_result ma_noise_set_type(ma_noise* pNoise, ma_noise_type type); #endif /* MA_NO_GENERATION */ /************************************************************************************************************************************************************ Resource Manager ************************************************************************************************************************************************************/ /* The resource manager cannot be enabled if there is no decoder. */ #if !defined(MA_NO_RESOURCE_MANAGER) && defined(MA_NO_DECODING) #define MA_NO_RESOURCE_MANAGER #endif #ifndef MA_NO_RESOURCE_MANAGER typedef struct ma_resource_manager ma_resource_manager; typedef struct ma_resource_manager_data_buffer_node ma_resource_manager_data_buffer_node; typedef struct ma_resource_manager_data_buffer ma_resource_manager_data_buffer; typedef struct ma_resource_manager_data_stream ma_resource_manager_data_stream; typedef struct ma_resource_manager_data_source ma_resource_manager_data_source; typedef enum { MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM = 0x00000001, /* When set, does not load the entire data source in memory. Disk I/O will happen on job threads. */ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE = 0x00000002, /* Decode data before storing in memory. When set, decoding is done at the resource manager level rather than the mixing thread. Results in faster mixing, but higher memory usage. */ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC = 0x00000004, /* When set, the resource manager will load the data source asynchronously. */ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT = 0x00000008, /* When set, waits for initialization of the underlying data source before returning from ma_resource_manager_data_source_init(). */ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_UNKNOWN_LENGTH = 0x00000010, /* Gives the resource manager a hint that the length of the data source is unknown and calling `ma_data_source_get_length_in_pcm_frames()` should be avoided. */ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_LOOPING = 0x00000020 /* When set, configures the data source to loop by default. */ } ma_resource_manager_data_source_flags; /* Pipeline notifications used by the resource manager. Made up of both an async notification and a fence, both of which are optional. */ typedef struct { ma_async_notification* pNotification; ma_fence* pFence; } ma_resource_manager_pipeline_stage_notification; typedef struct { ma_resource_manager_pipeline_stage_notification init; /* Initialization of the decoder. */ ma_resource_manager_pipeline_stage_notification done; /* Decoding fully completed. */ } ma_resource_manager_pipeline_notifications; MA_API ma_resource_manager_pipeline_notifications ma_resource_manager_pipeline_notifications_init(void); /* BEGIN BACKWARDS COMPATIBILITY */ /* TODO: Remove this block in version 0.12. */ #if 1 #define ma_resource_manager_job ma_job #define ma_resource_manager_job_init ma_job_init #define MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_FLAG_NON_BLOCKING MA_JOB_QUEUE_FLAG_NON_BLOCKING #define ma_resource_manager_job_queue_config ma_job_queue_config #define ma_resource_manager_job_queue_config_init ma_job_queue_config_init #define ma_resource_manager_job_queue ma_job_queue #define ma_resource_manager_job_queue_get_heap_size ma_job_queue_get_heap_size #define ma_resource_manager_job_queue_init_preallocated ma_job_queue_init_preallocated #define ma_resource_manager_job_queue_init ma_job_queue_init #define ma_resource_manager_job_queue_uninit ma_job_queue_uninit #define ma_resource_manager_job_queue_post ma_job_queue_post #define ma_resource_manager_job_queue_next ma_job_queue_next #endif /* END BACKWARDS COMPATIBILITY */ /* Maximum job thread count will be restricted to this, but this may be removed later and replaced with a heap allocation thereby removing any limitation. */ #ifndef MA_RESOURCE_MANAGER_MAX_JOB_THREAD_COUNT #define MA_RESOURCE_MANAGER_MAX_JOB_THREAD_COUNT 64 #endif typedef enum { /* Indicates ma_resource_manager_next_job() should not block. Only valid when the job thread count is 0. */ MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING = 0x00000001, /* Disables any kind of multithreading. Implicitly enables MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING. */ MA_RESOURCE_MANAGER_FLAG_NO_THREADING = 0x00000002 } ma_resource_manager_flags; typedef struct { const char* pFilePath; const wchar_t* pFilePathW; const ma_resource_manager_pipeline_notifications* pNotifications; ma_uint64 initialSeekPointInPCMFrames; ma_uint64 rangeBegInPCMFrames; ma_uint64 rangeEndInPCMFrames; ma_uint64 loopPointBegInPCMFrames; ma_uint64 loopPointEndInPCMFrames; ma_uint32 flags; ma_bool32 isLooping; /* Deprecated. Use the MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_LOOPING flag in `flags` instead. */ } ma_resource_manager_data_source_config; MA_API ma_resource_manager_data_source_config ma_resource_manager_data_source_config_init(void); typedef enum { ma_resource_manager_data_supply_type_unknown = 0, /* Used for determining whether or the data supply has been initialized. */ ma_resource_manager_data_supply_type_encoded, /* Data supply is an encoded buffer. Connector is ma_decoder. */ ma_resource_manager_data_supply_type_decoded, /* Data supply is a decoded buffer. Connector is ma_audio_buffer. */ ma_resource_manager_data_supply_type_decoded_paged /* Data supply is a linked list of decoded buffers. Connector is ma_paged_audio_buffer. */ } ma_resource_manager_data_supply_type; typedef struct { MA_ATOMIC(4, ma_resource_manager_data_supply_type) type; /* Read and written from different threads so needs to be accessed atomically. */ union { struct { const void* pData; size_t sizeInBytes; } encoded; struct { const void* pData; ma_uint64 totalFrameCount; ma_uint64 decodedFrameCount; ma_format format; ma_uint32 channels; ma_uint32 sampleRate; } decoded; struct { ma_paged_audio_buffer_data data; ma_uint64 decodedFrameCount; ma_uint32 sampleRate; } decodedPaged; } backend; } ma_resource_manager_data_supply; struct ma_resource_manager_data_buffer_node { ma_uint32 hashedName32; /* The hashed name. This is the key. */ ma_uint32 refCount; MA_ATOMIC(4, ma_result) result; /* Result from asynchronous loading. When loading set to MA_BUSY. When fully loaded set to MA_SUCCESS. When deleting set to MA_UNAVAILABLE. */ MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */ MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */ ma_bool32 isDataOwnedByResourceManager; /* Set to true when the underlying data buffer was allocated the resource manager. Set to false if it is owned by the application (via ma_resource_manager_register_*()). */ ma_resource_manager_data_supply data; ma_resource_manager_data_buffer_node* pParent; ma_resource_manager_data_buffer_node* pChildLo; ma_resource_manager_data_buffer_node* pChildHi; }; struct ma_resource_manager_data_buffer { ma_data_source_base ds; /* Base data source. A data buffer is a data source. */ ma_resource_manager* pResourceManager; /* A pointer to the resource manager that owns this buffer. */ ma_resource_manager_data_buffer_node* pNode; /* The data node. This is reference counted and is what supplies the data. */ ma_uint32 flags; /* The flags that were passed used to initialize the buffer. */ MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */ MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */ ma_uint64 seekTargetInPCMFrames; /* Only updated by the public API. Never written nor read from the job thread. */ ma_bool32 seekToCursorOnNextRead; /* On the next read we need to seek to the frame cursor. */ MA_ATOMIC(4, ma_result) result; /* Keeps track of a result of decoding. Set to MA_BUSY while the buffer is still loading. Set to MA_SUCCESS when loading is finished successfully. Otherwise set to some other code. */ MA_ATOMIC(4, ma_bool32) isLooping; /* Can be read and written by different threads at the same time. Must be used atomically. */ ma_atomic_bool32 isConnectorInitialized; /* Used for asynchronous loading to ensure we don't try to initialize the connector multiple times while waiting for the node to fully load. */ union { ma_decoder decoder; /* Supply type is ma_resource_manager_data_supply_type_encoded */ ma_audio_buffer buffer; /* Supply type is ma_resource_manager_data_supply_type_decoded */ ma_paged_audio_buffer pagedBuffer; /* Supply type is ma_resource_manager_data_supply_type_decoded_paged */ } connector; /* Connects this object to the node's data supply. */ }; struct ma_resource_manager_data_stream { ma_data_source_base ds; /* Base data source. A data stream is a data source. */ ma_resource_manager* pResourceManager; /* A pointer to the resource manager that owns this data stream. */ ma_uint32 flags; /* The flags that were passed used to initialize the stream. */ ma_decoder decoder; /* Used for filling pages with data. This is only ever accessed by the job thread. The public API should never touch this. */ ma_bool32 isDecoderInitialized; /* Required for determining whether or not the decoder should be uninitialized in MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_STREAM. */ ma_uint64 totalLengthInPCMFrames; /* This is calculated when first loaded by the MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_STREAM. */ ma_uint32 relativeCursor; /* The playback cursor, relative to the current page. Only ever accessed by the public API. Never accessed by the job thread. */ MA_ATOMIC(8, ma_uint64) absoluteCursor; /* The playback cursor, in absolute position starting from the start of the file. */ ma_uint32 currentPageIndex; /* Toggles between 0 and 1. Index 0 is the first half of pPageData. Index 1 is the second half. Only ever accessed by the public API. Never accessed by the job thread. */ MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */ MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */ /* Written by the public API, read by the job thread. */ MA_ATOMIC(4, ma_bool32) isLooping; /* Whether or not the stream is looping. It's important to set the looping flag at the data stream level for smooth loop transitions. */ /* Written by the job thread, read by the public API. */ void* pPageData; /* Buffer containing the decoded data of each page. Allocated once at initialization time. */ MA_ATOMIC(4, ma_uint32) pageFrameCount[2]; /* The number of valid PCM frames in each page. Used to determine the last valid frame. */ /* Written and read by both the public API and the job thread. These must be atomic. */ MA_ATOMIC(4, ma_result) result; /* Result from asynchronous loading. When loading set to MA_BUSY. When initialized set to MA_SUCCESS. When deleting set to MA_UNAVAILABLE. If an error occurs when loading, set to an error code. */ MA_ATOMIC(4, ma_bool32) isDecoderAtEnd; /* Whether or not the decoder has reached the end. */ MA_ATOMIC(4, ma_bool32) isPageValid[2]; /* Booleans to indicate whether or not a page is valid. Set to false by the public API, set to true by the job thread. Set to false as the pages are consumed, true when they are filled. */ MA_ATOMIC(4, ma_bool32) seekCounter; /* When 0, no seeking is being performed. When > 0, a seek is being performed and reading should be delayed with MA_BUSY. */ }; struct ma_resource_manager_data_source { union { ma_resource_manager_data_buffer buffer; ma_resource_manager_data_stream stream; } backend; /* Must be the first item because we need the first item to be the data source callbacks for the buffer or stream. */ ma_uint32 flags; /* The flags that were passed in to ma_resource_manager_data_source_init(). */ MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */ MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */ }; typedef struct { ma_allocation_callbacks allocationCallbacks; ma_log* pLog; ma_format decodedFormat; /* The decoded format to use. Set to ma_format_unknown (default) to use the file's native format. */ ma_uint32 decodedChannels; /* The decoded channel count to use. Set to 0 (default) to use the file's native channel count. */ ma_uint32 decodedSampleRate; /* the decoded sample rate to use. Set to 0 (default) to use the file's native sample rate. */ ma_uint32 jobThreadCount; /* Set to 0 if you want to self-manage your job threads. Defaults to 1. */ size_t jobThreadStackSize; ma_uint32 jobQueueCapacity; /* The maximum number of jobs that can fit in the queue at a time. Defaults to MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_CAPACITY. Cannot be zero. */ ma_uint32 flags; ma_vfs* pVFS; /* Can be NULL in which case defaults will be used. */ ma_decoding_backend_vtable** ppCustomDecodingBackendVTables; ma_uint32 customDecodingBackendCount; void* pCustomDecodingBackendUserData; ma_resampler_config resampling; } ma_resource_manager_config; MA_API ma_resource_manager_config ma_resource_manager_config_init(void); struct ma_resource_manager { ma_resource_manager_config config; ma_resource_manager_data_buffer_node* pRootDataBufferNode; /* The root buffer in the binary tree. */ #ifndef MA_NO_THREADING ma_mutex dataBufferBSTLock; /* For synchronizing access to the data buffer binary tree. */ ma_thread jobThreads[MA_RESOURCE_MANAGER_MAX_JOB_THREAD_COUNT]; /* The threads for executing jobs. */ #endif ma_job_queue jobQueue; /* Multi-consumer, multi-producer job queue for managing jobs for asynchronous decoding and streaming. */ ma_default_vfs defaultVFS; /* Only used if a custom VFS is not specified. */ ma_log log; /* Only used if no log was specified in the config. */ }; /* Init. */ MA_API ma_result ma_resource_manager_init(const ma_resource_manager_config* pConfig, ma_resource_manager* pResourceManager); MA_API void ma_resource_manager_uninit(ma_resource_manager* pResourceManager); MA_API ma_log* ma_resource_manager_get_log(ma_resource_manager* pResourceManager); /* Registration. */ MA_API ma_result ma_resource_manager_register_file(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags); MA_API ma_result ma_resource_manager_register_file_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags); MA_API ma_result ma_resource_manager_register_decoded_data(ma_resource_manager* pResourceManager, const char* pName, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate); /* Does not copy. Increments the reference count if already exists and returns MA_SUCCESS. */ MA_API ma_result ma_resource_manager_register_decoded_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate); MA_API ma_result ma_resource_manager_register_encoded_data(ma_resource_manager* pResourceManager, const char* pName, const void* pData, size_t sizeInBytes); /* Does not copy. Increments the reference count if already exists and returns MA_SUCCESS. */ MA_API ma_result ma_resource_manager_register_encoded_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName, const void* pData, size_t sizeInBytes); MA_API ma_result ma_resource_manager_unregister_file(ma_resource_manager* pResourceManager, const char* pFilePath); MA_API ma_result ma_resource_manager_unregister_file_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath); MA_API ma_result ma_resource_manager_unregister_data(ma_resource_manager* pResourceManager, const char* pName); MA_API ma_result ma_resource_manager_unregister_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName); /* Data Buffers. */ MA_API ma_result ma_resource_manager_data_buffer_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_buffer* pDataBuffer); MA_API ma_result ma_resource_manager_data_buffer_init(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_buffer* pDataBuffer); MA_API ma_result ma_resource_manager_data_buffer_init_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_buffer* pDataBuffer); MA_API ma_result ma_resource_manager_data_buffer_init_copy(ma_resource_manager* pResourceManager, const ma_resource_manager_data_buffer* pExistingDataBuffer, ma_resource_manager_data_buffer* pDataBuffer); MA_API ma_result ma_resource_manager_data_buffer_uninit(ma_resource_manager_data_buffer* pDataBuffer); MA_API ma_result ma_resource_manager_data_buffer_read_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); MA_API ma_result ma_resource_manager_data_buffer_seek_to_pcm_frame(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64 frameIndex); MA_API ma_result ma_resource_manager_data_buffer_get_data_format(ma_resource_manager_data_buffer* pDataBuffer, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap); MA_API ma_result ma_resource_manager_data_buffer_get_cursor_in_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pCursor); MA_API ma_result ma_resource_manager_data_buffer_get_length_in_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pLength); MA_API ma_result ma_resource_manager_data_buffer_result(const ma_resource_manager_data_buffer* pDataBuffer); MA_API ma_result ma_resource_manager_data_buffer_set_looping(ma_resource_manager_data_buffer* pDataBuffer, ma_bool32 isLooping); MA_API ma_bool32 ma_resource_manager_data_buffer_is_looping(const ma_resource_manager_data_buffer* pDataBuffer); MA_API ma_result ma_resource_manager_data_buffer_get_available_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pAvailableFrames); /* Data Streams. */ MA_API ma_result ma_resource_manager_data_stream_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_stream* pDataStream); MA_API ma_result ma_resource_manager_data_stream_init(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_stream* pDataStream); MA_API ma_result ma_resource_manager_data_stream_init_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_stream* pDataStream); MA_API ma_result ma_resource_manager_data_stream_uninit(ma_resource_manager_data_stream* pDataStream); MA_API ma_result ma_resource_manager_data_stream_read_pcm_frames(ma_resource_manager_data_stream* pDataStream, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); MA_API ma_result ma_resource_manager_data_stream_seek_to_pcm_frame(ma_resource_manager_data_stream* pDataStream, ma_uint64 frameIndex); MA_API ma_result ma_resource_manager_data_stream_get_data_format(ma_resource_manager_data_stream* pDataStream, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap); MA_API ma_result ma_resource_manager_data_stream_get_cursor_in_pcm_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pCursor); MA_API ma_result ma_resource_manager_data_stream_get_length_in_pcm_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pLength); MA_API ma_result ma_resource_manager_data_stream_result(const ma_resource_manager_data_stream* pDataStream); MA_API ma_result ma_resource_manager_data_stream_set_looping(ma_resource_manager_data_stream* pDataStream, ma_bool32 isLooping); MA_API ma_bool32 ma_resource_manager_data_stream_is_looping(const ma_resource_manager_data_stream* pDataStream); MA_API ma_result ma_resource_manager_data_stream_get_available_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pAvailableFrames); /* Data Sources. */ MA_API ma_result ma_resource_manager_data_source_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_source* pDataSource); MA_API ma_result ma_resource_manager_data_source_init(ma_resource_manager* pResourceManager, const char* pName, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_source* pDataSource); MA_API ma_result ma_resource_manager_data_source_init_w(ma_resource_manager* pResourceManager, const wchar_t* pName, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_source* pDataSource); MA_API ma_result ma_resource_manager_data_source_init_copy(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source* pExistingDataSource, ma_resource_manager_data_source* pDataSource); MA_API ma_result ma_resource_manager_data_source_uninit(ma_resource_manager_data_source* pDataSource); MA_API ma_result ma_resource_manager_data_source_read_pcm_frames(ma_resource_manager_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); MA_API ma_result ma_resource_manager_data_source_seek_to_pcm_frame(ma_resource_manager_data_source* pDataSource, ma_uint64 frameIndex); MA_API ma_result ma_resource_manager_data_source_get_data_format(ma_resource_manager_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap); MA_API ma_result ma_resource_manager_data_source_get_cursor_in_pcm_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pCursor); MA_API ma_result ma_resource_manager_data_source_get_length_in_pcm_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pLength); MA_API ma_result ma_resource_manager_data_source_result(const ma_resource_manager_data_source* pDataSource); MA_API ma_result ma_resource_manager_data_source_set_looping(ma_resource_manager_data_source* pDataSource, ma_bool32 isLooping); MA_API ma_bool32 ma_resource_manager_data_source_is_looping(const ma_resource_manager_data_source* pDataSource); MA_API ma_result ma_resource_manager_data_source_get_available_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pAvailableFrames); /* Job management. */ MA_API ma_result ma_resource_manager_post_job(ma_resource_manager* pResourceManager, const ma_job* pJob); MA_API ma_result ma_resource_manager_post_job_quit(ma_resource_manager* pResourceManager); /* Helper for posting a quit job. */ MA_API ma_result ma_resource_manager_next_job(ma_resource_manager* pResourceManager, ma_job* pJob); MA_API ma_result ma_resource_manager_process_job(ma_resource_manager* pResourceManager, ma_job* pJob); /* DEPRECATED. Use ma_job_process(). Will be removed in version 0.12. */ MA_API ma_result ma_resource_manager_process_next_job(ma_resource_manager* pResourceManager); /* Returns MA_CANCELLED if a MA_JOB_TYPE_QUIT job is found. In non-blocking mode, returns MA_NO_DATA_AVAILABLE if no jobs are available. */ #endif /* MA_NO_RESOURCE_MANAGER */ /************************************************************************************************************************************************************ Node Graph ************************************************************************************************************************************************************/ #ifndef MA_NO_NODE_GRAPH /* Must never exceed 254. */ #ifndef MA_MAX_NODE_BUS_COUNT #define MA_MAX_NODE_BUS_COUNT 254 #endif /* Used internally by miniaudio for memory management. Must never exceed MA_MAX_NODE_BUS_COUNT. */ #ifndef MA_MAX_NODE_LOCAL_BUS_COUNT #define MA_MAX_NODE_LOCAL_BUS_COUNT 2 #endif /* Use this when the bus count is determined by the node instance rather than the vtable. */ #define MA_NODE_BUS_COUNT_UNKNOWN 255 /* For some internal memory management of ma_node_graph. */ typedef struct { size_t offset; size_t sizeInBytes; unsigned char _data[1]; } ma_stack; typedef struct ma_node_graph ma_node_graph; typedef void ma_node; /* Node flags. */ typedef enum { MA_NODE_FLAG_PASSTHROUGH = 0x00000001, MA_NODE_FLAG_CONTINUOUS_PROCESSING = 0x00000002, MA_NODE_FLAG_ALLOW_NULL_INPUT = 0x00000004, MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES = 0x00000008, MA_NODE_FLAG_SILENT_OUTPUT = 0x00000010 } ma_node_flags; /* The playback state of a node. Either started or stopped. */ typedef enum { ma_node_state_started = 0, ma_node_state_stopped = 1 } ma_node_state; typedef struct { /* Extended processing callback. This callback is used for effects that process input and output at different rates (i.e. they perform resampling). This is similar to the simple version, only they take two separate frame counts: one for input, and one for output. On input, `pFrameCountOut` is equal to the capacity of the output buffer for each bus, whereas `pFrameCountIn` will be equal to the number of PCM frames in each of the buffers in `ppFramesIn`. On output, set `pFrameCountOut` to the number of PCM frames that were actually output and set `pFrameCountIn` to the number of input frames that were consumed. */ void (* onProcess)(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut); /* A callback for retrieving the number of input frames that are required to output the specified number of output frames. You would only want to implement this when the node performs resampling. This is optional, even for nodes that perform resampling, but it does offer a small reduction in latency as it allows miniaudio to calculate the exact number of input frames to read at a time instead of having to estimate. */ ma_result (* onGetRequiredInputFrameCount)(ma_node* pNode, ma_uint32 outputFrameCount, ma_uint32* pInputFrameCount); /* The number of input buses. This is how many sub-buffers will be contained in the `ppFramesIn` parameters of the callbacks above. */ ma_uint8 inputBusCount; /* The number of output buses. This is how many sub-buffers will be contained in the `ppFramesOut` parameters of the callbacks above. */ ma_uint8 outputBusCount; /* Flags describing characteristics of the node. This is currently just a placeholder for some ideas for later on. */ ma_uint32 flags; } ma_node_vtable; typedef struct { const ma_node_vtable* vtable; /* Should never be null. Initialization of the node will fail if so. */ ma_node_state initialState; /* Defaults to ma_node_state_started. */ ma_uint32 inputBusCount; /* Only used if the vtable specifies an input bus count of `MA_NODE_BUS_COUNT_UNKNOWN`, otherwise must be set to `MA_NODE_BUS_COUNT_UNKNOWN` (default). */ ma_uint32 outputBusCount; /* Only used if the vtable specifies an output bus count of `MA_NODE_BUS_COUNT_UNKNOWN`, otherwise be set to `MA_NODE_BUS_COUNT_UNKNOWN` (default). */ const ma_uint32* pInputChannels; /* The number of elements are determined by the input bus count as determined by the vtable, or `inputBusCount` if the vtable specifies `MA_NODE_BUS_COUNT_UNKNOWN`. */ const ma_uint32* pOutputChannels; /* The number of elements are determined by the output bus count as determined by the vtable, or `outputBusCount` if the vtable specifies `MA_NODE_BUS_COUNT_UNKNOWN`. */ } ma_node_config; MA_API ma_node_config ma_node_config_init(void); /* A node has multiple output buses. An output bus is attached to an input bus as an item in a linked list. Think of the input bus as a linked list, with the output bus being an item in that list. */ typedef struct ma_node_output_bus ma_node_output_bus; struct ma_node_output_bus { /* Immutable. */ ma_node* pNode; /* The node that owns this output bus. The input node. Will be null for dummy head and tail nodes. */ ma_uint8 outputBusIndex; /* The index of the output bus on pNode that this output bus represents. */ ma_uint8 channels; /* The number of channels in the audio stream for this bus. */ /* Mutable via multiple threads. Must be used atomically. The weird ordering here is for packing reasons. */ ma_uint8 inputNodeInputBusIndex; /* The index of the input bus on the input. Required for detaching. Will only be used within the spinlock so does not need to be atomic. */ MA_ATOMIC(4, ma_uint32) flags; /* Some state flags for tracking the read state of the output buffer. A combination of MA_NODE_OUTPUT_BUS_FLAG_*. */ MA_ATOMIC(4, ma_uint32) refCount; /* Reference count for some thread-safety when detaching. */ MA_ATOMIC(4, ma_bool32) isAttached; /* This is used to prevent iteration of nodes that are in the middle of being detached. Used for thread safety. */ MA_ATOMIC(4, ma_spinlock) lock; /* Unfortunate lock, but significantly simplifies the implementation. Required for thread-safe attaching and detaching. */ MA_ATOMIC(4, float) volume; /* Linear. */ MA_ATOMIC(MA_SIZEOF_PTR, ma_node_output_bus*) pNext; /* If null, it's the tail node or detached. */ MA_ATOMIC(MA_SIZEOF_PTR, ma_node_output_bus*) pPrev; /* If null, it's the head node or detached. */ MA_ATOMIC(MA_SIZEOF_PTR, ma_node*) pInputNode; /* The node that this output bus is attached to. Required for detaching. */ }; /* A node has multiple input buses. The output buses of a node are connecting to the input busses of another. An input bus is essentially just a linked list of output buses. */ typedef struct ma_node_input_bus ma_node_input_bus; struct ma_node_input_bus { /* Mutable via multiple threads. */ ma_node_output_bus head; /* Dummy head node for simplifying some lock-free thread-safety stuff. */ MA_ATOMIC(4, ma_uint32) nextCounter; /* This is used to determine whether or not the input bus is finding the next node in the list. Used for thread safety when detaching output buses. */ MA_ATOMIC(4, ma_spinlock) lock; /* Unfortunate lock, but significantly simplifies the implementation. Required for thread-safe attaching and detaching. */ /* Set once at startup. */ ma_uint8 channels; /* The number of channels in the audio stream for this bus. */ }; typedef struct ma_node_base ma_node_base; struct ma_node_base { /* These variables are set once at startup. */ ma_node_graph* pNodeGraph; /* The graph this node belongs to. */ const ma_node_vtable* vtable; ma_uint32 inputBusCount; ma_uint32 outputBusCount; ma_node_input_bus* pInputBuses; ma_node_output_bus* pOutputBuses; float* pCachedData; /* Allocated on the heap. Fixed size. Needs to be stored on the heap because reading from output buses is done in separate function calls. */ ma_uint16 cachedDataCapInFramesPerBus; /* The capacity of the input data cache in frames, per bus. */ /* These variables are read and written only from the audio thread. */ ma_uint16 cachedFrameCountOut; ma_uint16 cachedFrameCountIn; ma_uint16 consumedFrameCountIn; /* These variables are read and written between different threads. */ MA_ATOMIC(4, ma_node_state) state; /* When set to stopped, nothing will be read, regardless of the times in stateTimes. */ MA_ATOMIC(8, ma_uint64) stateTimes[2]; /* Indexed by ma_node_state. Specifies the time based on the global clock that a node should be considered to be in the relevant state. */ MA_ATOMIC(8, ma_uint64) localTime; /* The node's local clock. This is just a running sum of the number of output frames that have been processed. Can be modified by any thread with `ma_node_set_time()`. */ /* Memory management. */ ma_node_input_bus _inputBuses[MA_MAX_NODE_LOCAL_BUS_COUNT]; ma_node_output_bus _outputBuses[MA_MAX_NODE_LOCAL_BUS_COUNT]; void* _pHeap; /* A heap allocation for internal use only. pInputBuses and/or pOutputBuses will point to this if the bus count exceeds MA_MAX_NODE_LOCAL_BUS_COUNT. */ ma_bool32 _ownsHeap; /* If set to true, the node owns the heap allocation and _pHeap will be freed in ma_node_uninit(). */ }; MA_API ma_result ma_node_get_heap_size(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_node_init_preallocated(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, void* pHeap, ma_node* pNode); MA_API ma_result ma_node_init(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_node* pNode); MA_API void ma_node_uninit(ma_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_node_graph* ma_node_get_node_graph(const ma_node* pNode); MA_API ma_uint32 ma_node_get_input_bus_count(const ma_node* pNode); MA_API ma_uint32 ma_node_get_output_bus_count(const ma_node* pNode); MA_API ma_uint32 ma_node_get_input_channels(const ma_node* pNode, ma_uint32 inputBusIndex); MA_API ma_uint32 ma_node_get_output_channels(const ma_node* pNode, ma_uint32 outputBusIndex); MA_API ma_result ma_node_attach_output_bus(ma_node* pNode, ma_uint32 outputBusIndex, ma_node* pOtherNode, ma_uint32 otherNodeInputBusIndex); MA_API ma_result ma_node_detach_output_bus(ma_node* pNode, ma_uint32 outputBusIndex); MA_API ma_result ma_node_detach_all_output_buses(ma_node* pNode); MA_API ma_result ma_node_set_output_bus_volume(ma_node* pNode, ma_uint32 outputBusIndex, float volume); MA_API float ma_node_get_output_bus_volume(const ma_node* pNode, ma_uint32 outputBusIndex); MA_API ma_result ma_node_set_state(ma_node* pNode, ma_node_state state); MA_API ma_node_state ma_node_get_state(const ma_node* pNode); MA_API ma_result ma_node_set_state_time(ma_node* pNode, ma_node_state state, ma_uint64 globalTime); MA_API ma_uint64 ma_node_get_state_time(const ma_node* pNode, ma_node_state state); MA_API ma_node_state ma_node_get_state_by_time(const ma_node* pNode, ma_uint64 globalTime); MA_API ma_node_state ma_node_get_state_by_time_range(const ma_node* pNode, ma_uint64 globalTimeBeg, ma_uint64 globalTimeEnd); MA_API ma_uint64 ma_node_get_time(const ma_node* pNode); MA_API ma_result ma_node_set_time(ma_node* pNode, ma_uint64 localTime); typedef struct { ma_uint32 channels; ma_uint32 processingSizeInFrames; /* This is the preferred processing size for node processing callbacks unless overridden by a node itself. Can be 0 in which case it will be based on the frame count passed into ma_node_graph_read_pcm_frames(), but will not be well defined. */ size_t preMixStackSizeInBytes; /* Defaults to 512KB per channel. Reducing this will save memory, but the depth of your node graph will be more restricted. */ } ma_node_graph_config; MA_API ma_node_graph_config ma_node_graph_config_init(ma_uint32 channels); struct ma_node_graph { /* Immutable. */ ma_node_base base; /* The node graph itself is a node so it can be connected as an input to different node graph. This has zero inputs and calls ma_node_graph_read_pcm_frames() to generate it's output. */ ma_node_base endpoint; /* Special node that all nodes eventually connect to. Data is read from this node in ma_node_graph_read_pcm_frames(). */ float* pProcessingCache; /* This will be allocated when processingSizeInFrames is non-zero. This is needed because ma_node_graph_read_pcm_frames() can be called with a variable number of frames, and we may need to do some buffering in situations where the caller requests a frame count that's not a multiple of processingSizeInFrames. */ ma_uint32 processingCacheFramesRemaining; ma_uint32 processingSizeInFrames; /* Read and written by multiple threads. */ MA_ATOMIC(4, ma_bool32) isReading; /* Modified only by the audio thread. */ ma_stack* pPreMixStack; }; MA_API ma_result ma_node_graph_init(const ma_node_graph_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_node_graph* pNodeGraph); MA_API void ma_node_graph_uninit(ma_node_graph* pNodeGraph, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_node* ma_node_graph_get_endpoint(ma_node_graph* pNodeGraph); MA_API ma_result ma_node_graph_read_pcm_frames(ma_node_graph* pNodeGraph, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); MA_API ma_uint32 ma_node_graph_get_channels(const ma_node_graph* pNodeGraph); MA_API ma_uint64 ma_node_graph_get_time(const ma_node_graph* pNodeGraph); MA_API ma_result ma_node_graph_set_time(ma_node_graph* pNodeGraph, ma_uint64 globalTime); MA_API ma_uint32 ma_node_graph_get_processing_size_in_frames(const ma_node_graph* pNodeGraph); /* Data source node. 0 input buses, 1 output bus. Used for reading from a data source. */ typedef struct { ma_node_config nodeConfig; ma_data_source* pDataSource; } ma_data_source_node_config; MA_API ma_data_source_node_config ma_data_source_node_config_init(ma_data_source* pDataSource); typedef struct { ma_node_base base; ma_data_source* pDataSource; } ma_data_source_node; MA_API ma_result ma_data_source_node_init(ma_node_graph* pNodeGraph, const ma_data_source_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source_node* pDataSourceNode); MA_API void ma_data_source_node_uninit(ma_data_source_node* pDataSourceNode, const ma_allocation_callbacks* pAllocationCallbacks); MA_API ma_result ma_data_source_node_set_looping(ma_data_source_node* pDataSourceNode, ma_bool32 isLooping); MA_API ma_bool32 ma_data_source_node_is_looping(ma_data_source_node* pDataSourceNode); /* Splitter Node. 1 input, many outputs. Used for splitting/copying a stream so it can be as input into two separate output nodes. */ typedef struct { ma_node_config nodeConfig; ma_uint32 channels; ma_uint32 outputBusCount; } ma_splitter_node_config; MA_API ma_splitter_node_config ma_splitter_node_config_init(ma_uint32 channels); typedef struct { ma_node_base base; } ma_splitter_node; MA_API ma_result ma_splitter_node_init(ma_node_graph* pNodeGraph, const ma_splitter_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_splitter_node* pSplitterNode); MA_API void ma_splitter_node_uninit(ma_splitter_node* pSplitterNode, const ma_allocation_callbacks* pAllocationCallbacks); /* Biquad Node */ typedef struct { ma_node_config nodeConfig; ma_biquad_config biquad; } ma_biquad_node_config; MA_API ma_biquad_node_config ma_biquad_node_config_init(ma_uint32 channels, float b0, float b1, float b2, float a0, float a1, float a2); typedef struct { ma_node_base baseNode; ma_biquad biquad; } ma_biquad_node; MA_API ma_result ma_biquad_node_init(ma_node_graph* pNodeGraph, const ma_biquad_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_biquad_node* pNode); MA_API ma_result ma_biquad_node_reinit(const ma_biquad_config* pConfig, ma_biquad_node* pNode); MA_API void ma_biquad_node_uninit(ma_biquad_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks); /* Low Pass Filter Node */ typedef struct { ma_node_config nodeConfig; ma_lpf_config lpf; } ma_lpf_node_config; MA_API ma_lpf_node_config ma_lpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order); typedef struct { ma_node_base baseNode; ma_lpf lpf; } ma_lpf_node; MA_API ma_result ma_lpf_node_init(ma_node_graph* pNodeGraph, const ma_lpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf_node* pNode); MA_API ma_result ma_lpf_node_reinit(const ma_lpf_config* pConfig, ma_lpf_node* pNode); MA_API void ma_lpf_node_uninit(ma_lpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks); /* High Pass Filter Node */ typedef struct { ma_node_config nodeConfig; ma_hpf_config hpf; } ma_hpf_node_config; MA_API ma_hpf_node_config ma_hpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order); typedef struct { ma_node_base baseNode; ma_hpf hpf; } ma_hpf_node; MA_API ma_result ma_hpf_node_init(ma_node_graph* pNodeGraph, const ma_hpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf_node* pNode); MA_API ma_result ma_hpf_node_reinit(const ma_hpf_config* pConfig, ma_hpf_node* pNode); MA_API void ma_hpf_node_uninit(ma_hpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks); /* Band Pass Filter Node */ typedef struct { ma_node_config nodeConfig; ma_bpf_config bpf; } ma_bpf_node_config; MA_API ma_bpf_node_config ma_bpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order); typedef struct { ma_node_base baseNode; ma_bpf bpf; } ma_bpf_node; MA_API ma_result ma_bpf_node_init(ma_node_graph* pNodeGraph, const ma_bpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf_node* pNode); MA_API ma_result ma_bpf_node_reinit(const ma_bpf_config* pConfig, ma_bpf_node* pNode); MA_API void ma_bpf_node_uninit(ma_bpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks); /* Notching Filter Node */ typedef struct { ma_node_config nodeConfig; ma_notch_config notch; } ma_notch_node_config; MA_API ma_notch_node_config ma_notch_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double q, double frequency); typedef struct { ma_node_base baseNode; ma_notch2 notch; } ma_notch_node; MA_API ma_result ma_notch_node_init(ma_node_graph* pNodeGraph, const ma_notch_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_notch_node* pNode); MA_API ma_result ma_notch_node_reinit(const ma_notch_config* pConfig, ma_notch_node* pNode); MA_API void ma_notch_node_uninit(ma_notch_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks); /* Peaking Filter Node */ typedef struct { ma_node_config nodeConfig; ma_peak_config peak; } ma_peak_node_config; MA_API ma_peak_node_config ma_peak_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency); typedef struct { ma_node_base baseNode; ma_peak2 peak; } ma_peak_node; MA_API ma_result ma_peak_node_init(ma_node_graph* pNodeGraph, const ma_peak_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_peak_node* pNode); MA_API ma_result ma_peak_node_reinit(const ma_peak_config* pConfig, ma_peak_node* pNode); MA_API void ma_peak_node_uninit(ma_peak_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks); /* Low Shelf Filter Node */ typedef struct { ma_node_config nodeConfig; ma_loshelf_config loshelf; } ma_loshelf_node_config; MA_API ma_loshelf_node_config ma_loshelf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency); typedef struct { ma_node_base baseNode; ma_loshelf2 loshelf; } ma_loshelf_node; MA_API ma_result ma_loshelf_node_init(ma_node_graph* pNodeGraph, const ma_loshelf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_loshelf_node* pNode); MA_API ma_result ma_loshelf_node_reinit(const ma_loshelf_config* pConfig, ma_loshelf_node* pNode); MA_API void ma_loshelf_node_uninit(ma_loshelf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks); /* High Shelf Filter Node */ typedef struct { ma_node_config nodeConfig; ma_hishelf_config hishelf; } ma_hishelf_node_config; MA_API ma_hishelf_node_config ma_hishelf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency); typedef struct { ma_node_base baseNode; ma_hishelf2 hishelf; } ma_hishelf_node; MA_API ma_result ma_hishelf_node_init(ma_node_graph* pNodeGraph, const ma_hishelf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hishelf_node* pNode); MA_API ma_result ma_hishelf_node_reinit(const ma_hishelf_config* pConfig, ma_hishelf_node* pNode); MA_API void ma_hishelf_node_uninit(ma_hishelf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks); typedef struct { ma_node_config nodeConfig; ma_delay_config delay; } ma_delay_node_config; MA_API ma_delay_node_config ma_delay_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 delayInFrames, float decay); typedef struct { ma_node_base baseNode; ma_delay delay; } ma_delay_node; MA_API ma_result ma_delay_node_init(ma_node_graph* pNodeGraph, const ma_delay_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_delay_node* pDelayNode); MA_API void ma_delay_node_uninit(ma_delay_node* pDelayNode, const ma_allocation_callbacks* pAllocationCallbacks); MA_API void ma_delay_node_set_wet(ma_delay_node* pDelayNode, float value); MA_API float ma_delay_node_get_wet(const ma_delay_node* pDelayNode); MA_API void ma_delay_node_set_dry(ma_delay_node* pDelayNode, float value); MA_API float ma_delay_node_get_dry(const ma_delay_node* pDelayNode); MA_API void ma_delay_node_set_decay(ma_delay_node* pDelayNode, float value); MA_API float ma_delay_node_get_decay(const ma_delay_node* pDelayNode); #endif /* MA_NO_NODE_GRAPH */ /* SECTION: miniaudio_engine.h */ /************************************************************************************************************************************************************ Engine ************************************************************************************************************************************************************/ #if !defined(MA_NO_ENGINE) && !defined(MA_NO_NODE_GRAPH) typedef struct ma_engine ma_engine; typedef struct ma_sound ma_sound; /* Sound flags. */ typedef enum { /* Resource manager flags. */ MA_SOUND_FLAG_STREAM = 0x00000001, /* MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM */ MA_SOUND_FLAG_DECODE = 0x00000002, /* MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE */ MA_SOUND_FLAG_ASYNC = 0x00000004, /* MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC */ MA_SOUND_FLAG_WAIT_INIT = 0x00000008, /* MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT */ MA_SOUND_FLAG_UNKNOWN_LENGTH = 0x00000010, /* MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_UNKNOWN_LENGTH */ MA_SOUND_FLAG_LOOPING = 0x00000020, /* MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_LOOPING */ /* ma_sound specific flags. */ MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT = 0x00001000, /* Do not attach to the endpoint by default. Useful for when setting up nodes in a complex graph system. */ MA_SOUND_FLAG_NO_PITCH = 0x00002000, /* Disable pitch shifting with ma_sound_set_pitch() and ma_sound_group_set_pitch(). This is an optimization. */ MA_SOUND_FLAG_NO_SPATIALIZATION = 0x00004000 /* Disable spatialization. */ } ma_sound_flags; #ifndef MA_ENGINE_MAX_LISTENERS #define MA_ENGINE_MAX_LISTENERS 4 #endif #define MA_LISTENER_INDEX_CLOSEST ((ma_uint8)-1) typedef enum { ma_engine_node_type_sound, ma_engine_node_type_group } ma_engine_node_type; typedef struct { ma_engine* pEngine; ma_engine_node_type type; ma_uint32 channelsIn; ma_uint32 channelsOut; ma_uint32 sampleRate; /* Only used when the type is set to ma_engine_node_type_sound. */ ma_uint32 volumeSmoothTimeInPCMFrames; /* The number of frames to smooth over volume changes. Defaults to 0 in which case no smoothing is used. */ ma_mono_expansion_mode monoExpansionMode; ma_bool8 isPitchDisabled; /* Pitching can be explicitly disabled with MA_SOUND_FLAG_NO_PITCH to optimize processing. */ ma_bool8 isSpatializationDisabled; /* Spatialization can be explicitly disabled with MA_SOUND_FLAG_NO_SPATIALIZATION. */ ma_uint8 pinnedListenerIndex; /* The index of the listener this node should always use for spatialization. If set to MA_LISTENER_INDEX_CLOSEST the engine will use the closest listener. */ ma_resampler_config resampling; } ma_engine_node_config; MA_API ma_engine_node_config ma_engine_node_config_init(ma_engine* pEngine, ma_engine_node_type type, ma_uint32 flags); /* Base node object for both ma_sound and ma_sound_group. */ typedef struct { ma_node_base baseNode; /* Must be the first member for compatibility with the ma_node API. */ ma_engine* pEngine; /* A pointer to the engine. Set based on the value from the config. */ ma_uint32 sampleRate; /* The sample rate of the input data. For sounds backed by a data source, this will be the data source's sample rate. Otherwise it'll be the engine's sample rate. */ ma_uint32 volumeSmoothTimeInPCMFrames; ma_mono_expansion_mode monoExpansionMode; ma_fader fader; ma_resampler resampler; /* For pitch shift. */ ma_spatializer spatializer; ma_panner panner; ma_gainer volumeGainer; /* This will only be used if volumeSmoothTimeInPCMFrames is > 0. */ ma_atomic_float volume; /* Defaults to 1. */ MA_ATOMIC(4, float) pitch; float oldPitch; /* For determining whether or not the resampler needs to be updated to reflect the new pitch. The resampler will be updated on the mixing thread. */ float oldDopplerPitch; /* For determining whether or not the resampler needs to be updated to take a new doppler pitch into account. */ MA_ATOMIC(4, ma_bool32) isPitchDisabled; /* When set to true, pitching will be disabled which will allow the resampler to be bypassed to save some computation. */ MA_ATOMIC(4, ma_bool32) isSpatializationDisabled; /* Set to false by default. When set to false, will not have spatialisation applied. */ MA_ATOMIC(4, ma_uint32) pinnedListenerIndex; /* The index of the listener this node should always use for spatialization. If set to MA_LISTENER_INDEX_CLOSEST the engine will use the closest listener. */ /* When setting a fade, it's not done immediately in ma_sound_set_fade(). It's deferred to the audio thread which means we need to store the settings here. */ struct { ma_atomic_float volumeBeg; ma_atomic_float volumeEnd; ma_atomic_uint64 fadeLengthInFrames; /* <-- Defaults to (~(ma_uint64)0) which is used to indicate that no fade should be applied. */ ma_atomic_uint64 absoluteGlobalTimeInFrames; /* <-- The time to start the fade. */ } fadeSettings; /* Memory management. */ ma_bool8 _ownsHeap; void* _pHeap; } ma_engine_node; MA_API ma_result ma_engine_node_get_heap_size(const ma_engine_node_config* pConfig, size_t* pHeapSizeInBytes); MA_API ma_result ma_engine_node_init_preallocated(const ma_engine_node_config* pConfig, void* pHeap, ma_engine_node* pEngineNode); MA_API ma_result ma_engine_node_init(const ma_engine_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_engine_node* pEngineNode); MA_API void ma_engine_node_uninit(ma_engine_node* pEngineNode, const ma_allocation_callbacks* pAllocationCallbacks); #define MA_SOUND_SOURCE_CHANNEL_COUNT 0xFFFFFFFF /* Callback for when a sound reaches the end. */ typedef void (* ma_sound_end_proc)(void* pUserData, ma_sound* pSound); typedef struct { const char* pFilePath; /* Set this to load from the resource manager. */ const wchar_t* pFilePathW; /* Set this to load from the resource manager. */ ma_data_source* pDataSource; /* Set this to load from an existing data source. */ ma_node* pInitialAttachment; /* If set, the sound will be attached to an input of this node. This can be set to a ma_sound. If set to NULL, the sound will be attached directly to the endpoint unless MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT is set in `flags`. */ ma_uint32 initialAttachmentInputBusIndex; /* The index of the input bus of pInitialAttachment to attach the sound to. */ ma_uint32 channelsIn; /* Ignored if using a data source as input (the data source's channel count will be used always). Otherwise, setting to 0 will cause the engine's channel count to be used. */ ma_uint32 channelsOut; /* Set this to 0 (default) to use the engine's channel count. Set to MA_SOUND_SOURCE_CHANNEL_COUNT to use the data source's channel count (only used if using a data source as input). */ ma_mono_expansion_mode monoExpansionMode; /* Controls how the mono channel should be expanded to other channels when spatialization is disabled on a sound. */ ma_uint32 flags; /* A combination of MA_SOUND_FLAG_* flags. */ ma_uint32 volumeSmoothTimeInPCMFrames; /* The number of frames to smooth over volume changes. Defaults to 0 in which case no smoothing is used. */ ma_uint64 initialSeekPointInPCMFrames; /* Initializes the sound such that it's seeked to this location by default. */ ma_uint64 rangeBegInPCMFrames; ma_uint64 rangeEndInPCMFrames; ma_uint64 loopPointBegInPCMFrames; ma_uint64 loopPointEndInPCMFrames; ma_sound_end_proc endCallback; /* Fired when the sound reaches the end. Will be fired from the audio thread. Do not restart, uninitialize or otherwise change the state of the sound from here. Instead fire an event or set a variable to indicate to a different thread to change the start of the sound. Will not be fired in response to a scheduled stop with ma_sound_set_stop_time_*(). */ void* pEndCallbackUserData; ma_resampler_config pitchResampling; #ifndef MA_NO_RESOURCE_MANAGER ma_resource_manager_pipeline_notifications initNotifications; #endif ma_fence* pDoneFence; /* Deprecated. Use initNotifications instead. Released when the resource manager has finished decoding the entire sound. Not used with streams. */ ma_bool32 isLooping; /* Deprecated. Use the MA_SOUND_FLAG_LOOPING flag in `flags` instead. */ } ma_sound_config; MA_API ma_sound_config ma_sound_config_init(void); /* Deprecated. Will be removed in version 0.12. Use ma_sound_config_2() instead. */ MA_API ma_sound_config ma_sound_config_init_2(ma_engine* pEngine); /* Will be renamed to ma_sound_config_init() in version 0.12. */ struct ma_sound { ma_engine_node engineNode; /* Must be the first member for compatibility with the ma_node API. */ ma_data_source* pDataSource; MA_ATOMIC(8, ma_uint64) seekTarget; /* The PCM frame index to seek to in the mixing thread. Set to (~(ma_uint64)0) to not perform any seeking. */ MA_ATOMIC(4, ma_bool32) atEnd; ma_sound_end_proc endCallback; void* pEndCallbackUserData; float* pProcessingCache; /* Will be null if pDataSource is null. */ ma_uint32 processingCacheFramesRemaining; ma_uint32 processingCacheCap; ma_bool8 ownsDataSource; /* We're declaring a resource manager data source object here to save us a malloc when loading a sound via the resource manager, which I *think* will be the most common scenario. */ #ifndef MA_NO_RESOURCE_MANAGER ma_resource_manager_data_source* pResourceManagerDataSource; #endif }; /* Structure specifically for sounds played with ma_engine_play_sound(). Making this a separate structure to reduce overhead. */ typedef struct ma_sound_inlined ma_sound_inlined; struct ma_sound_inlined { ma_sound sound; ma_sound_inlined* pNext; ma_sound_inlined* pPrev; }; /* A sound group is just a sound. */ typedef ma_sound_config ma_sound_group_config; typedef ma_sound ma_sound_group; MA_API ma_sound_group_config ma_sound_group_config_init(void); /* Deprecated. Will be removed in version 0.12. Use ma_sound_config_2() instead. */ MA_API ma_sound_group_config ma_sound_group_config_init_2(ma_engine* pEngine); /* Will be renamed to ma_sound_config_init() in version 0.12. */ typedef void (* ma_engine_process_proc)(void* pUserData, float* pFramesOut, ma_uint64 frameCount); typedef struct { #if !defined(MA_NO_RESOURCE_MANAGER) ma_resource_manager* pResourceManager; /* Can be null in which case a resource manager will be created for you. */ #endif #if !defined(MA_NO_DEVICE_IO) ma_context* pContext; ma_device* pDevice; /* If set, the caller is responsible for calling ma_engine_data_callback() in the device's data callback. */ ma_device_id* pPlaybackDeviceID; /* The ID of the playback device to use with the default listener. */ ma_device_data_proc dataCallback; /* Can be null. Can be used to provide a custom device data callback. */ ma_device_notification_proc notificationCallback; #endif ma_log* pLog; /* When set to NULL, will use the context's log. */ ma_uint32 listenerCount; /* Must be between 1 and MA_ENGINE_MAX_LISTENERS. */ ma_uint32 channels; /* The number of channels to use when mixing and spatializing. When set to 0, will use the native channel count of the device. */ ma_uint32 sampleRate; /* The sample rate. When set to 0 will use the native sample rate of the device. */ ma_uint32 periodSizeInFrames; /* If set to something other than 0, updates will always be exactly this size. The underlying device may be a different size, but from the perspective of the mixer that won't matter.*/ ma_uint32 periodSizeInMilliseconds; /* Used if periodSizeInFrames is unset. */ ma_uint32 gainSmoothTimeInFrames; /* The number of frames to interpolate the gain of spatialized sounds across. If set to 0, will use gainSmoothTimeInMilliseconds. */ ma_uint32 gainSmoothTimeInMilliseconds; /* When set to 0, gainSmoothTimeInFrames will be used. If both are set to 0, a default value will be used. */ ma_uint32 defaultVolumeSmoothTimeInPCMFrames; /* Defaults to 0. Controls the default amount of smoothing to apply to volume changes to sounds. High values means more smoothing at the expense of high latency (will take longer to reach the new volume). */ ma_uint32 preMixStackSizeInBytes; /* A stack is used for internal processing in the node graph. This allows you to configure the size of this stack. Smaller values will reduce the maximum depth of your node graph. You should rarely need to modify this. */ ma_allocation_callbacks allocationCallbacks; ma_bool32 noAutoStart; /* When set to true, requires an explicit call to ma_engine_start(). This is false by default, meaning the engine will be started automatically in ma_engine_init(). */ ma_bool32 noDevice; /* When set to true, don't create a default device. ma_engine_read_pcm_frames() can be called manually to read data. */ ma_mono_expansion_mode monoExpansionMode; /* Controls how the mono channel should be expanded to other channels when spatialization is disabled on a sound. */ ma_vfs* pResourceManagerVFS; /* A pointer to a pre-allocated VFS object to use with the resource manager. This is ignored if pResourceManager is not NULL. */ ma_engine_process_proc onProcess; /* Fired at the end of each call to ma_engine_read_pcm_frames(). For engine's that manage their own internal device (the default configuration), this will be fired from the audio thread, and you do not need to call ma_engine_read_pcm_frames() manually in order to trigger this. */ void* pProcessUserData; /* User data that's passed into onProcess. */ ma_resampler_config resourceManagerResampling; /* The resampling config to use with the resource manager. */ ma_resampler_config pitchResampling; /* The resampling config for the pitch and Doppler effects. You will typically want this to be a fast resampler. For high quality stuff, it's recommended that you pre-resample. */ } ma_engine_config; MA_API ma_engine_config ma_engine_config_init(void); struct ma_engine { ma_node_graph nodeGraph; /* An engine is a node graph. It should be able to be plugged into any ma_node_graph API (with a cast) which means this must be the first member of this struct. */ #if !defined(MA_NO_RESOURCE_MANAGER) ma_resource_manager* pResourceManager; #endif #if !defined(MA_NO_DEVICE_IO) ma_device* pDevice; /* Optionally set via the config, otherwise allocated by the engine in ma_engine_init(). */ #endif ma_log* pLog; ma_uint32 sampleRate; ma_uint32 listenerCount; ma_spatializer_listener listeners[MA_ENGINE_MAX_LISTENERS]; ma_allocation_callbacks allocationCallbacks; ma_bool8 ownsResourceManager; ma_bool8 ownsDevice; ma_spinlock inlinedSoundLock; /* For synchronizing access to the inlined sound list. */ ma_sound_inlined* pInlinedSoundHead; /* The first inlined sound. Inlined sounds are tracked in a linked list. */ MA_ATOMIC(4, ma_uint32) inlinedSoundCount; /* The total number of allocated inlined sound objects. Used for debugging. */ ma_uint32 gainSmoothTimeInFrames; /* The number of frames to interpolate the gain of spatialized sounds across. */ ma_uint32 defaultVolumeSmoothTimeInPCMFrames; ma_mono_expansion_mode monoExpansionMode; ma_engine_process_proc onProcess; void* pProcessUserData; ma_resampler_config pitchResamplingConfig; }; MA_API ma_result ma_engine_init(const ma_engine_config* pConfig, ma_engine* pEngine); MA_API void ma_engine_uninit(ma_engine* pEngine); MA_API ma_result ma_engine_read_pcm_frames(ma_engine* pEngine, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); MA_API ma_node_graph* ma_engine_get_node_graph(ma_engine* pEngine); #if !defined(MA_NO_RESOURCE_MANAGER) MA_API ma_resource_manager* ma_engine_get_resource_manager(ma_engine* pEngine); #endif MA_API ma_device* ma_engine_get_device(ma_engine* pEngine); MA_API ma_log* ma_engine_get_log(ma_engine* pEngine); MA_API ma_node* ma_engine_get_endpoint(ma_engine* pEngine); MA_API ma_uint64 ma_engine_get_time_in_pcm_frames(const ma_engine* pEngine); MA_API ma_uint64 ma_engine_get_time_in_milliseconds(const ma_engine* pEngine); MA_API ma_result ma_engine_set_time_in_pcm_frames(ma_engine* pEngine, ma_uint64 globalTime); MA_API ma_result ma_engine_set_time_in_milliseconds(ma_engine* pEngine, ma_uint64 globalTime); MA_API ma_uint64 ma_engine_get_time(const ma_engine* pEngine); /* Deprecated. Use ma_engine_get_time_in_pcm_frames(). Will be removed in version 0.12. */ MA_API ma_result ma_engine_set_time(ma_engine* pEngine, ma_uint64 globalTime); /* Deprecated. Use ma_engine_set_time_in_pcm_frames(). Will be removed in version 0.12. */ MA_API ma_uint32 ma_engine_get_channels(const ma_engine* pEngine); MA_API ma_uint32 ma_engine_get_sample_rate(const ma_engine* pEngine); MA_API ma_result ma_engine_start(ma_engine* pEngine); MA_API ma_result ma_engine_stop(ma_engine* pEngine); MA_API ma_result ma_engine_set_volume(ma_engine* pEngine, float volume); MA_API float ma_engine_get_volume(ma_engine* pEngine); MA_API ma_result ma_engine_set_gain_db(ma_engine* pEngine, float gainDB); MA_API float ma_engine_get_gain_db(ma_engine* pEngine); MA_API ma_uint32 ma_engine_get_listener_count(const ma_engine* pEngine); MA_API ma_uint32 ma_engine_find_closest_listener(const ma_engine* pEngine, float absolutePosX, float absolutePosY, float absolutePosZ); MA_API void ma_engine_listener_set_position(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z); MA_API ma_vec3f ma_engine_listener_get_position(const ma_engine* pEngine, ma_uint32 listenerIndex); MA_API void ma_engine_listener_set_direction(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z); MA_API ma_vec3f ma_engine_listener_get_direction(const ma_engine* pEngine, ma_uint32 listenerIndex); MA_API void ma_engine_listener_set_velocity(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z); MA_API ma_vec3f ma_engine_listener_get_velocity(const ma_engine* pEngine, ma_uint32 listenerIndex); MA_API void ma_engine_listener_set_cone(ma_engine* pEngine, ma_uint32 listenerIndex, float innerAngleInRadians, float outerAngleInRadians, float outerGain); MA_API void ma_engine_listener_get_cone(const ma_engine* pEngine, ma_uint32 listenerIndex, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain); MA_API void ma_engine_listener_set_world_up(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z); MA_API ma_vec3f ma_engine_listener_get_world_up(const ma_engine* pEngine, ma_uint32 listenerIndex); MA_API void ma_engine_listener_set_enabled(ma_engine* pEngine, ma_uint32 listenerIndex, ma_bool32 isEnabled); MA_API ma_bool32 ma_engine_listener_is_enabled(const ma_engine* pEngine, ma_uint32 listenerIndex); #ifndef MA_NO_RESOURCE_MANAGER MA_API ma_result ma_engine_play_sound_ex(ma_engine* pEngine, const char* pFilePath, ma_node* pNode, ma_uint32 nodeInputBusIndex); MA_API ma_result ma_engine_play_sound(ma_engine* pEngine, const char* pFilePath, ma_sound_group* pGroup); /* Fire and forget. */ #endif #ifndef MA_NO_RESOURCE_MANAGER MA_API ma_result ma_sound_init_from_file(ma_engine* pEngine, const char* pFilePath, ma_uint32 flags, ma_sound_group* pGroup, ma_fence* pDoneFence, ma_sound* pSound); MA_API ma_result ma_sound_init_from_file_w(ma_engine* pEngine, const wchar_t* pFilePath, ma_uint32 flags, ma_sound_group* pGroup, ma_fence* pDoneFence, ma_sound* pSound); MA_API ma_result ma_sound_init_copy(ma_engine* pEngine, const ma_sound* pExistingSound, ma_uint32 flags, ma_sound_group* pGroup, ma_sound* pSound); #endif MA_API ma_result ma_sound_init_from_data_source(ma_engine* pEngine, ma_data_source* pDataSource, ma_uint32 flags, ma_sound_group* pGroup, ma_sound* pSound); MA_API ma_result ma_sound_init_ex(ma_engine* pEngine, const ma_sound_config* pConfig, ma_sound* pSound); MA_API void ma_sound_uninit(ma_sound* pSound); MA_API ma_engine* ma_sound_get_engine(const ma_sound* pSound); MA_API ma_data_source* ma_sound_get_data_source(const ma_sound* pSound); MA_API ma_result ma_sound_start(ma_sound* pSound); MA_API ma_result ma_sound_stop(ma_sound* pSound); MA_API ma_result ma_sound_stop_with_fade_in_pcm_frames(ma_sound* pSound, ma_uint64 fadeLengthInFrames); /* Will overwrite any scheduled stop and fade. If you want to restart the sound, first reset it with `ma_sound_reset_stop_time_and_fade()`. There are plans to make this less awkward in the future. */ MA_API ma_result ma_sound_stop_with_fade_in_milliseconds(ma_sound* pSound, ma_uint64 fadeLengthInFrames); /* Will overwrite any scheduled stop and fade. If you want to restart the sound, first reset it with `ma_sound_reset_stop_time_and_fade()`. There are plans to make this less awkward in the future. */ MA_API void ma_sound_reset_start_time(ma_sound* pSound); MA_API void ma_sound_reset_stop_time(ma_sound* pSound); MA_API void ma_sound_reset_fade(ma_sound* pSound); MA_API void ma_sound_reset_stop_time_and_fade(ma_sound* pSound); /* Resets fades and scheduled stop time. Does not seek back to the start. */ MA_API void ma_sound_set_volume(ma_sound* pSound, float volume); MA_API float ma_sound_get_volume(const ma_sound* pSound); MA_API void ma_sound_set_pan(ma_sound* pSound, float pan); MA_API float ma_sound_get_pan(const ma_sound* pSound); MA_API void ma_sound_set_pan_mode(ma_sound* pSound, ma_pan_mode panMode); MA_API ma_pan_mode ma_sound_get_pan_mode(const ma_sound* pSound); MA_API void ma_sound_set_pitch(ma_sound* pSound, float pitch); MA_API float ma_sound_get_pitch(const ma_sound* pSound); MA_API void ma_sound_set_spatialization_enabled(ma_sound* pSound, ma_bool32 enabled); MA_API ma_bool32 ma_sound_is_spatialization_enabled(const ma_sound* pSound); MA_API void ma_sound_set_pinned_listener_index(ma_sound* pSound, ma_uint32 listenerIndex); MA_API ma_uint32 ma_sound_get_pinned_listener_index(const ma_sound* pSound); MA_API ma_uint32 ma_sound_get_listener_index(const ma_sound* pSound); MA_API ma_vec3f ma_sound_get_direction_to_listener(const ma_sound* pSound); MA_API void ma_sound_set_position(ma_sound* pSound, float x, float y, float z); MA_API ma_vec3f ma_sound_get_position(const ma_sound* pSound); MA_API void ma_sound_set_direction(ma_sound* pSound, float x, float y, float z); MA_API ma_vec3f ma_sound_get_direction(const ma_sound* pSound); MA_API void ma_sound_set_velocity(ma_sound* pSound, float x, float y, float z); MA_API ma_vec3f ma_sound_get_velocity(const ma_sound* pSound); MA_API void ma_sound_set_attenuation_model(ma_sound* pSound, ma_attenuation_model attenuationModel); MA_API ma_attenuation_model ma_sound_get_attenuation_model(const ma_sound* pSound); MA_API void ma_sound_set_positioning(ma_sound* pSound, ma_positioning positioning); MA_API ma_positioning ma_sound_get_positioning(const ma_sound* pSound); MA_API void ma_sound_set_rolloff(ma_sound* pSound, float rolloff); MA_API float ma_sound_get_rolloff(const ma_sound* pSound); MA_API void ma_sound_set_min_gain(ma_sound* pSound, float minGain); MA_API float ma_sound_get_min_gain(const ma_sound* pSound); MA_API void ma_sound_set_max_gain(ma_sound* pSound, float maxGain); MA_API float ma_sound_get_max_gain(const ma_sound* pSound); MA_API void ma_sound_set_min_distance(ma_sound* pSound, float minDistance); MA_API float ma_sound_get_min_distance(const ma_sound* pSound); MA_API void ma_sound_set_max_distance(ma_sound* pSound, float maxDistance); MA_API float ma_sound_get_max_distance(const ma_sound* pSound); MA_API void ma_sound_set_cone(ma_sound* pSound, float innerAngleInRadians, float outerAngleInRadians, float outerGain); MA_API void ma_sound_get_cone(const ma_sound* pSound, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain); MA_API void ma_sound_set_doppler_factor(ma_sound* pSound, float dopplerFactor); MA_API float ma_sound_get_doppler_factor(const ma_sound* pSound); MA_API void ma_sound_set_directional_attenuation_factor(ma_sound* pSound, float directionalAttenuationFactor); MA_API float ma_sound_get_directional_attenuation_factor(const ma_sound* pSound); MA_API void ma_sound_set_fade_in_pcm_frames(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInFrames); MA_API void ma_sound_set_fade_in_milliseconds(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInMilliseconds); MA_API void ma_sound_set_fade_start_in_pcm_frames(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInFrames, ma_uint64 absoluteGlobalTimeInFrames); MA_API void ma_sound_set_fade_start_in_milliseconds(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInMilliseconds, ma_uint64 absoluteGlobalTimeInMilliseconds); MA_API float ma_sound_get_current_fade_volume(const ma_sound* pSound); MA_API void ma_sound_set_start_time_in_pcm_frames(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInFrames); MA_API void ma_sound_set_start_time_in_milliseconds(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInMilliseconds); MA_API void ma_sound_set_stop_time_in_pcm_frames(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInFrames); MA_API void ma_sound_set_stop_time_in_milliseconds(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInMilliseconds); MA_API void ma_sound_set_stop_time_with_fade_in_pcm_frames(ma_sound* pSound, ma_uint64 stopAbsoluteGlobalTimeInFrames, ma_uint64 fadeLengthInFrames); MA_API void ma_sound_set_stop_time_with_fade_in_milliseconds(ma_sound* pSound, ma_uint64 stopAbsoluteGlobalTimeInMilliseconds, ma_uint64 fadeLengthInMilliseconds); MA_API ma_bool32 ma_sound_is_playing(const ma_sound* pSound); MA_API ma_uint64 ma_sound_get_time_in_pcm_frames(const ma_sound* pSound); MA_API ma_uint64 ma_sound_get_time_in_milliseconds(const ma_sound* pSound); MA_API void ma_sound_set_looping(ma_sound* pSound, ma_bool32 isLooping); MA_API ma_bool32 ma_sound_is_looping(const ma_sound* pSound); MA_API ma_bool32 ma_sound_at_end(const ma_sound* pSound); MA_API ma_result ma_sound_seek_to_pcm_frame(ma_sound* pSound, ma_uint64 frameIndex); /* Just a wrapper around ma_data_source_seek_to_pcm_frame(). */ MA_API ma_result ma_sound_seek_to_second(ma_sound* pSound, float seekPointInSeconds); /* Abstraction to ma_sound_seek_to_pcm_frame() */ MA_API ma_result ma_sound_get_data_format(const ma_sound* pSound, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap); MA_API ma_result ma_sound_get_cursor_in_pcm_frames(const ma_sound* pSound, ma_uint64* pCursor); MA_API ma_result ma_sound_get_length_in_pcm_frames(const ma_sound* pSound, ma_uint64* pLength); MA_API ma_result ma_sound_get_cursor_in_seconds(const ma_sound* pSound, float* pCursor); MA_API ma_result ma_sound_get_length_in_seconds(const ma_sound* pSound, float* pLength); MA_API ma_result ma_sound_set_end_callback(ma_sound* pSound, ma_sound_end_proc callback, void* pUserData); MA_API ma_result ma_sound_group_init(ma_engine* pEngine, ma_uint32 flags, ma_sound_group* pParentGroup, ma_sound_group* pGroup); MA_API ma_result ma_sound_group_init_ex(ma_engine* pEngine, const ma_sound_group_config* pConfig, ma_sound_group* pGroup); MA_API void ma_sound_group_uninit(ma_sound_group* pGroup); MA_API ma_engine* ma_sound_group_get_engine(const ma_sound_group* pGroup); MA_API ma_result ma_sound_group_start(ma_sound_group* pGroup); MA_API ma_result ma_sound_group_stop(ma_sound_group* pGroup); MA_API void ma_sound_group_set_volume(ma_sound_group* pGroup, float volume); MA_API float ma_sound_group_get_volume(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_pan(ma_sound_group* pGroup, float pan); MA_API float ma_sound_group_get_pan(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_pan_mode(ma_sound_group* pGroup, ma_pan_mode panMode); MA_API ma_pan_mode ma_sound_group_get_pan_mode(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_pitch(ma_sound_group* pGroup, float pitch); MA_API float ma_sound_group_get_pitch(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_spatialization_enabled(ma_sound_group* pGroup, ma_bool32 enabled); MA_API ma_bool32 ma_sound_group_is_spatialization_enabled(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_pinned_listener_index(ma_sound_group* pGroup, ma_uint32 listenerIndex); MA_API ma_uint32 ma_sound_group_get_pinned_listener_index(const ma_sound_group* pGroup); MA_API ma_uint32 ma_sound_group_get_listener_index(const ma_sound_group* pGroup); MA_API ma_vec3f ma_sound_group_get_direction_to_listener(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_position(ma_sound_group* pGroup, float x, float y, float z); MA_API ma_vec3f ma_sound_group_get_position(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_direction(ma_sound_group* pGroup, float x, float y, float z); MA_API ma_vec3f ma_sound_group_get_direction(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_velocity(ma_sound_group* pGroup, float x, float y, float z); MA_API ma_vec3f ma_sound_group_get_velocity(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_attenuation_model(ma_sound_group* pGroup, ma_attenuation_model attenuationModel); MA_API ma_attenuation_model ma_sound_group_get_attenuation_model(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_positioning(ma_sound_group* pGroup, ma_positioning positioning); MA_API ma_positioning ma_sound_group_get_positioning(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_rolloff(ma_sound_group* pGroup, float rolloff); MA_API float ma_sound_group_get_rolloff(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_min_gain(ma_sound_group* pGroup, float minGain); MA_API float ma_sound_group_get_min_gain(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_max_gain(ma_sound_group* pGroup, float maxGain); MA_API float ma_sound_group_get_max_gain(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_min_distance(ma_sound_group* pGroup, float minDistance); MA_API float ma_sound_group_get_min_distance(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_max_distance(ma_sound_group* pGroup, float maxDistance); MA_API float ma_sound_group_get_max_distance(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_cone(ma_sound_group* pGroup, float innerAngleInRadians, float outerAngleInRadians, float outerGain); MA_API void ma_sound_group_get_cone(const ma_sound_group* pGroup, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain); MA_API void ma_sound_group_set_doppler_factor(ma_sound_group* pGroup, float dopplerFactor); MA_API float ma_sound_group_get_doppler_factor(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_directional_attenuation_factor(ma_sound_group* pGroup, float directionalAttenuationFactor); MA_API float ma_sound_group_get_directional_attenuation_factor(const ma_sound_group* pGroup); MA_API void ma_sound_group_set_fade_in_pcm_frames(ma_sound_group* pGroup, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInFrames); MA_API void ma_sound_group_set_fade_in_milliseconds(ma_sound_group* pGroup, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInMilliseconds); MA_API float ma_sound_group_get_current_fade_volume(ma_sound_group* pGroup); MA_API void ma_sound_group_set_start_time_in_pcm_frames(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInFrames); MA_API void ma_sound_group_set_start_time_in_milliseconds(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInMilliseconds); MA_API void ma_sound_group_set_stop_time_in_pcm_frames(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInFrames); MA_API void ma_sound_group_set_stop_time_in_milliseconds(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInMilliseconds); MA_API ma_bool32 ma_sound_group_is_playing(const ma_sound_group* pGroup); MA_API ma_uint64 ma_sound_group_get_time_in_pcm_frames(const ma_sound_group* pGroup); #endif /* MA_NO_ENGINE */ /* END SECTION: miniaudio_engine.h */ #ifdef __cplusplus } #endif #endif /* miniaudio_h */ /* This is for preventing greying out of the implementation section. */ #if defined(Q_CREATOR_RUN) || defined(__INTELLISENSE__) || defined(__CDT_PARSER__) #define MINIAUDIO_IMPLEMENTATION #endif /************************************************************************************************************************************************************ ************************************************************************************************************************************************************* IMPLEMENTATION ************************************************************************************************************************************************************* ************************************************************************************************************************************************************/ #if defined(MINIAUDIO_IMPLEMENTATION) || defined(MA_IMPLEMENTATION) #ifndef miniaudio_c #define miniaudio_c #include <assert.h> #include <limits.h> /* For INT_MAX */ #include <math.h> /* sin(), etc. */ #include <stdlib.h> /* For malloc(), free(), wcstombs(). */ #include <string.h> /* For memset() */ #include <stdarg.h> #include <stdio.h> #if !defined(_MSC_VER) && !defined(__DMC__) #include <strings.h> /* For strcasecmp(). */ #include <wchar.h> /* For wcslen(), wcsrtombs() */ #endif #ifdef _MSC_VER #include <float.h> /* For _controlfp_s constants */ #endif #if defined(MA_WIN32) #include <windows.h> /* There's a possibility that WIN32_LEAN_AND_MEAN has been defined which will exclude some symbols such as STGM_READ and CLSCTL_ALL. We need to check these and define them ourselves if they're unavailable. */ #ifndef STGM_READ #define STGM_READ 0x00000000L #endif #ifndef CLSCTX_ALL #define CLSCTX_ALL 23 #endif /* IUnknown is used by both the WASAPI and DirectSound backends. It easier to just declare our version here. */ typedef struct ma_IUnknown ma_IUnknown; #endif #if !defined(MA_WIN32) #if !defined(MA_NO_THREADING) #include <sched.h> #include <pthread.h> /* For pthreads. */ #endif #include <sys/time.h> /* select() (used for ma_sleep()). */ #include <time.h> /* For nanosleep() */ #include <unistd.h> #endif /* For fstat(), etc. */ #if defined(MA_XBOX_NXDK) #include <stat.h> /* Suggestion for NXDK: Add a sys/stat.h wrapper for compatibility. */ #else #include <sys/stat.h> #endif #ifdef MA_EMSCRIPTEN #include <emscripten/emscripten.h> #endif /* Architecture Detection */ #if !defined(MA_64BIT) && !defined(MA_32BIT) #ifdef _WIN32 #ifdef _WIN64 #define MA_64BIT #else #define MA_32BIT #endif #endif #endif #if !defined(MA_64BIT) && !defined(MA_32BIT) #ifdef __GNUC__ #ifdef __LP64__ #define MA_64BIT #else #define MA_32BIT #endif #endif #endif #if !defined(MA_64BIT) && !defined(MA_32BIT) #include <stdint.h> #if INTPTR_MAX == INT64_MAX #define MA_64BIT #else #define MA_32BIT #endif #endif #if defined(__arm__) || defined(_M_ARM) #define MA_ARM32 #endif #if defined(__arm64) || defined(__arm64__) || defined(__aarch64__) || defined(_M_ARM64) #define MA_ARM64 #endif #if defined(__x86_64__) || defined(_M_X64) #define MA_X64 #elif defined(__i386) || defined(_M_IX86) #define MA_X86 #elif defined(MA_ARM32) || defined(MA_ARM64) #define MA_ARM #endif /* Intrinsics Support */ #if defined(MA_X64) || defined(MA_X86) #if defined(_MSC_VER) && !defined(__clang__) /* MSVC. */ #if _MSC_VER >= 1400 && !defined(MA_NO_SSE2) /* 2005 */ #define MA_SUPPORT_SSE2 #endif /*#if _MSC_VER >= 1600 && !defined(MA_NO_AVX)*/ /* 2010 */ /* #define MA_SUPPORT_AVX*/ /*#endif*/ #if _MSC_VER >= 1700 && !defined(MA_NO_AVX2) /* 2012 */ #define MA_SUPPORT_AVX2 #endif #else /* Assume GNUC-style. */ #if defined(__SSE2__) && !defined(MA_NO_SSE2) #define MA_SUPPORT_SSE2 #endif /*#if defined(__AVX__) && !defined(MA_NO_AVX)*/ /* #define MA_SUPPORT_AVX*/ /*#endif*/ #if defined(__AVX2__) && !defined(MA_NO_AVX2) #define MA_SUPPORT_AVX2 #endif #endif /* If at this point we still haven't determined compiler support for the intrinsics just fall back to __has_include. */ #if !defined(__GNUC__) && !defined(__clang__) && defined(__has_include) #if !defined(MA_SUPPORT_SSE2) && !defined(MA_NO_SSE2) && __has_include(<emmintrin.h>) #define MA_SUPPORT_SSE2 #endif /*#if !defined(MA_SUPPORT_AVX) && !defined(MA_NO_AVX) && __has_include(<immintrin.h>)*/ /* #define MA_SUPPORT_AVX*/ /*#endif*/ #if !defined(MA_SUPPORT_AVX2) && !defined(MA_NO_AVX2) && __has_include(<immintrin.h>) #define MA_SUPPORT_AVX2 #endif #endif #if defined(MA_SUPPORT_AVX2) || defined(MA_SUPPORT_AVX) #include <immintrin.h> #elif defined(MA_SUPPORT_SSE2) #include <emmintrin.h> #endif #endif #if defined(MA_ARM) #if !defined(MA_NO_NEON) && (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64)) #define MA_SUPPORT_NEON #include <arm_neon.h> #endif #endif /* Begin globally disabled warnings. */ #if defined(_MSC_VER) #pragma warning(push) #pragma warning(disable:4752) /* found Intel(R) Advanced Vector Extensions; consider using /arch:AVX */ #pragma warning(disable:4049) /* compiler limit : terminating line number emission */ #endif #if defined(MA_X64) || defined(MA_X86) #if defined(_MSC_VER) && !defined(__clang__) #if _MSC_VER >= 1400 #include <intrin.h> static MA_INLINE void ma_cpuid(int info[4], int fid) { __cpuid(info, fid); } #else #define MA_NO_CPUID #endif #if _MSC_VER >= 1600 && (defined(_MSC_FULL_VER) && _MSC_FULL_VER >= 160040219) static MA_INLINE unsigned __int64 ma_xgetbv(int reg) { return _xgetbv(reg); } #else #define MA_NO_XGETBV #endif #elif (defined(__GNUC__) || defined(__clang__)) && !defined(MA_ANDROID) static MA_INLINE void ma_cpuid(int info[4], int fid) { /* It looks like the -fPIC option uses the ebx register which GCC complains about. We can work around this by just using a different register, the specific register of which I'm letting the compiler decide on. The "k" prefix is used to specify a 32-bit register. The {...} syntax is for supporting different assembly dialects. What's basically happening is that we're saving and restoring the ebx register manually. */ #if defined(MA_X86) && defined(__PIC__) __asm__ __volatile__ ( "xchg{l} {%%}ebx, %k1;" "cpuid;" "xchg{l} {%%}ebx, %k1;" : "=a"(info[0]), "=&r"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0) ); #else __asm__ __volatile__ ( "cpuid" : "=a"(info[0]), "=b"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0) ); #endif } static MA_INLINE ma_uint64 ma_xgetbv(int reg) { unsigned int hi; unsigned int lo; __asm__ __volatile__ ( "xgetbv" : "=a"(lo), "=d"(hi) : "c"(reg) ); return ((ma_uint64)hi << 32) | (ma_uint64)lo; } #else #define MA_NO_CPUID #define MA_NO_XGETBV #endif #else #define MA_NO_CPUID #define MA_NO_XGETBV #endif static MA_INLINE ma_bool32 ma_has_sse2(void) { #if defined(MA_SUPPORT_SSE2) #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_SSE2) #if defined(MA_X64) return MA_TRUE; /* 64-bit targets always support SSE2. */ #elif (defined(_M_IX86_FP) && _M_IX86_FP == 2) || defined(__SSE2__) return MA_TRUE; /* If the compiler is allowed to freely generate SSE2 code we can assume support. */ #else #if defined(MA_NO_CPUID) return MA_FALSE; #else int info[4]; ma_cpuid(info, 1); return (info[3] & (1 << 26)) != 0; #endif #endif #else return MA_FALSE; /* SSE2 is only supported on x86 and x64 architectures. */ #endif #else return MA_FALSE; /* No compiler support. */ #endif } #if 0 static MA_INLINE ma_bool32 ma_has_avx() { #if defined(MA_SUPPORT_AVX) #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX) #if defined(_AVX_) || defined(__AVX__) return MA_TRUE; /* If the compiler is allowed to freely generate AVX code we can assume support. */ #else /* AVX requires both CPU and OS support. */ #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV) return MA_FALSE; #else int info[4]; ma_cpuid(info, 1); if (((info[2] & (1 << 27)) != 0) && ((info[2] & (1 << 28)) != 0)) { ma_uint64 xrc = ma_xgetbv(0); if ((xrc & 0x06) == 0x06) { return MA_TRUE; } else { return MA_FALSE; } } else { return MA_FALSE; } #endif #endif #else return MA_FALSE; /* AVX is only supported on x86 and x64 architectures. */ #endif #else return MA_FALSE; /* No compiler support. */ #endif } #endif static MA_INLINE ma_bool32 ma_has_avx2(void) { #if defined(MA_SUPPORT_AVX2) #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX2) #if defined(_AVX2_) || defined(__AVX2__) return MA_TRUE; /* If the compiler is allowed to freely generate AVX2 code we can assume support. */ #else /* AVX2 requires both CPU and OS support. */ #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV) return MA_FALSE; #else int info1[4]; int info7[4]; ma_cpuid(info1, 1); ma_cpuid(info7, 7); if (((info1[2] & (1 << 27)) != 0) && ((info7[1] & (1 << 5)) != 0)) { ma_uint64 xrc = ma_xgetbv(0); if ((xrc & 0x06) == 0x06) { return MA_TRUE; } else { return MA_FALSE; } } else { return MA_FALSE; } #endif #endif #else return MA_FALSE; /* AVX2 is only supported on x86 and x64 architectures. */ #endif #else return MA_FALSE; /* No compiler support. */ #endif } static MA_INLINE ma_bool32 ma_has_neon(void) { #if defined(MA_SUPPORT_NEON) #if defined(MA_ARM) && !defined(MA_NO_NEON) #if (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64)) return MA_TRUE; /* If the compiler is allowed to freely generate NEON code we can assume support. */ #else /* TODO: Runtime check. */ return MA_FALSE; #endif #else return MA_FALSE; /* NEON is only supported on ARM architectures. */ #endif #else return MA_FALSE; /* No compiler support. */ #endif } #if defined(__has_builtin) #define MA_COMPILER_HAS_BUILTIN(x) __has_builtin(x) #else #define MA_COMPILER_HAS_BUILTIN(x) 0 #endif #ifndef MA_ASSUME #if MA_COMPILER_HAS_BUILTIN(__builtin_assume) #define MA_ASSUME(x) __builtin_assume(x) #elif MA_COMPILER_HAS_BUILTIN(__builtin_unreachable) #define MA_ASSUME(x) do { if (!(x)) __builtin_unreachable(); } while (0) #elif defined(_MSC_VER) #define MA_ASSUME(x) __assume(x) #else #define MA_ASSUME(x) (void)(x) #endif #endif #ifndef MA_RESTRICT #if defined(__clang__) || defined(_MSC_VER) || (defined(__GNUC__) && (__GNUC__ > 2 || (__GNUC__ == 2 && __GNUC_MINOR__ >= 95))) #define MA_RESTRICT __restrict #else #define MA_RESTRICT #endif #endif #if defined(_MSC_VER) && _MSC_VER >= 1400 #define MA_HAS_BYTESWAP16_INTRINSIC #define MA_HAS_BYTESWAP32_INTRINSIC #define MA_HAS_BYTESWAP64_INTRINSIC #elif defined(__clang__) #if MA_COMPILER_HAS_BUILTIN(__builtin_bswap16) #define MA_HAS_BYTESWAP16_INTRINSIC #endif #if MA_COMPILER_HAS_BUILTIN(__builtin_bswap32) #define MA_HAS_BYTESWAP32_INTRINSIC #endif #if MA_COMPILER_HAS_BUILTIN(__builtin_bswap64) #define MA_HAS_BYTESWAP64_INTRINSIC #endif #elif defined(__GNUC__) #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 3)) #define MA_HAS_BYTESWAP32_INTRINSIC #define MA_HAS_BYTESWAP64_INTRINSIC #endif #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8)) #define MA_HAS_BYTESWAP16_INTRINSIC #endif #endif static MA_INLINE ma_bool32 ma_is_little_endian(void) { #if defined(MA_X86) || defined(MA_X64) return MA_TRUE; #else int n = 1; return (*(char*)&n) == 1; #endif } static MA_INLINE ma_bool32 ma_is_big_endian(void) { return !ma_is_little_endian(); } static MA_INLINE ma_uint32 ma_swap_endian_uint32(ma_uint32 n) { #ifdef MA_HAS_BYTESWAP32_INTRINSIC #if defined(_MSC_VER) return _byteswap_ulong(n); #elif defined(__GNUC__) || defined(__clang__) #if defined(MA_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 6) && !defined(MA_64BIT) /* <-- 64-bit inline assembly has not been tested, so disabling for now. */ /* Inline assembly optimized implementation for ARM. In my testing, GCC does not generate optimized code with __builtin_bswap32(). */ ma_uint32 r; __asm__ __volatile__ ( #if defined(MA_64BIT) "rev %w[out], %w[in]" : [out]"=r"(r) : [in]"r"(n) /* <-- This is untested. If someone in the community could test this, that would be appreciated! */ #else "rev %[out], %[in]" : [out]"=r"(r) : [in]"r"(n) #endif ); return r; #else return __builtin_bswap32(n); #endif #else #error "This compiler does not support the byte swap intrinsic." #endif #else return ((n & 0xFF000000) >> 24) | ((n & 0x00FF0000) >> 8) | ((n & 0x0000FF00) << 8) | ((n & 0x000000FF) << 24); #endif } #if !defined(MA_EMSCRIPTEN) #ifdef MA_WIN32 static void ma_sleep__win32(ma_uint32 milliseconds) { Sleep((DWORD)milliseconds); } #endif #ifdef MA_POSIX static void ma_sleep__posix(ma_uint32 milliseconds) { #ifdef MA_EMSCRIPTEN (void)milliseconds; MA_ASSERT(MA_FALSE); /* The Emscripten build should never sleep. */ #else #if (defined(_POSIX_C_SOURCE) && _POSIX_C_SOURCE >= 199309L) || defined(MA_SWITCH) struct timespec ts; ts.tv_sec = milliseconds / 1000; ts.tv_nsec = milliseconds % 1000 * 1000000; nanosleep(&ts, NULL); #else struct timeval tv; tv.tv_sec = milliseconds / 1000; tv.tv_usec = milliseconds % 1000 * 1000; select(0, NULL, NULL, NULL, &tv); #endif #endif } #endif static MA_INLINE void ma_sleep(ma_uint32 milliseconds) { #ifdef MA_WIN32 ma_sleep__win32(milliseconds); #endif #ifdef MA_POSIX ma_sleep__posix(milliseconds); #endif } #endif static MA_INLINE void ma_yield(void) { #if defined(__i386) || defined(_M_IX86) || defined(__x86_64__) || defined(_M_X64) /* x86/x64 */ #if (defined(_MSC_VER) || defined(__WATCOMC__) || defined(__DMC__)) && !defined(__clang__) #if _MSC_VER >= 1400 _mm_pause(); #else #if defined(__DMC__) /* Digital Mars does not recognize the PAUSE opcode. Fall back to NOP. */ __asm nop; #else __asm pause; #endif #endif #else __asm__ __volatile__ ("rep; nop"); #endif #elif (defined(__arm__) && defined(__ARM_ARCH) && __ARM_ARCH >= 7) || defined(_M_ARM64) || (defined(_M_ARM) && _M_ARM >= 7) || defined(__ARM_ARCH_6K__) || defined(__ARM_ARCH_6T2__) /* ARM */ #if defined(_MSC_VER) /* Apparently there is a __yield() intrinsic that's compatible with ARM, but I cannot find documentation for it nor can I find where it's declared. */ __yield(); #else __asm__ __volatile__ ("yield"); /* ARMv6K/ARMv6T2 and above. */ #endif #else /* Unknown or unsupported architecture. No-op. */ #endif } #define MA_MM_DENORMALS_ZERO_MASK 0x0040 #define MA_MM_FLUSH_ZERO_MASK 0x8000 static MA_INLINE unsigned int ma_disable_denormals(void) { unsigned int prevState; #if defined(_MSC_VER) && !defined(MA_XBOX_NXDK) { /* Older versions of Visual Studio don't support the "safe" versions of _controlfp_s(). I don't know which version of Visual Studio first added support for _controlfp_s(), but I do know that VC6 lacks support. _MSC_VER = 1200 is VC6, but if you get compilation errors on older versions of Visual Studio, let me know and I'll make the necessary adjustment. */ #if _MSC_VER <= 1200 { prevState = _statusfp(); _controlfp(prevState | _DN_FLUSH, _MCW_DN); } #else { unsigned int unused; _controlfp_s(&prevState, 0, 0); _controlfp_s(&unused, prevState | _DN_FLUSH, _MCW_DN); } #endif } #elif defined(MA_X86) || defined(MA_X64) { #if defined(MA_SUPPORT_SSE2) && defined(__SSE2__) && !(defined(__TINYC__) || defined(__WATCOMC__)) /* <-- Add compilers that lack support for _mm_getcsr() and _mm_setcsr() to this list. */ { prevState = _mm_getcsr(); _mm_setcsr(prevState | MA_MM_DENORMALS_ZERO_MASK | MA_MM_FLUSH_ZERO_MASK); } #else { /* x88/64, but no support for _mm_getcsr()/_mm_setcsr(). May need to fall back to inlined assembly here. */ prevState = 0; } #endif } #else { /* Unknown or unsupported architecture. No-op. */ prevState = 0; } #endif return prevState; } static MA_INLINE void ma_restore_denormals(unsigned int prevState) { #if defined(_MSC_VER) && !defined(MA_XBOX_NXDK) { /* Older versions of Visual Studio do not support _controlfp_s(). See ma_disable_denormals(). */ #if _MSC_VER <= 1200 { _controlfp(prevState, _MCW_DN); } #else { unsigned int unused; _controlfp_s(&unused, prevState, _MCW_DN); } #endif } #elif defined(MA_X86) || defined(MA_X64) { #if defined(MA_SUPPORT_SSE2) && defined(__SSE2__) && !(defined(__TINYC__) || defined(__WATCOMC__)) /* <-- Add compilers that lack support for _mm_getcsr() and _mm_setcsr() to this list. */ { _mm_setcsr(prevState); } #else { /* x88/64, but no support for _mm_getcsr()/_mm_setcsr(). May need to fall back to inlined assembly here. */ (void)prevState; } #endif } #else { /* Unknown or unsupported architecture. No-op. */ (void)prevState; } #endif } #ifdef MA_ANDROID #include <sys/system_properties.h> int ma_android_sdk_version() { char sdkVersion[PROP_VALUE_MAX + 1] = {0, }; if (__system_property_get("ro.build.version.sdk", sdkVersion)) { return atoi(sdkVersion); } return 0; } #endif #ifndef MA_COINIT_VALUE #define MA_COINIT_VALUE 0 /* 0 = COINIT_MULTITHREADED */ #endif #ifndef MA_FLT_MAX #ifdef FLT_MAX #define MA_FLT_MAX FLT_MAX #else #define MA_FLT_MAX 3.402823466e+38F #endif #endif #ifndef MA_PI #define MA_PI 3.14159265358979323846264f #endif #ifndef MA_PI_D #define MA_PI_D 3.14159265358979323846264 #endif #ifndef MA_TAU #define MA_TAU 6.28318530717958647693f #endif #ifndef MA_TAU_D #define MA_TAU_D 6.28318530717958647693 #endif /* The default format when ma_format_unknown (0) is requested when initializing a device. */ #ifndef MA_DEFAULT_FORMAT #define MA_DEFAULT_FORMAT ma_format_f32 #endif /* The default channel count to use when 0 is used when initializing a device. */ #ifndef MA_DEFAULT_CHANNELS #define MA_DEFAULT_CHANNELS 2 #endif /* The default sample rate to use when 0 is used when initializing a device. */ #ifndef MA_DEFAULT_SAMPLE_RATE #define MA_DEFAULT_SAMPLE_RATE 48000 #endif /* Default periods when none is specified in ma_device_init(). More periods means more work on the CPU. */ #ifndef MA_DEFAULT_PERIODS #define MA_DEFAULT_PERIODS 3 #endif /* The default period size in milliseconds for low latency mode. */ #ifndef MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY #define MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY 10 #endif /* The default buffer size in milliseconds for conservative mode. */ #ifndef MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE #define MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE 100 #endif /* The default LPF filter order for linear resampling. Note that this is clamped to MA_MAX_FILTER_ORDER. */ #ifndef MA_DEFAULT_RESAMPLER_LPF_ORDER #if MA_MAX_FILTER_ORDER >= 4 #define MA_DEFAULT_RESAMPLER_LPF_ORDER 4 #else #define MA_DEFAULT_RESAMPLER_LPF_ORDER MA_MAX_FILTER_ORDER #endif #endif #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) #pragma GCC diagnostic push #pragma GCC diagnostic ignored "-Wunused-variable" #endif /* Standard sample rates, in order of priority. */ static ma_uint32 g_maStandardSampleRatePriorities[] = { (ma_uint32)ma_standard_sample_rate_48000, (ma_uint32)ma_standard_sample_rate_44100, (ma_uint32)ma_standard_sample_rate_32000, (ma_uint32)ma_standard_sample_rate_24000, (ma_uint32)ma_standard_sample_rate_22050, (ma_uint32)ma_standard_sample_rate_88200, (ma_uint32)ma_standard_sample_rate_96000, (ma_uint32)ma_standard_sample_rate_176400, (ma_uint32)ma_standard_sample_rate_192000, (ma_uint32)ma_standard_sample_rate_16000, (ma_uint32)ma_standard_sample_rate_11025, (ma_uint32)ma_standard_sample_rate_8000, (ma_uint32)ma_standard_sample_rate_352800, (ma_uint32)ma_standard_sample_rate_384000 }; static MA_INLINE ma_bool32 ma_is_standard_sample_rate(ma_uint32 sampleRate) { ma_uint32 iSampleRate; for (iSampleRate = 0; iSampleRate < sizeof(g_maStandardSampleRatePriorities) / sizeof(g_maStandardSampleRatePriorities[0]); iSampleRate += 1) { if (g_maStandardSampleRatePriorities[iSampleRate] == sampleRate) { return MA_TRUE; } } /* Getting here means the sample rate is not supported. */ return MA_FALSE; } static ma_format g_maFormatPriorities[] = { ma_format_s16, /* Most common */ ma_format_f32, /*ma_format_s24_32,*/ /* Clean alignment */ ma_format_s32, ma_format_s24, /* Unclean alignment */ ma_format_u8 /* Low quality */ }; #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) #pragma GCC diagnostic pop #endif MA_API void ma_version(ma_uint32* pMajor, ma_uint32* pMinor, ma_uint32* pRevision) { if (pMajor) { *pMajor = MA_VERSION_MAJOR; } if (pMinor) { *pMinor = MA_VERSION_MINOR; } if (pRevision) { *pRevision = MA_VERSION_REVISION; } } MA_API const char* ma_version_string(void) { return MA_VERSION_STRING; } /****************************************************************************** Standard Library Stuff ******************************************************************************/ #ifndef MA_ASSERT #define MA_ASSERT(condition) assert(condition) #endif #ifndef MA_MALLOC #define MA_MALLOC(sz) malloc((sz)) #endif #ifndef MA_REALLOC #define MA_REALLOC(p, sz) realloc((p), (sz)) #endif #ifndef MA_FREE #define MA_FREE(p) free((p)) #endif static MA_INLINE void ma_zero_memory_default(void* p, size_t sz) { if (p == NULL) { MA_ASSERT(sz == 0); /* If this is triggered there's an error with the calling code. */ return; } if (sz > 0) { memset(p, 0, sz); } } #ifndef MA_ZERO_MEMORY #define MA_ZERO_MEMORY(p, sz) ma_zero_memory_default((p), (sz)) #endif #ifndef MA_COPY_MEMORY #define MA_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) #endif #ifndef MA_MOVE_MEMORY #define MA_MOVE_MEMORY(dst, src, sz) memmove((dst), (src), (sz)) #endif #define MA_ZERO_OBJECT(p) MA_ZERO_MEMORY((p), sizeof(*(p))) #define ma_countof(x) (sizeof(x) / sizeof(x[0])) #define ma_max(x, y) (((x) > (y)) ? (x) : (y)) #define ma_min(x, y) (((x) < (y)) ? (x) : (y)) #define ma_abs(x) (((x) > 0) ? (x) : -(x)) #define ma_clamp(x, lo, hi) (ma_max(lo, ma_min(x, hi))) #define ma_offset_ptr(p, offset) (((ma_uint8*)(p)) + (offset)) #define ma_align(x, a) (((x) + ((a)-1)) & ~((a)-1)) #define ma_align_64(x) ma_align(x, 8) #define ma_buffer_frame_capacity(buffer, channels, format) (sizeof(buffer) / ma_get_bytes_per_sample(format) / (channels)) static MA_INLINE double ma_sind(double x) { /* TODO: Implement custom sin(x). */ return sin(x); } static MA_INLINE double ma_expd(double x) { /* TODO: Implement custom exp(x). */ return exp(x); } static MA_INLINE double ma_logd(double x) { /* TODO: Implement custom log(x). */ return log(x); } static MA_INLINE double ma_powd(double x, double y) { /* TODO: Implement custom pow(x, y). */ return pow(x, y); } static MA_INLINE double ma_sqrtd(double x) { /* TODO: Implement custom sqrt(x). */ return sqrt(x); } static MA_INLINE float ma_rsqrtf(float x) { #if defined(MA_SUPPORT_SSE2) && !defined(MA_NO_SSE2) && (defined(MA_X64) || (defined(_M_IX86_FP) && _M_IX86_FP == 2) || defined(__SSE2__)) { /* For SSE we can use RSQRTSS. This Stack Overflow post suggests that compilers don't necessarily generate optimal code when using intrinsics: https://web.archive.org/web/20221211012522/https://stackoverflow.com/questions/32687079/getting-fewest-instructions-for-rsqrtss-wrapper I'm going to do something similar here, but a bit simpler. */ #if defined(__GNUC__) || defined(__clang__) { float result; __asm__ __volatile__("rsqrtss %1, %0" : "=x"(result) : "x"(x)); return result; } #else { return _mm_cvtss_f32(_mm_rsqrt_ss(_mm_set_ps1(x))); } #endif } #else { return 1 / (float)ma_sqrtd(x); } #endif } static MA_INLINE float ma_sinf(float x) { return (float)ma_sind((float)x); } static MA_INLINE double ma_cosd(double x) { return ma_sind((MA_PI_D*0.5) - x); } static MA_INLINE float ma_cosf(float x) { return (float)ma_cosd((float)x); } static MA_INLINE double ma_log10d(double x) { return ma_logd(x) * 0.43429448190325182765; } static MA_INLINE float ma_powf(float x, float y) { return (float)ma_powd((double)x, (double)y); } static MA_INLINE float ma_log10f(float x) { return (float)ma_log10d((double)x); } static MA_INLINE double ma_degrees_to_radians(double degrees) { return degrees * 0.01745329252; } static MA_INLINE double ma_radians_to_degrees(double radians) { return radians * 57.295779512896; } static MA_INLINE float ma_degrees_to_radians_f(float degrees) { return degrees * 0.01745329252f; } static MA_INLINE float ma_radians_to_degrees_f(float radians) { return radians * 57.295779512896f; } /* Return Values: 0: Success 22: EINVAL 34: ERANGE Not using symbolic constants for errors because I want to avoid #including errno.h These are marked as no-inline because of some bad code generation by Clang. None of these functions are used in any performance-critical code within miniaudio. */ MA_API MA_NO_INLINE int ma_strcpy_s(char* dst, size_t dstSizeInBytes, const char* src) { size_t i; if (dst == 0) { return 22; } if (dstSizeInBytes == 0) { return 34; } if (src == 0) { dst[0] = '\0'; return 22; } for (i = 0; i < dstSizeInBytes && src[i] != '\0'; ++i) { dst[i] = src[i]; } if (i < dstSizeInBytes) { dst[i] = '\0'; return 0; } dst[0] = '\0'; return 34; } MA_API MA_NO_INLINE int ma_wcscpy_s(wchar_t* dst, size_t dstCap, const wchar_t* src) { size_t i; if (dst == 0) { return 22; } if (dstCap == 0) { return 34; } if (src == 0) { dst[0] = '\0'; return 22; } for (i = 0; i < dstCap && src[i] != '\0'; ++i) { dst[i] = src[i]; } if (i < dstCap) { dst[i] = '\0'; return 0; } dst[0] = '\0'; return 34; } MA_API MA_NO_INLINE int ma_strncpy_s(char* dst, size_t dstSizeInBytes, const char* src, size_t count) { size_t maxcount; size_t i; if (dst == 0) { return 22; } if (dstSizeInBytes == 0) { return 34; } if (src == 0) { dst[0] = '\0'; return 22; } maxcount = count; if (count == ((size_t)-1) || count >= dstSizeInBytes) { /* -1 = _TRUNCATE */ maxcount = dstSizeInBytes - 1; } for (i = 0; i < maxcount && src[i] != '\0'; ++i) { dst[i] = src[i]; } if (src[i] == '\0' || i == count || count == ((size_t)-1)) { dst[i] = '\0'; return 0; } dst[0] = '\0'; return 34; } MA_API MA_NO_INLINE int ma_strcat_s(char* dst, size_t dstSizeInBytes, const char* src) { char* dstorig; if (dst == 0) { return 22; } if (dstSizeInBytes == 0) { return 34; } if (src == 0) { dst[0] = '\0'; return 22; } dstorig = dst; while (dstSizeInBytes > 0 && dst[0] != '\0') { dst += 1; dstSizeInBytes -= 1; } if (dstSizeInBytes == 0) { return 22; /* Unterminated. */ } while (dstSizeInBytes > 0 && src[0] != '\0') { *dst++ = *src++; dstSizeInBytes -= 1; } if (dstSizeInBytes > 0) { dst[0] = '\0'; } else { dstorig[0] = '\0'; return 34; } return 0; } MA_API MA_NO_INLINE int ma_strncat_s(char* dst, size_t dstSizeInBytes, const char* src, size_t count) { char* dstorig; if (dst == 0) { return 22; } if (dstSizeInBytes == 0) { return 34; } if (src == 0) { return 22; } dstorig = dst; while (dstSizeInBytes > 0 && dst[0] != '\0') { dst += 1; dstSizeInBytes -= 1; } if (dstSizeInBytes == 0) { return 22; /* Unterminated. */ } if (count == ((size_t)-1)) { /* _TRUNCATE */ count = dstSizeInBytes - 1; } while (dstSizeInBytes > 0 && src[0] != '\0' && count > 0) { *dst++ = *src++; dstSizeInBytes -= 1; count -= 1; } if (dstSizeInBytes > 0) { dst[0] = '\0'; } else { dstorig[0] = '\0'; return 34; } return 0; } MA_API MA_NO_INLINE int ma_itoa_s(int value, char* dst, size_t dstSizeInBytes, int radix) { int sign; unsigned int valueU; char* dstEnd; if (dst == NULL || dstSizeInBytes == 0) { return 22; } if (radix < 2 || radix > 36) { dst[0] = '\0'; return 22; } sign = (value < 0 && radix == 10) ? -1 : 1; /* The negative sign is only used when the base is 10. */ if (value < 0) { valueU = -value; } else { valueU = value; } dstEnd = dst; do { int remainder = valueU % radix; if (remainder > 9) { *dstEnd = (char)((remainder - 10) + 'a'); } else { *dstEnd = (char)(remainder + '0'); } dstEnd += 1; dstSizeInBytes -= 1; valueU /= radix; } while (dstSizeInBytes > 0 && valueU > 0); if (dstSizeInBytes == 0) { dst[0] = '\0'; return 22; /* Ran out of room in the output buffer. */ } if (sign < 0) { *dstEnd++ = '-'; dstSizeInBytes -= 1; } if (dstSizeInBytes == 0) { dst[0] = '\0'; return 22; /* Ran out of room in the output buffer. */ } *dstEnd = '\0'; /* At this point the string will be reversed. */ dstEnd -= 1; while (dst < dstEnd) { char temp = *dst; *dst = *dstEnd; *dstEnd = temp; dst += 1; dstEnd -= 1; } return 0; } MA_API MA_NO_INLINE int ma_strcmp(const char* str1, const char* str2) { if (str1 == str2) return 0; /* These checks differ from the standard implementation. It's not important, but I prefer it just for sanity. */ if (str1 == NULL) return -1; if (str2 == NULL) return 1; for (;;) { if (str1[0] == '\0') { break; } if (str1[0] != str2[0]) { break; } str1 += 1; str2 += 1; } return ((unsigned char*)str1)[0] - ((unsigned char*)str2)[0]; } MA_API MA_NO_INLINE int ma_wcscmp(const wchar_t* str1, const wchar_t* str2) { if (str1 == str2) return 0; /* These checks differ from the standard implementation. It's not important, but I prefer it just for sanity. */ if (str1 == NULL) return -1; if (str2 == NULL) return 1; for (;;) { if (str1[0] == L'\0') { break; } if (str1[0] != str2[0]) { break; } str1 += 1; str2 += 1; } return ((unsigned short*)str1)[0] - ((unsigned short*)str2)[0]; } MA_API MA_NO_INLINE int ma_strappend(char* dst, size_t dstSize, const char* srcA, const char* srcB) { int result; result = ma_strncpy_s(dst, dstSize, srcA, (size_t)-1); if (result != 0) { return result; } result = ma_strncat_s(dst, dstSize, srcB, (size_t)-1); if (result != 0) { return result; } return result; } MA_API MA_NO_INLINE size_t ma_wcslen(const wchar_t* str) { const wchar_t* end; if (str == NULL) { return 0; } end = str; while (end[0] != '\0') { end += 1; } return end - str; } MA_API MA_NO_INLINE char* ma_copy_string(const char* src, const ma_allocation_callbacks* pAllocationCallbacks) { size_t sz; char* dst; if (src == NULL) { return NULL; } sz = strlen(src)+1; dst = (char*)ma_malloc(sz, pAllocationCallbacks); if (dst == NULL) { return NULL; } ma_strcpy_s(dst, sz, src); return dst; } MA_API MA_NO_INLINE wchar_t* ma_copy_string_w(const wchar_t* src, const ma_allocation_callbacks* pAllocationCallbacks) { size_t sz = ma_wcslen(src)+1; wchar_t* dst = (wchar_t*)ma_malloc(sz * sizeof(*dst), pAllocationCallbacks); if (dst == NULL) { return NULL; } ma_wcscpy_s(dst, sz, src); return dst; } #include <errno.h> static ma_result ma_result_from_errno(int e) { if (e == 0) { return MA_SUCCESS; } #ifdef EPERM else if (e == EPERM) { return MA_INVALID_OPERATION; } #endif #ifdef ENOENT else if (e == ENOENT) { return MA_DOES_NOT_EXIST; } #endif #ifdef ESRCH else if (e == ESRCH) { return MA_DOES_NOT_EXIST; } #endif #ifdef EINTR else if (e == EINTR) { return MA_INTERRUPT; } #endif #ifdef EIO else if (e == EIO) { return MA_IO_ERROR; } #endif #ifdef ENXIO else if (e == ENXIO) { return MA_DOES_NOT_EXIST; } #endif #ifdef E2BIG else if (e == E2BIG) { return MA_INVALID_ARGS; } #endif #ifdef ENOEXEC else if (e == ENOEXEC) { return MA_INVALID_FILE; } #endif #ifdef EBADF else if (e == EBADF) { return MA_INVALID_FILE; } #endif #ifdef ECHILD else if (e == ECHILD) { return MA_ERROR; } #endif #ifdef EAGAIN else if (e == EAGAIN) { return MA_UNAVAILABLE; } #endif #ifdef ENOMEM else if (e == ENOMEM) { return MA_OUT_OF_MEMORY; } #endif #ifdef EACCES else if (e == EACCES) { return MA_ACCESS_DENIED; } #endif #ifdef EFAULT else if (e == EFAULT) { return MA_BAD_ADDRESS; } #endif #ifdef ENOTBLK else if (e == ENOTBLK) { return MA_ERROR; } #endif #ifdef EBUSY else if (e == EBUSY) { return MA_BUSY; } #endif #ifdef EEXIST else if (e == EEXIST) { return MA_ALREADY_EXISTS; } #endif #ifdef EXDEV else if (e == EXDEV) { return MA_ERROR; } #endif #ifdef ENODEV else if (e == ENODEV) { return MA_DOES_NOT_EXIST; } #endif #ifdef ENOTDIR else if (e == ENOTDIR) { return MA_NOT_DIRECTORY; } #endif #ifdef EISDIR else if (e == EISDIR) { return MA_IS_DIRECTORY; } #endif #ifdef EINVAL else if (e == EINVAL) { return MA_INVALID_ARGS; } #endif #ifdef ENFILE else if (e == ENFILE) { return MA_TOO_MANY_OPEN_FILES; } #endif #ifdef EMFILE else if (e == EMFILE) { return MA_TOO_MANY_OPEN_FILES; } #endif #ifdef ENOTTY else if (e == ENOTTY) { return MA_INVALID_OPERATION; } #endif #ifdef ETXTBSY else if (e == ETXTBSY) { return MA_BUSY; } #endif #ifdef EFBIG else if (e == EFBIG) { return MA_TOO_BIG; } #endif #ifdef ENOSPC else if (e == ENOSPC) { return MA_NO_SPACE; } #endif #ifdef ESPIPE else if (e == ESPIPE) { return MA_BAD_SEEK; } #endif #ifdef EROFS else if (e == EROFS) { return MA_ACCESS_DENIED; } #endif #ifdef EMLINK else if (e == EMLINK) { return MA_TOO_MANY_LINKS; } #endif #ifdef EPIPE else if (e == EPIPE) { return MA_BAD_PIPE; } #endif #ifdef EDOM else if (e == EDOM) { return MA_OUT_OF_RANGE; } #endif #ifdef ERANGE else if (e == ERANGE) { return MA_OUT_OF_RANGE; } #endif #ifdef EDEADLK else if (e == EDEADLK) { return MA_DEADLOCK; } #endif #ifdef ENAMETOOLONG else if (e == ENAMETOOLONG) { return MA_PATH_TOO_LONG; } #endif #ifdef ENOLCK else if (e == ENOLCK) { return MA_ERROR; } #endif #ifdef ENOSYS else if (e == ENOSYS) { return MA_NOT_IMPLEMENTED; } #endif #ifdef ENOTEMPTY else if (e == ENOTEMPTY) { return MA_DIRECTORY_NOT_EMPTY; } #endif #ifdef ELOOP else if (e == ELOOP) { return MA_TOO_MANY_LINKS; } #endif #ifdef ENOMSG else if (e == ENOMSG) { return MA_NO_MESSAGE; } #endif #ifdef EIDRM else if (e == EIDRM) { return MA_ERROR; } #endif #ifdef ECHRNG else if (e == ECHRNG) { return MA_ERROR; } #endif #ifdef EL2NSYNC else if (e == EL2NSYNC) { return MA_ERROR; } #endif #ifdef EL3HLT else if (e == EL3HLT) { return MA_ERROR; } #endif #ifdef EL3RST else if (e == EL3RST) { return MA_ERROR; } #endif #ifdef ELNRNG else if (e == ELNRNG) { return MA_OUT_OF_RANGE; } #endif #ifdef EUNATCH else if (e == EUNATCH) { return MA_ERROR; } #endif #ifdef ENOCSI else if (e == ENOCSI) { return MA_ERROR; } #endif #ifdef EL2HLT else if (e == EL2HLT) { return MA_ERROR; } #endif #ifdef EBADE else if (e == EBADE) { return MA_ERROR; } #endif #ifdef EBADR else if (e == EBADR) { return MA_ERROR; } #endif #ifdef EXFULL else if (e == EXFULL) { return MA_ERROR; } #endif #ifdef ENOANO else if (e == ENOANO) { return MA_ERROR; } #endif #ifdef EBADRQC else if (e == EBADRQC) { return MA_ERROR; } #endif #ifdef EBADSLT else if (e == EBADSLT) { return MA_ERROR; } #endif #ifdef EBFONT else if (e == EBFONT) { return MA_INVALID_FILE; } #endif #ifdef ENOSTR else if (e == ENOSTR) { return MA_ERROR; } #endif #ifdef ENODATA else if (e == ENODATA) { return MA_NO_DATA_AVAILABLE; } #endif #ifdef ETIME else if (e == ETIME) { return MA_TIMEOUT; } #endif #ifdef ENOSR else if (e == ENOSR) { return MA_NO_DATA_AVAILABLE; } #endif #ifdef ENONET else if (e == ENONET) { return MA_NO_NETWORK; } #endif #ifdef ENOPKG else if (e == ENOPKG) { return MA_ERROR; } #endif #ifdef EREMOTE else if (e == EREMOTE) { return MA_ERROR; } #endif #ifdef ENOLINK else if (e == ENOLINK) { return MA_ERROR; } #endif #ifdef EADV else if (e == EADV) { return MA_ERROR; } #endif #ifdef ESRMNT else if (e == ESRMNT) { return MA_ERROR; } #endif #ifdef ECOMM else if (e == ECOMM) { return MA_ERROR; } #endif #ifdef EPROTO else if (e == EPROTO) { return MA_ERROR; } #endif #ifdef EMULTIHOP else if (e == EMULTIHOP) { return MA_ERROR; } #endif #ifdef EDOTDOT else if (e == EDOTDOT) { return MA_ERROR; } #endif #ifdef EBADMSG else if (e == EBADMSG) { return MA_BAD_MESSAGE; } #endif #ifdef EOVERFLOW else if (e == EOVERFLOW) { return MA_TOO_BIG; } #endif #ifdef ENOTUNIQ else if (e == ENOTUNIQ) { return MA_NOT_UNIQUE; } #endif #ifdef EBADFD else if (e == EBADFD) { return MA_ERROR; } #endif #ifdef EREMCHG else if (e == EREMCHG) { return MA_ERROR; } #endif #ifdef ELIBACC else if (e == ELIBACC) { return MA_ACCESS_DENIED; } #endif #ifdef ELIBBAD else if (e == ELIBBAD) { return MA_INVALID_FILE; } #endif #ifdef ELIBSCN else if (e == ELIBSCN) { return MA_INVALID_FILE; } #endif #ifdef ELIBMAX else if (e == ELIBMAX) { return MA_ERROR; } #endif #ifdef ELIBEXEC else if (e == ELIBEXEC) { return MA_ERROR; } #endif #ifdef EILSEQ else if (e == EILSEQ) { return MA_INVALID_DATA; } #endif #ifdef ERESTART else if (e == ERESTART) { return MA_ERROR; } #endif #ifdef ESTRPIPE else if (e == ESTRPIPE) { return MA_ERROR; } #endif #ifdef EUSERS else if (e == EUSERS) { return MA_ERROR; } #endif #ifdef ENOTSOCK else if (e == ENOTSOCK) { return MA_NOT_SOCKET; } #endif #ifdef EDESTADDRREQ else if (e == EDESTADDRREQ) { return MA_NO_ADDRESS; } #endif #ifdef EMSGSIZE else if (e == EMSGSIZE) { return MA_TOO_BIG; } #endif #ifdef EPROTOTYPE else if (e == EPROTOTYPE) { return MA_BAD_PROTOCOL; } #endif #ifdef ENOPROTOOPT else if (e == ENOPROTOOPT) { return MA_PROTOCOL_UNAVAILABLE; } #endif #ifdef EPROTONOSUPPORT else if (e == EPROTONOSUPPORT) { return MA_PROTOCOL_NOT_SUPPORTED; } #endif #ifdef ESOCKTNOSUPPORT else if (e == ESOCKTNOSUPPORT) { return MA_SOCKET_NOT_SUPPORTED; } #endif #ifdef EOPNOTSUPP else if (e == EOPNOTSUPP) { return MA_INVALID_OPERATION; } #endif #ifdef EPFNOSUPPORT else if (e == EPFNOSUPPORT) { return MA_PROTOCOL_FAMILY_NOT_SUPPORTED; } #endif #ifdef EAFNOSUPPORT else if (e == EAFNOSUPPORT) { return MA_ADDRESS_FAMILY_NOT_SUPPORTED; } #endif #ifdef EADDRINUSE else if (e == EADDRINUSE) { return MA_ALREADY_IN_USE; } #endif #ifdef EADDRNOTAVAIL else if (e == EADDRNOTAVAIL) { return MA_ERROR; } #endif #ifdef ENETDOWN else if (e == ENETDOWN) { return MA_NO_NETWORK; } #endif #ifdef ENETUNREACH else if (e == ENETUNREACH) { return MA_NO_NETWORK; } #endif #ifdef ENETRESET else if (e == ENETRESET) { return MA_NO_NETWORK; } #endif #ifdef ECONNABORTED else if (e == ECONNABORTED) { return MA_NO_NETWORK; } #endif #ifdef ECONNRESET else if (e == ECONNRESET) { return MA_CONNECTION_RESET; } #endif #ifdef ENOBUFS else if (e == ENOBUFS) { return MA_NO_SPACE; } #endif #ifdef EISCONN else if (e == EISCONN) { return MA_ALREADY_CONNECTED; } #endif #ifdef ENOTCONN else if (e == ENOTCONN) { return MA_NOT_CONNECTED; } #endif #ifdef ESHUTDOWN else if (e == ESHUTDOWN) { return MA_ERROR; } #endif #ifdef ETOOMANYREFS else if (e == ETOOMANYREFS) { return MA_ERROR; } #endif #ifdef ETIMEDOUT else if (e == ETIMEDOUT) { return MA_TIMEOUT; } #endif #ifdef ECONNREFUSED else if (e == ECONNREFUSED) { return MA_CONNECTION_REFUSED; } #endif #ifdef EHOSTDOWN else if (e == EHOSTDOWN) { return MA_NO_HOST; } #endif #ifdef EHOSTUNREACH else if (e == EHOSTUNREACH) { return MA_NO_HOST; } #endif #ifdef EALREADY else if (e == EALREADY) { return MA_IN_PROGRESS; } #endif #ifdef EINPROGRESS else if (e == EINPROGRESS) { return MA_IN_PROGRESS; } #endif #ifdef ESTALE else if (e == ESTALE) { return MA_INVALID_FILE; } #endif #ifdef EUCLEAN else if (e == EUCLEAN) { return MA_ERROR; } #endif #ifdef ENOTNAM else if (e == ENOTNAM) { return MA_ERROR; } #endif #ifdef ENAVAIL else if (e == ENAVAIL) { return MA_ERROR; } #endif #ifdef EISNAM else if (e == EISNAM) { return MA_ERROR; } #endif #ifdef EREMOTEIO else if (e == EREMOTEIO) { return MA_IO_ERROR; } #endif #ifdef EDQUOT else if (e == EDQUOT) { return MA_NO_SPACE; } #endif #ifdef ENOMEDIUM else if (e == ENOMEDIUM) { return MA_DOES_NOT_EXIST; } #endif #ifdef EMEDIUMTYPE else if (e == EMEDIUMTYPE) { return MA_ERROR; } #endif #ifdef ECANCELED else if (e == ECANCELED) { return MA_CANCELLED; } #endif #ifdef ENOKEY else if (e == ENOKEY) { return MA_ERROR; } #endif #ifdef EKEYEXPIRED else if (e == EKEYEXPIRED) { return MA_ERROR; } #endif #ifdef EKEYREVOKED else if (e == EKEYREVOKED) { return MA_ERROR; } #endif #ifdef EKEYREJECTED else if (e == EKEYREJECTED) { return MA_ERROR; } #endif #ifdef EOWNERDEAD else if (e == EOWNERDEAD) { return MA_ERROR; } #endif #ifdef ENOTRECOVERABLE else if (e == ENOTRECOVERABLE) { return MA_ERROR; } #endif #ifdef ERFKILL else if (e == ERFKILL) { return MA_ERROR; } #endif #ifdef EHWPOISON else if (e == EHWPOISON) { return MA_ERROR; } #endif else { return MA_ERROR; } } MA_API ma_result ma_fopen(FILE** ppFile, const char* pFilePath, const char* pOpenMode) { #if defined(_MSC_VER) && _MSC_VER >= 1400 errno_t err; #endif if (ppFile != NULL) { *ppFile = NULL; /* Safety. */ } if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { return MA_INVALID_ARGS; } #if (defined(_MSC_VER) && _MSC_VER >= 1400) && !defined(MA_XBOX_NXDK) err = fopen_s(ppFile, pFilePath, pOpenMode); if (err != 0) { return ma_result_from_errno(err); } #else #if defined(_WIN32) || defined(__APPLE__) *ppFile = fopen(pFilePath, pOpenMode); #else #if defined(_FILE_OFFSET_BITS) && _FILE_OFFSET_BITS == 64 && defined(_LARGEFILE64_SOURCE) *ppFile = fopen64(pFilePath, pOpenMode); #else *ppFile = fopen(pFilePath, pOpenMode); #endif #endif if (*ppFile == NULL) { ma_result result = ma_result_from_errno(errno); if (result == MA_SUCCESS) { result = MA_ERROR; /* Just a safety check to make sure we never ever return success when pFile == NULL. */ } return result; } #endif return MA_SUCCESS; } /* _wfopen() isn't always available in all compilation environments. * Windows only. * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back). * MinGW-64 (both 32- and 64-bit) seems to support it. * MinGW wraps it in !defined(__STRICT_ANSI__). * OpenWatcom wraps it in !defined(_NO_EXT_KEYS). This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs() fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support. */ #if defined(_WIN32) && !defined(MA_XBOX_NXDK) #if defined(_MSC_VER) || defined(__MINGW64__) || (!defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS)) #define MA_HAS_WFOPEN #endif #endif MA_API ma_result ma_wfopen(FILE** ppFile, const wchar_t* pFilePath, const wchar_t* pOpenMode, const ma_allocation_callbacks* pAllocationCallbacks) { if (ppFile != NULL) { *ppFile = NULL; /* Safety. */ } if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { return MA_INVALID_ARGS; } #if defined(MA_HAS_WFOPEN) { /* Use _wfopen() on Windows. */ #if defined(_MSC_VER) && _MSC_VER >= 1400 { errno_t err = _wfopen_s(ppFile, pFilePath, pOpenMode); if (err != 0) { return ma_result_from_errno(err); } } #else { *ppFile = _wfopen(pFilePath, pOpenMode); if (*ppFile == NULL) { return ma_result_from_errno(errno); } } #endif (void)pAllocationCallbacks; } #elif !defined(MA_XBOX_NXDK) && !defined(MA_DOS) /* If your compiler does not support wcsrtombs(), add it here. */ { /* Use fopen() on anything other than Windows. Requires a conversion. This is annoying because fopen() is locale specific. The only real way I can think of to do this is with wcsrtombs(). Note that wcstombs() is apparently not thread-safe because it uses a static global mbstate_t object for maintaining state. I've checked this with -std=c89 and it works, but if somebody get's a compiler error I'll look into improving compatibility. */ mbstate_t mbs; size_t lenMB; const wchar_t* pFilePathTemp = pFilePath; char* pFilePathMB = NULL; char pOpenModeMB[32] = {0}; /* Get the length first. */ MA_ZERO_OBJECT(&mbs); lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs); if (lenMB == (size_t)-1) { return ma_result_from_errno(errno); } pFilePathMB = (char*)ma_malloc(lenMB + 1, pAllocationCallbacks); if (pFilePathMB == NULL) { return MA_OUT_OF_MEMORY; } pFilePathTemp = pFilePath; MA_ZERO_OBJECT(&mbs); wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs); /* The open mode should always consist of ASCII characters so we should be able to do a trivial conversion. */ { size_t i = 0; for (;;) { if (pOpenMode[i] == 0) { pOpenModeMB[i] = '\0'; break; } pOpenModeMB[i] = (char)pOpenMode[i]; i += 1; } } *ppFile = fopen(pFilePathMB, pOpenModeMB); ma_free(pFilePathMB, pAllocationCallbacks); } #else { /* Getting here means there is no way to open the file with a wide character string. */ *ppFile = NULL; } #endif if (*ppFile == NULL) { return MA_ERROR; } return MA_SUCCESS; } static MA_INLINE void ma_copy_memory_64(void* dst, const void* src, ma_uint64 sizeInBytes) { #if MA_SIZE_MAX > 0xFFFFFFFF MA_COPY_MEMORY(dst, src, (size_t)sizeInBytes); #else while (sizeInBytes > 0) { ma_uint64 bytesToCopyNow = sizeInBytes; if (bytesToCopyNow > MA_SIZE_MAX) { bytesToCopyNow = MA_SIZE_MAX; } MA_COPY_MEMORY(dst, src, (size_t)bytesToCopyNow); /* Safe cast to size_t. */ sizeInBytes -= bytesToCopyNow; dst = ( void*)(( ma_uint8*)dst + bytesToCopyNow); src = (const void*)((const ma_uint8*)src + bytesToCopyNow); } #endif } static MA_INLINE void ma_zero_memory_64(void* dst, ma_uint64 sizeInBytes) { #if MA_SIZE_MAX > 0xFFFFFFFF MA_ZERO_MEMORY(dst, (size_t)sizeInBytes); #else while (sizeInBytes > 0) { ma_uint64 bytesToZeroNow = sizeInBytes; if (bytesToZeroNow > MA_SIZE_MAX) { bytesToZeroNow = MA_SIZE_MAX; } MA_ZERO_MEMORY(dst, (size_t)bytesToZeroNow); /* Safe cast to size_t. */ sizeInBytes -= bytesToZeroNow; dst = (void*)((ma_uint8*)dst + bytesToZeroNow); } #endif } /* Thanks to good old Bit Twiddling Hacks for this one: http://graphics.stanford.edu/~seander/bithacks.html#RoundUpPowerOf2 */ static MA_INLINE unsigned int ma_next_power_of_2(unsigned int x) { x--; x |= x >> 1; x |= x >> 2; x |= x >> 4; x |= x >> 8; x |= x >> 16; x++; return x; } static MA_INLINE unsigned int ma_prev_power_of_2(unsigned int x) { return ma_next_power_of_2(x) >> 1; } static MA_INLINE unsigned int ma_round_to_power_of_2(unsigned int x) { unsigned int prev = ma_prev_power_of_2(x); unsigned int next = ma_next_power_of_2(x); if ((next - x) > (x - prev)) { return prev; } else { return next; } } static MA_INLINE unsigned int ma_count_set_bits(unsigned int x) { unsigned int count = 0; while (x != 0) { if (x & 1) { count += 1; } x = x >> 1; } return count; } /************************************************************************************************************************************************************** Allocation Callbacks **************************************************************************************************************************************************************/ static void* ma__malloc_default(size_t sz, void* pUserData) { (void)pUserData; return MA_MALLOC(sz); } static void* ma__realloc_default(void* p, size_t sz, void* pUserData) { (void)pUserData; return MA_REALLOC(p, sz); } static void ma__free_default(void* p, void* pUserData) { (void)pUserData; MA_FREE(p); } static ma_allocation_callbacks ma_allocation_callbacks_init_default(void) { ma_allocation_callbacks callbacks; callbacks.pUserData = NULL; callbacks.onMalloc = ma__malloc_default; callbacks.onRealloc = ma__realloc_default; callbacks.onFree = ma__free_default; return callbacks; } static ma_result ma_allocation_callbacks_init_copy(ma_allocation_callbacks* pDst, const ma_allocation_callbacks* pSrc) { if (pDst == NULL) { return MA_INVALID_ARGS; } if (pSrc == NULL) { *pDst = ma_allocation_callbacks_init_default(); } else { if (pSrc->pUserData == NULL && pSrc->onFree == NULL && pSrc->onMalloc == NULL && pSrc->onRealloc == NULL) { *pDst = ma_allocation_callbacks_init_default(); } else { if (pSrc->onFree == NULL || (pSrc->onMalloc == NULL && pSrc->onRealloc == NULL)) { return MA_INVALID_ARGS; /* Invalid allocation callbacks. */ } else { *pDst = *pSrc; } } } return MA_SUCCESS; } /************************************************************************************************************************************************************** Logging **************************************************************************************************************************************************************/ #ifndef ma_va_copy #if !defined(_MSC_VER) || _MSC_VER >= 1800 #if (defined(__GNUC__) && __GNUC__ < 3) #define ma_va_copy(dst, src) ((dst) = (src)) /* This is untested. Not sure if this is correct for old GCC. */ #else #define ma_va_copy(dst, src) va_copy((dst), (src)) #endif #else #define ma_va_copy(dst, src) ((dst) = (src)) #endif #endif MA_API const char* ma_log_level_to_string(ma_uint32 logLevel) { switch (logLevel) { case MA_LOG_LEVEL_DEBUG: return "DEBUG"; case MA_LOG_LEVEL_INFO: return "INFO"; case MA_LOG_LEVEL_WARNING: return "WARNING"; case MA_LOG_LEVEL_ERROR: return "ERROR"; default: return "ERROR"; } } #if defined(MA_DEBUG_OUTPUT) #if defined(MA_ANDROID) #include <android/log.h> #endif /* Customize this to use a specific tag in __android_log_print() for debug output messages. */ #ifndef MA_ANDROID_LOG_TAG #define MA_ANDROID_LOG_TAG "miniaudio" #endif void ma_log_callback_debug(void* pUserData, ma_uint32 level, const char* pMessage) { (void)pUserData; /* Special handling for some platforms. */ #if defined(MA_ANDROID) { /* Android. */ __android_log_print(ANDROID_LOG_DEBUG, MA_ANDROID_LOG_TAG, "%s: %s", ma_log_level_to_string(level), pMessage); } #else { /* Everything else. */ printf("%s: %s", ma_log_level_to_string(level), pMessage); } #endif } #endif MA_API ma_log_callback ma_log_callback_init(ma_log_callback_proc onLog, void* pUserData) { ma_log_callback callback; MA_ZERO_OBJECT(&callback); callback.onLog = onLog; callback.pUserData = pUserData; return callback; } MA_API ma_result ma_log_init(const ma_allocation_callbacks* pAllocationCallbacks, ma_log* pLog) { if (pLog == NULL) { return MA_INVALID_ARGS; } MA_ZERO_OBJECT(pLog); ma_allocation_callbacks_init_copy(&pLog->allocationCallbacks, pAllocationCallbacks); /* We need a mutex for thread safety. */ #ifndef MA_NO_THREADING { ma_result result = ma_mutex_init(&pLog->lock); if (result != MA_SUCCESS) { return result; } } #endif /* If we're using debug output, enable it. */ #if defined(MA_DEBUG_OUTPUT) { ma_log_register_callback(pLog, ma_log_callback_init(ma_log_callback_debug, NULL)); /* Doesn't really matter if this fails. */ } #endif return MA_SUCCESS; } MA_API void ma_log_uninit(ma_log* pLog) { if (pLog == NULL) { return; } #ifndef MA_NO_THREADING ma_mutex_uninit(&pLog->lock); #endif } static void ma_log_lock(ma_log* pLog) { #ifndef MA_NO_THREADING ma_mutex_lock(&pLog->lock); #else (void)pLog; #endif } static void ma_log_unlock(ma_log* pLog) { #ifndef MA_NO_THREADING ma_mutex_unlock(&pLog->lock); #else (void)pLog; #endif } MA_API ma_result ma_log_register_callback(ma_log* pLog, ma_log_callback callback) { ma_result result = MA_SUCCESS; if (pLog == NULL || callback.onLog == NULL) { return MA_INVALID_ARGS; } ma_log_lock(pLog); { if (pLog->callbackCount == ma_countof(pLog->callbacks)) { result = MA_OUT_OF_MEMORY; /* Reached the maximum allowed log callbacks. */ } else { pLog->callbacks[pLog->callbackCount] = callback; pLog->callbackCount += 1; } } ma_log_unlock(pLog); return result; } MA_API ma_result ma_log_unregister_callback(ma_log* pLog, ma_log_callback callback) { if (pLog == NULL) { return MA_INVALID_ARGS; } ma_log_lock(pLog); { ma_uint32 iLog; for (iLog = 0; iLog < pLog->callbackCount; ) { if (pLog->callbacks[iLog].onLog == callback.onLog) { /* Found. Move everything down a slot. */ ma_uint32 jLog; for (jLog = iLog; jLog < pLog->callbackCount-1; jLog += 1) { pLog->callbacks[jLog] = pLog->callbacks[jLog + 1]; } pLog->callbackCount -= 1; } else { /* Not found. */ iLog += 1; } } } ma_log_unlock(pLog); return MA_SUCCESS; } MA_API ma_result ma_log_post(ma_log* pLog, ma_uint32 level, const char* pMessage) { if (pLog == NULL || pMessage == NULL) { return MA_INVALID_ARGS; } ma_log_lock(pLog); { ma_uint32 iLog; for (iLog = 0; iLog < pLog->callbackCount; iLog += 1) { if (pLog->callbacks[iLog].onLog) { pLog->callbacks[iLog].onLog(pLog->callbacks[iLog].pUserData, level, pMessage); } } } ma_log_unlock(pLog); return MA_SUCCESS; } /* We need to emulate _vscprintf() for the VC6 build. This can be more efficient, but since it's only VC6, and it's just a logging function, I'm happy to keep this simple. In the VC6 build we can implement this in terms of _vsnprintf(). */ #if defined(_MSC_VER) && _MSC_VER < 1900 static int ma_vscprintf(const ma_allocation_callbacks* pAllocationCallbacks, const char* format, va_list args) { #if _MSC_VER > 1200 return _vscprintf(format, args); #else int result; char* pTempBuffer = NULL; size_t tempBufferCap = 1024; if (format == NULL) { errno = EINVAL; return -1; } for (;;) { char* pNewTempBuffer = (char*)ma_realloc(pTempBuffer, tempBufferCap, pAllocationCallbacks); if (pNewTempBuffer == NULL) { ma_free(pTempBuffer, pAllocationCallbacks); errno = ENOMEM; return -1; /* Out of memory. */ } pTempBuffer = pNewTempBuffer; result = _vsnprintf(pTempBuffer, tempBufferCap, format, args); ma_free(pTempBuffer, NULL); if (result != -1) { break; /* Got it. */ } /* Buffer wasn't big enough. Ideally it'd be nice to use an error code to know the reason for sure, but this is reliable enough. */ tempBufferCap *= 2; } return result; #endif } #endif MA_API ma_result ma_log_postv(ma_log* pLog, ma_uint32 level, const char* pFormat, va_list args) { if (pLog == NULL || pFormat == NULL) { return MA_INVALID_ARGS; } #if (defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || ((!defined(_MSC_VER) || _MSC_VER >= 1900) && !defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS)) || (defined(__cplusplus) && __cplusplus >= 201103L) { ma_result result; int length; char pFormattedMessageStack[1024]; char* pFormattedMessageHeap = NULL; va_list args2; /* First try formatting into our fixed sized stack allocated buffer. If this is too small we'll fallback to a heap allocation. */ ma_va_copy(args2, args); { length = vsnprintf(pFormattedMessageStack, sizeof(pFormattedMessageStack), pFormat, args2); } va_end(args2); if (length < 0) { return MA_INVALID_OPERATION; /* An error occurred when trying to convert the buffer. */ } if ((size_t)length < sizeof(pFormattedMessageStack)) { /* The string was written to the stack. */ result = ma_log_post(pLog, level, pFormattedMessageStack); } else { /* The stack buffer was too small, try the heap. */ pFormattedMessageHeap = (char*)ma_malloc(length + 1, &pLog->allocationCallbacks); if (pFormattedMessageHeap == NULL) { return MA_OUT_OF_MEMORY; } length = vsnprintf(pFormattedMessageHeap, length + 1, pFormat, args); if (length < 0) { ma_free(pFormattedMessageHeap, &pLog->allocationCallbacks); return MA_INVALID_OPERATION; } result = ma_log_post(pLog, level, pFormattedMessageHeap); ma_free(pFormattedMessageHeap, &pLog->allocationCallbacks); } return result; } #else { /* Without snprintf() we need to first measure the string and then heap allocate it. I'm only aware of Visual Studio having support for this without snprintf(), so we'll need to restrict this branch to Visual Studio. For other compilers we need to just not support formatted logging because I don't want the security risk of overflowing a fixed sized stack allocated buffer. */ #if defined(_MSC_VER) && _MSC_VER >= 1200 /* 1200 = VC6 */ { ma_result result; int formattedLen; char* pFormattedMessage = NULL; va_list args2; ma_va_copy(args2, args); { formattedLen = ma_vscprintf(&pLog->allocationCallbacks, pFormat, args2); } va_end(args2); if (formattedLen <= 0) { return MA_INVALID_OPERATION; } pFormattedMessage = (char*)ma_malloc(formattedLen + 1, &pLog->allocationCallbacks); if (pFormattedMessage == NULL) { return MA_OUT_OF_MEMORY; } /* We'll get errors on newer versions of Visual Studio if we try to use vsprintf(). */ #if _MSC_VER >= 1400 /* 1400 = Visual Studio 2005 */ { vsprintf_s(pFormattedMessage, formattedLen + 1, pFormat, args); } #else { vsprintf(pFormattedMessage, pFormat, args); } #endif result = ma_log_post(pLog, level, pFormattedMessage); ma_free(pFormattedMessage, &pLog->allocationCallbacks); return result; } #else { /* Can't do anything because we don't have a safe way of to emulate vsnprintf() without a manual solution. */ (void)level; (void)args; return MA_INVALID_OPERATION; } #endif } #endif } MA_API ma_result ma_log_postf(ma_log* pLog, ma_uint32 level, const char* pFormat, ...) { ma_result result; va_list args; if (pLog == NULL || pFormat == NULL) { return MA_INVALID_ARGS; } va_start(args, pFormat); { result = ma_log_postv(pLog, level, pFormat, args); } va_end(args); return result; } static MA_INLINE ma_uint8 ma_clip_u8(ma_int32 x) { return (ma_uint8)(ma_clamp(x, -128, 127) + 128); } static MA_INLINE ma_int16 ma_clip_s16(ma_int32 x) { return (ma_int16)ma_clamp(x, -32768, 32767); } static MA_INLINE ma_int64 ma_clip_s24(ma_int64 x) { return (ma_int64)ma_clamp(x, -8388608, 8388607); } static MA_INLINE ma_int32 ma_clip_s32(ma_int64 x) { /* This dance is to silence warnings with -std=c89. A good compiler should be able to optimize this away. */ ma_int64 clipMin; ma_int64 clipMax; clipMin = -((ma_int64)2147483647 + 1); clipMax = (ma_int64)2147483647; return (ma_int32)ma_clamp(x, clipMin, clipMax); } static MA_INLINE float ma_clip_f32(float x) { if (x < -1) return -1; if (x > +1) return +1; return x; } static MA_INLINE float ma_mix_f32(float x, float y, float a) { return x*(1-a) + y*a; } static MA_INLINE float ma_mix_f32_fast(float x, float y, float a) { float r0 = (y - x); float r1 = r0*a; return x + r1; /*return x + (y - x)*a;*/ } #if defined(MA_SUPPORT_SSE2) static MA_INLINE __m128 ma_mix_f32_fast__sse2(__m128 x, __m128 y, __m128 a) { return _mm_add_ps(x, _mm_mul_ps(_mm_sub_ps(y, x), a)); } #endif #if defined(MA_SUPPORT_AVX2) static MA_INLINE __m256 ma_mix_f32_fast__avx2(__m256 x, __m256 y, __m256 a) { return _mm256_add_ps(x, _mm256_mul_ps(_mm256_sub_ps(y, x), a)); } #endif #if defined(MA_SUPPORT_NEON) static MA_INLINE float32x4_t ma_mix_f32_fast__neon(float32x4_t x, float32x4_t y, float32x4_t a) { return vaddq_f32(x, vmulq_f32(vsubq_f32(y, x), a)); } #endif static MA_INLINE double ma_mix_f64(double x, double y, double a) { return x*(1-a) + y*a; } static MA_INLINE double ma_mix_f64_fast(double x, double y, double a) { return x + (y - x)*a; } static MA_INLINE float ma_scale_to_range_f32(float x, float lo, float hi) { return lo + x*(hi-lo); } /* Greatest common factor using Euclid's algorithm iteratively. */ static MA_INLINE ma_uint32 ma_gcf_u32(ma_uint32 a, ma_uint32 b) { for (;;) { if (b == 0) { break; } else { ma_uint32 t = a; a = b; b = t % a; } } return a; } static ma_uint32 ma_ffs_32(ma_uint32 x) { ma_uint32 i; /* Just a naive implementation just to get things working for now. Will optimize this later. */ for (i = 0; i < 32; i += 1) { if ((x & (1U << i)) != 0) { return i; } } return i; } static MA_INLINE ma_int16 ma_float_to_fixed_16(float x) { return (ma_int16)(x * (1 << 8)); } /* Random Number Generation miniaudio uses the LCG random number generation algorithm. This is good enough for audio. Note that miniaudio's global LCG implementation uses global state which is _not_ thread-local. When this is called across multiple threads, results will be unpredictable. However, it won't crash and results will still be random enough for miniaudio's purposes. */ #ifndef MA_DEFAULT_LCG_SEED #define MA_DEFAULT_LCG_SEED 4321 #endif #define MA_LCG_M 2147483647 #define MA_LCG_A 48271 #define MA_LCG_C 0 static ma_lcg g_maLCG = {MA_DEFAULT_LCG_SEED}; /* Non-zero initial seed. Use ma_lcg_seed() to use an explicit seed. */ static MA_INLINE void ma_lcg_seed(ma_lcg* pLCG, ma_int32 seed) { MA_ASSERT(pLCG != NULL); pLCG->state = seed; } static MA_INLINE ma_int32 ma_lcg_rand_s32(ma_lcg* pLCG) { pLCG->state = (MA_LCG_A * pLCG->state + MA_LCG_C) % MA_LCG_M; return pLCG->state; } static MA_INLINE ma_uint32 ma_lcg_rand_u32(ma_lcg* pLCG) { return (ma_uint32)ma_lcg_rand_s32(pLCG); } static MA_INLINE ma_int16 ma_lcg_rand_s16(ma_lcg* pLCG) { return (ma_int16)(ma_lcg_rand_s32(pLCG) & 0xFFFF); } static MA_INLINE double ma_lcg_rand_f64(ma_lcg* pLCG) { return ma_lcg_rand_s32(pLCG) / (double)0x7FFFFFFF; } static MA_INLINE float ma_lcg_rand_f32(ma_lcg* pLCG) { return (float)ma_lcg_rand_f64(pLCG); } static MA_INLINE float ma_lcg_rand_range_f32(ma_lcg* pLCG, float lo, float hi) { return ma_scale_to_range_f32(ma_lcg_rand_f32(pLCG), lo, hi); } static MA_INLINE ma_int32 ma_lcg_rand_range_s32(ma_lcg* pLCG, ma_int32 lo, ma_int32 hi) { if (lo == hi) { return lo; } return lo + ma_lcg_rand_u32(pLCG) / (0xFFFFFFFF / (hi - lo + 1) + 1); } #if 0 /* Currently unused. */ static MA_INLINE void ma_seed(ma_int32 seed) { ma_lcg_seed(&g_maLCG, seed); } static MA_INLINE ma_int32 ma_rand_s32(void) { return ma_lcg_rand_s32(&g_maLCG); } static MA_INLINE ma_uint32 ma_rand_u32(void) { return ma_lcg_rand_u32(&g_maLCG); } static MA_INLINE double ma_rand_f64(void) { return ma_lcg_rand_f64(&g_maLCG); } static MA_INLINE float ma_rand_f32(void) { return ma_lcg_rand_f32(&g_maLCG); } #endif static MA_INLINE float ma_rand_range_f32(float lo, float hi) { return ma_lcg_rand_range_f32(&g_maLCG, lo, hi); } static MA_INLINE ma_int32 ma_rand_range_s32(ma_int32 lo, ma_int32 hi) { return ma_lcg_rand_range_s32(&g_maLCG, lo, hi); } static MA_INLINE float ma_dither_f32_rectangle(float ditherMin, float ditherMax) { return ma_rand_range_f32(ditherMin, ditherMax); } static MA_INLINE float ma_dither_f32_triangle(float ditherMin, float ditherMax) { float a = ma_rand_range_f32(ditherMin, 0); float b = ma_rand_range_f32(0, ditherMax); return a + b; } static MA_INLINE float ma_dither_f32(ma_dither_mode ditherMode, float ditherMin, float ditherMax) { if (ditherMode == ma_dither_mode_rectangle) { return ma_dither_f32_rectangle(ditherMin, ditherMax); } if (ditherMode == ma_dither_mode_triangle) { return ma_dither_f32_triangle(ditherMin, ditherMax); } return 0; } static MA_INLINE ma_int32 ma_dither_s32(ma_dither_mode ditherMode, ma_int32 ditherMin, ma_int32 ditherMax) { if (ditherMode == ma_dither_mode_rectangle) { ma_int32 a = ma_rand_range_s32(ditherMin, ditherMax); return a; } if (ditherMode == ma_dither_mode_triangle) { ma_int32 a = ma_rand_range_s32(ditherMin, 0); ma_int32 b = ma_rand_range_s32(0, ditherMax); return a + b; } return 0; } /************************************************************************************************************************************************************** Atomics **************************************************************************************************************************************************************/ /* c89atomic.h begin */ #ifndef ma_atomic_h #define ma_atomic_h #if defined(__cplusplus) extern "C" { #endif #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) #pragma GCC diagnostic push #pragma GCC diagnostic ignored "-Wlong-long" #if defined(__clang__) #pragma GCC diagnostic ignored "-Wc++11-long-long" #endif #endif typedef int ma_atomic_memory_order; #if !defined(MA_ATOMIC_MODERN_MSVC) && \ !defined(MA_ATOMIC_LEGACY_MSVC) && \ !defined(MA_ATOMIC_LEGACY_MSVC_ASM) && \ !defined(MA_ATOMIC_MODERN_GCC) && \ !defined(MA_ATOMIC_LEGACY_GCC) && \ !defined(MA_ATOMIC_LEGACY_GCC_ASM) #if defined(_MSC_VER) || defined(__WATCOMC__) || defined(__DMC__) || defined(__BORLANDC__) #if (defined(_MSC_VER) && _MSC_VER > 1600) #define MA_ATOMIC_MODERN_MSVC #else #if defined(MA_X64) #define MA_ATOMIC_LEGACY_MSVC #else #define MA_ATOMIC_LEGACY_MSVC_ASM #endif #endif #elif (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 7))) || defined(__clang__) #define MA_ATOMIC_MODERN_GCC #else #if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1)) #define MA_ATOMIC_LEGACY_GCC #else #define MA_ATOMIC_LEGACY_GCC_ASM #endif #endif #endif #if defined(MA_ATOMIC_MODERN_MSVC) || defined(MA_ATOMIC_LEGACY_MSVC) #include <intrin.h> #define ma_atomic_memory_order_relaxed 1 #define ma_atomic_memory_order_consume 2 #define ma_atomic_memory_order_acquire 3 #define ma_atomic_memory_order_release 4 #define ma_atomic_memory_order_acq_rel 5 #define ma_atomic_memory_order_seq_cst 6 #define MA_ATOMIC_MSVC_ARM_INTRINSIC_NORETURN(dst, src, order, intrin, ma_atomicType, msvcType) \ switch (order) \ { \ case ma_atomic_memory_order_relaxed: \ { \ intrin##_nf((volatile msvcType*)dst, (msvcType)src); \ } break; \ case ma_atomic_memory_order_consume: \ case ma_atomic_memory_order_acquire: \ { \ intrin##_acq((volatile msvcType*)dst, (msvcType)src); \ } break; \ case ma_atomic_memory_order_release: \ { \ intrin##_rel((volatile msvcType*)dst, (msvcType)src); \ } break; \ case ma_atomic_memory_order_acq_rel: \ case ma_atomic_memory_order_seq_cst: \ default: \ { \ intrin((volatile msvcType*)dst, (msvcType)src); \ } break; \ } #define MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, intrin, ma_atomicType, msvcType) \ ma_atomicType result; \ switch (order) \ { \ case ma_atomic_memory_order_relaxed: \ { \ result = (ma_atomicType)intrin##_nf((volatile msvcType*)dst, (msvcType)src); \ } break; \ case ma_atomic_memory_order_consume: \ case ma_atomic_memory_order_acquire: \ { \ result = (ma_atomicType)intrin##_acq((volatile msvcType*)dst, (msvcType)src); \ } break; \ case ma_atomic_memory_order_release: \ { \ result = (ma_atomicType)intrin##_rel((volatile msvcType*)dst, (msvcType)src); \ } break; \ case ma_atomic_memory_order_acq_rel: \ case ma_atomic_memory_order_seq_cst: \ default: \ { \ result = (ma_atomicType)intrin((volatile msvcType*)dst, (msvcType)src); \ } break; \ } \ return result; typedef ma_uint32 ma_atomic_flag; static MA_INLINE ma_atomic_flag ma_atomic_flag_test_and_set_explicit(volatile ma_atomic_flag* dst, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, 1, order, _InterlockedExchange, ma_atomic_flag, long); } #else { (void)order; return (ma_atomic_flag)_InterlockedExchange((volatile long*)dst, (long)1); } #endif } static MA_INLINE void ma_atomic_flag_clear_explicit(volatile ma_atomic_flag* dst, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC_NORETURN(dst, 0, order, _InterlockedExchange, ma_atomic_flag, long); } #else { (void)order; _InterlockedExchange((volatile long*)dst, (long)0); } #endif } static MA_INLINE ma_atomic_flag ma_atomic_flag_load_explicit(volatile const ma_atomic_flag* dst, ma_atomic_memory_order order) { (void)order; return (ma_uint32)_InterlockedCompareExchange((volatile long*)dst, 0, 0); } #endif #if defined(MA_ATOMIC_LEGACY_MSVC_ASM) #define ma_atomic_memory_order_relaxed 1 #define ma_atomic_memory_order_consume 2 #define ma_atomic_memory_order_acquire 3 #define ma_atomic_memory_order_release 4 #define ma_atomic_memory_order_acq_rel 5 #define ma_atomic_memory_order_seq_cst 6 typedef ma_uint32 ma_atomic_flag; static MA_INLINE ma_atomic_flag ma_atomic_flag_test_and_set_explicit(volatile ma_atomic_flag* dst, ma_atomic_memory_order order) { ma_atomic_flag result = 0; (void)order; __asm { mov ecx, dst mov eax, 1 xchg [ecx], eax mov result, eax } return result; } static MA_INLINE void ma_atomic_flag_clear_explicit(volatile ma_atomic_flag* dst, ma_atomic_memory_order order) { if (order == ma_atomic_memory_order_relaxed) { __asm { mov esi, dst mov dword ptr [esi], 0 } } else { __asm { mov esi, dst mov eax, 0 xchg [esi], eax } } } static MA_INLINE ma_atomic_flag ma_atomic_flag_load_explicit(volatile const ma_atomic_flag* dst, ma_atomic_memory_order order) { ma_atomic_flag result = 0; if (order == ma_atomic_memory_order_relaxed) { __asm { mov esi, dst mov eax, [esi] mov result, eax } } else if (order <= ma_atomic_memory_order_release) { __asm { mov esi, dst mov eax, [esi] lock add dword ptr [esp], 0 mov result, eax } } else { __asm { lock add dword ptr [esp], 0 mov esi, dst mov eax, [esi] mov result, eax lock add dword ptr [esp], 0 } } return result; } #endif #if defined(MA_ATOMIC_MODERN_GCC) #define ma_atomic_memory_order_relaxed __ATOMIC_RELAXED #define ma_atomic_memory_order_consume __ATOMIC_CONSUME #define ma_atomic_memory_order_acquire __ATOMIC_ACQUIRE #define ma_atomic_memory_order_release __ATOMIC_RELEASE #define ma_atomic_memory_order_acq_rel __ATOMIC_ACQ_REL #define ma_atomic_memory_order_seq_cst __ATOMIC_SEQ_CST typedef ma_uint32 ma_atomic_flag; #define ma_atomic_flag_test_and_set_explicit(dst, order) __atomic_exchange_n(dst, 1, order) #define ma_atomic_flag_clear_explicit(dst, order) __atomic_store_n(dst, 0, order) #define ma_atomic_flag_load_explicit(dst, order) __atomic_load_n(dst, order) #endif #if defined(MA_ATOMIC_LEGACY_GCC) #define ma_atomic_memory_order_relaxed 1 #define ma_atomic_memory_order_consume 2 #define ma_atomic_memory_order_acquire 3 #define ma_atomic_memory_order_release 4 #define ma_atomic_memory_order_acq_rel 5 #define ma_atomic_memory_order_seq_cst 6 typedef ma_uint32 ma_atomic_flag; static MA_INLINE ma_atomic_flag ma_atomic_flag_test_and_set_explicit(volatile ma_atomic_flag* dst, ma_atomic_memory_order order) { if (order > ma_atomic_memory_order_acquire) { __sync_synchronize(); } return __sync_lock_test_and_set(dst, 1); } static MA_INLINE void ma_atomic_flag_clear_explicit(volatile ma_atomic_flag* dst, ma_atomic_memory_order order) { if (order > ma_atomic_memory_order_release) { __sync_synchronize(); } __sync_lock_release(dst); } static MA_INLINE ma_atomic_flag ma_atomic_flag_load_explicit(volatile const ma_atomic_flag* dst, ma_atomic_memory_order order) { (void)order; return __sync_val_compare_and_swap((ma_atomic_flag*)dst, 0, 0); } #endif #if defined(MA_ATOMIC_LEGACY_GCC_ASM) #define ma_atomic_memory_order_relaxed 1 #define ma_atomic_memory_order_consume 2 #define ma_atomic_memory_order_acquire 3 #define ma_atomic_memory_order_release 4 #define ma_atomic_memory_order_acq_rel 5 #define ma_atomic_memory_order_seq_cst 6 #if defined(MA_X86) #define ma_atomic_thread_fence(order) __asm__ __volatile__("lock; addl $0, (%%esp)" ::: "memory") #elif defined(MA_X64) #define ma_atomic_thread_fence(order) __asm__ __volatile__("lock; addq $0, (%%rsp)" ::: "memory") #else #error Unsupported architecture. #endif #define MA_ATOMIC_XCHG_GCC_X86(instructionSizeSuffix, result, dst, src) \ __asm__ __volatile__( \ "xchg"instructionSizeSuffix" %0, %1" \ : "=r"(result), \ "=m"(*dst) \ : "0"(src), \ "m"(*dst) \ : "memory" \ ) #define MA_ATOMIC_LOAD_RELAXED_GCC_X86(instructionSizeSuffix, result, dst) \ __asm__ __volatile__( \ "mov"instructionSizeSuffix" %1, %0" \ : "=r"(result) \ : "m"(*dst) \ ) #define MA_ATOMIC_LOAD_RELEASE_GCC_X86(instructionSizeSuffix, result, dst) \ ma_atomic_thread_fence(ma_atomic_memory_order_release); \ __asm__ __volatile__( \ "mov"instructionSizeSuffix" %1, %0" \ : "=r"(result) \ : "m"(*dst) \ : "memory" \ ) #define MA_ATOMIC_LOAD_SEQ_CST_GCC_X86(instructionSizeSuffix, result, dst) \ ma_atomic_thread_fence(ma_atomic_memory_order_seq_cst); \ __asm__ __volatile__( \ "mov"instructionSizeSuffix" %1, %0" \ : "=r"(result) \ : "m"(*dst) \ : "memory" \ ); \ ma_atomic_thread_fence(ma_atomic_memory_order_seq_cst) typedef ma_uint32 ma_atomic_flag; static MA_INLINE ma_atomic_flag ma_atomic_flag_test_and_set_explicit(volatile ma_atomic_flag* dst, ma_atomic_memory_order order) { ma_atomic_flag result; #if defined(MA_X86) || defined(MA_X64) { (void)order; MA_ATOMIC_XCHG_GCC_X86("l", result, dst, 1); } #else { #error Unsupported architecture. } #endif return result; } static MA_INLINE void ma_atomic_flag_clear_explicit(volatile ma_atomic_flag* dst, ma_atomic_memory_order order) { #if defined(MA_X86) || defined(MA_X64) { if (order == ma_atomic_memory_order_relaxed) { __asm__ __volatile__( "movl $0, %0" : "=m"(*dst) ); } else if (order == ma_atomic_memory_order_release) { __asm__ __volatile__( "movl $0, %0" : "=m"(*dst) : : "memory" ); } else { ma_atomic_flag tmp = 0; __asm__ __volatile__( "xchgl %0, %1" : "=r"(tmp), "=m"(*dst) : "0"(tmp), "m"(*dst) : "memory" ); } } #else { #error Unsupported architecture. } #endif } static MA_INLINE ma_atomic_flag ma_atomic_flag_load_explicit(volatile const ma_atomic_flag* dst, ma_atomic_memory_order order) { #if defined(MA_X86) || defined(MA_X64) { ma_atomic_flag result; if (order == ma_atomic_memory_order_relaxed) { MA_ATOMIC_LOAD_RELAXED_GCC_X86("l", result, dst); } else if (order <= ma_atomic_memory_order_release) { MA_ATOMIC_LOAD_RELEASE_GCC_X86("l", result, dst); } else { MA_ATOMIC_LOAD_SEQ_CST_GCC_X86("l", result, dst); } return result; } #else { #error Unsupported architecture. } #endif } #endif #define ma_atomic_flag_test_and_set(dst) ma_atomic_flag_test_and_set_explicit(dst, ma_atomic_memory_order_acquire) #define ma_atomic_flag_clear(dst) ma_atomic_flag_clear_explicit(dst, ma_atomic_memory_order_release) typedef ma_atomic_flag ma_atomic_spinlock; static MA_INLINE void ma_atomic_spinlock_lock(volatile ma_atomic_spinlock* pSpinlock) { for (;;) { if (ma_atomic_flag_test_and_set_explicit(pSpinlock, ma_atomic_memory_order_acquire) == 0) { break; } while (ma_atomic_flag_load_explicit(pSpinlock, ma_atomic_memory_order_relaxed) == 1) { } } } static MA_INLINE void ma_atomic_spinlock_unlock(volatile ma_atomic_spinlock* pSpinlock) { ma_atomic_flag_clear_explicit(pSpinlock, ma_atomic_memory_order_release); } ma_atomic_spinlock ma_atomic_global_lock; #if defined(MA_ATOMIC_MODERN_MSVC) || defined(MA_ATOMIC_LEGACY_MSVC) || defined(MA_ATOMIC_LEGACY_MSVC_ASM) || defined(MA_ATOMIC_LEGACY_GCC) || defined(MA_ATOMIC_LEGACY_GCC_ASM) #if defined(MA_X64) || (defined(MA_X86) && ((defined(__GNUC__) && defined(__i486__)) || (defined(_M_IX86) && _M_IX86 >= 400))) #if defined(MA_ATOMIC_LEGACY_MSVC) && defined(MA_X64) #else #define MA_ATOMIC_IS_LOCK_FREE_8 1 #define MA_ATOMIC_IS_LOCK_FREE_16 1 #endif #define MA_ATOMIC_IS_LOCK_FREE_32 1 #if defined(MA_X64) || (defined(MA_X86) && ((defined(__GNUC__) && defined(__i586__)) || (defined(_M_IX86) && _M_IX86 >= 500))) #define MA_ATOMIC_IS_LOCK_FREE_64 1 #else #endif #else #endif #if defined(MA_ARM32) || defined(MA_ARM64) #define MA_ATOMIC_IS_LOCK_FREE_8 1 #define MA_ATOMIC_IS_LOCK_FREE_16 1 #define MA_ATOMIC_IS_LOCK_FREE_32 1 #if defined(MA_ARM64) || defined(__ARM_ARCH_7A__) || defined(__ARM_ARCH_7R__) || defined(__ARM_ARCH_6K__) || defined(__ARM_ARCH_6Z__) || defined(__ARM_ARCH_6ZK__) #define MA_ATOMIC_IS_LOCK_FREE_64 1 #endif #endif #if defined(MA_ATOMIC_PPC32) || defined(MA_ATOMIC_PPC64) #if (defined(__GNUC__) && (__GNUC__ < 4 || (__GNUC__ == 4 && __GNUC_MINOR__ < 7))) && !defined(__clang__) #else #define MA_ATOMIC_IS_LOCK_FREE_8 1 #define MA_ATOMIC_IS_LOCK_FREE_16 1 #endif #define MA_ATOMIC_IS_LOCK_FREE_32 1 #if defined(MA_ATOMIC_PPC64) #define MA_ATOMIC_IS_LOCK_FREE_64 1 #endif #endif static MA_INLINE ma_bool32 ma_atomic_is_lock_free_8(volatile void* ptr) { (void)ptr; #if defined(MA_ATOMIC_IS_LOCK_FREE_8) return 1; #else return 0; #endif } static MA_INLINE ma_bool32 ma_atomic_is_lock_free_16(volatile void* ptr) { (void)ptr; #if defined(MA_ATOMIC_IS_LOCK_FREE_16) return 1; #else return 0; #endif } static MA_INLINE ma_bool32 ma_atomic_is_lock_free_32(volatile void* ptr) { (void)ptr; #if defined(MA_ATOMIC_IS_LOCK_FREE_32) return 1; #else return 0; #endif } static MA_INLINE ma_bool32 ma_atomic_is_lock_free_64(volatile void* ptr) { (void)ptr; #if defined(MA_ATOMIC_IS_LOCK_FREE_64) return 1; #else return 0; #endif } #endif #define MA_ATOMIC_COMPARE_AND_SWAP_LOCK(sizeInBits, dst, expected, replacement) \ ma_uint##sizeInBits result; \ ma_atomic_spinlock_lock(&ma_atomic_global_lock); \ { \ result = *dst; \ if (result == expected) { \ *dst = replacement; \ } \ } \ ma_atomic_spinlock_unlock(&ma_atomic_global_lock); \ return result #define MA_ATOMIC_LOAD_EXPLICIT_LOCK(sizeInBits, ptr, order) \ ma_uint##sizeInBits result; \ ma_atomic_spinlock_lock(&ma_atomic_global_lock); \ { \ result = *ptr; \ (void)order; \ } \ ma_atomic_spinlock_unlock(&ma_atomic_global_lock); \ return result #define MA_ATOMIC_STORE_EXPLICIT_LOCK(sizeInBits, dst, src, order) \ ma_atomic_spinlock_lock(&ma_atomic_global_lock); \ { \ *dst = src; \ (void)order; \ } \ ma_atomic_spinlock_unlock(&ma_atomic_global_lock) #define MA_ATOMIC_STORE_EXPLICIT_CAS(sizeInBits, dst, src, order) \ ma_uint##sizeInBits oldValue; \ do { \ oldValue = ma_atomic_load_explicit_##sizeInBits(dst, ma_atomic_memory_order_relaxed); \ } while (ma_atomic_compare_and_swap_##sizeInBits(dst, oldValue, src) != oldValue); \ (void)order #define MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(sizeInBits, dst, src, order) \ ma_uint##sizeInBits result; \ ma_atomic_spinlock_lock(&ma_atomic_global_lock); \ { \ result = *dst; \ *dst = src; \ (void)order; \ } \ ma_atomic_spinlock_unlock(&ma_atomic_global_lock); \ return result #define MA_ATOMIC_EXCHANGE_EXPLICIT_CAS(sizeInBits, dst, src, order) \ ma_uint##sizeInBits oldValue; \ do { \ oldValue = ma_atomic_load_explicit_##sizeInBits(dst, ma_atomic_memory_order_relaxed); \ } while (ma_atomic_compare_and_swap_##sizeInBits(dst, oldValue, src) != oldValue); \ (void)order; \ return oldValue #define MA_ATOMIC_FETCH_ADD_LOCK(sizeInBits, dst, src, order) \ ma_uint##sizeInBits result; \ ma_atomic_spinlock_lock(&ma_atomic_global_lock); \ { \ result = *dst; \ *dst += src; \ (void)order; \ } \ ma_atomic_spinlock_unlock(&ma_atomic_global_lock); \ return result #define MA_ATOMIC_FETCH_ADD_CAS(sizeInBits, dst, src, order) \ ma_uint##sizeInBits oldValue; \ ma_uint##sizeInBits newValue; \ do { \ oldValue = ma_atomic_load_explicit_##sizeInBits(dst, ma_atomic_memory_order_relaxed); \ newValue = oldValue + src; \ } while (ma_atomic_compare_and_swap_##sizeInBits(dst, oldValue, newValue) != oldValue); \ (void)order; \ return oldValue #define MA_ATOMIC_FETCH_AND_CAS(sizeInBits, dst, src, order) \ ma_uint##sizeInBits oldValue; \ ma_uint##sizeInBits newValue; \ do { \ oldValue = ma_atomic_load_explicit_##sizeInBits(dst, ma_atomic_memory_order_relaxed); \ newValue = (ma_uint##sizeInBits)(oldValue & src); \ } while (ma_atomic_compare_and_swap_##sizeInBits(dst, oldValue, newValue) != oldValue); \ (void)order; \ return oldValue #define MA_ATOMIC_FETCH_OR_CAS(sizeInBits, dst, src, order) \ ma_uint##sizeInBits oldValue; \ ma_uint##sizeInBits newValue; \ do { \ oldValue = ma_atomic_load_explicit_##sizeInBits(dst, ma_atomic_memory_order_relaxed); \ newValue = (ma_uint##sizeInBits)(oldValue | src); \ } while (ma_atomic_compare_and_swap_##sizeInBits(dst, oldValue, newValue) != oldValue); \ (void)order; \ return oldValue #define MA_ATOMIC_FETCH_XOR_CAS(sizeInBits, dst, src, order) \ ma_uint##sizeInBits oldValue; \ ma_uint##sizeInBits newValue; \ do { \ oldValue = ma_atomic_load_explicit_##sizeInBits(dst, ma_atomic_memory_order_relaxed); \ newValue = (ma_uint##sizeInBits)(oldValue ^ src); \ } while (ma_atomic_compare_and_swap_##sizeInBits(dst, oldValue, newValue) != oldValue); \ (void)order; \ return oldValue #if defined(MA_ATOMIC_MODERN_MSVC) || defined(MA_ATOMIC_LEGACY_MSVC) #define MA_ATOMIC_MSVC_ARM_INTRINSIC_COMPARE_EXCHANGE(ptr, expected, replacement, order, intrin, ma_atomicType, msvcType) \ ma_atomicType result; \ switch (order) \ { \ case ma_atomic_memory_order_relaxed: \ { \ result = (ma_atomicType)intrin##_nf((volatile msvcType*)ptr, (msvcType)expected, (msvcType)replacement); \ } break; \ case ma_atomic_memory_order_consume: \ case ma_atomic_memory_order_acquire: \ { \ result = (ma_atomicType)intrin##_acq((volatile msvcType*)ptr, (msvcType)expected, (msvcType)replacement); \ } break; \ case ma_atomic_memory_order_release: \ { \ result = (ma_atomicType)intrin##_rel((volatile msvcType*)ptr, (msvcType)expected, (msvcType)replacement); \ } break; \ case ma_atomic_memory_order_acq_rel: \ case ma_atomic_memory_order_seq_cst: \ default: \ { \ result = (ma_atomicType)intrin((volatile msvcType*)ptr, (msvcType)expected, (msvcType)replacement); \ } break; \ } \ return result; #if defined(MA_ATOMIC_IS_LOCK_FREE_8) #define ma_atomic_compare_and_swap_8( dst, expected, replacement) (ma_uint8 )_InterlockedCompareExchange8((volatile char*)dst, (char)replacement, (char)expected) #else static MA_INLINE ma_uint8 __stdcall ma_atomic_compare_and_swap_8(volatile ma_uint8* dst, ma_uint8 expected, ma_uint8 replacement) { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(8, dst, expected, replacement); } #endif #if defined(MA_ATOMIC_IS_LOCK_FREE_16) #define ma_atomic_compare_and_swap_16(dst, expected, replacement) (ma_uint16)_InterlockedCompareExchange16((volatile short*)dst, (short)replacement, (short)expected) #else static MA_INLINE ma_uint16 __stdcall ma_atomic_compare_and_swap_16(volatile ma_uint16* dst, ma_uint16 expected, ma_uint16 replacement) { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(16, dst, expected, replacement); } #endif #if defined(MA_ATOMIC_IS_LOCK_FREE_32) #define ma_atomic_compare_and_swap_32(dst, expected, replacement) (ma_uint32)_InterlockedCompareExchange((volatile long*)dst, (long)replacement, (long)expected) #else static MA_INLINE ma_uint32 __stdcall ma_atomic_compare_and_swap_32(volatile ma_uint32* dst, ma_uint32 expected, ma_uint32 replacement) { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(32, dst, expected, replacement); } #endif #if defined(MA_ATOMIC_IS_LOCK_FREE_32) #define ma_atomic_compare_and_swap_64(dst, expected, replacement) (ma_uint64)_InterlockedCompareExchange64((volatile ma_int64*)dst, (ma_int64)replacement, (ma_int64)expected) #else static MA_INLINE ma_uint64 __stdcall ma_atomic_compare_and_swap_64(volatile ma_uint64* dst, ma_uint64 expected, ma_uint64 replacement) { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(64, dst, expected, replacement); } #endif static MA_INLINE ma_uint8 ma_atomic_load_explicit_8(volatile const ma_uint8* ptr, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC_COMPARE_EXCHANGE(ptr, 0, 0, order, _InterlockedCompareExchange8, ma_uint8, char); } #else { (void)order; return ma_atomic_compare_and_swap_8((volatile ma_uint8*)ptr, 0, 0); } #endif } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(8, ptr, order); } #endif } static MA_INLINE ma_uint16 ma_atomic_load_explicit_16(volatile const ma_uint16* ptr, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC_COMPARE_EXCHANGE(ptr, 0, 0, order, _InterlockedCompareExchange16, ma_uint16, short); } #else { (void)order; return ma_atomic_compare_and_swap_16((volatile ma_uint16*)ptr, 0, 0); } #endif } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(16, ptr, order); } #endif } static MA_INLINE ma_uint32 ma_atomic_load_explicit_32(volatile const ma_uint32* ptr, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC_COMPARE_EXCHANGE(ptr, 0, 0, order, _InterlockedCompareExchange, ma_uint32, long); } #else { (void)order; return ma_atomic_compare_and_swap_32((volatile ma_uint32*)ptr, 0, 0); } #endif } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(32, ptr, order); } #endif } static MA_INLINE ma_uint64 ma_atomic_load_explicit_64(volatile const ma_uint64* ptr, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC_COMPARE_EXCHANGE(ptr, 0, 0, order, _InterlockedCompareExchange64, ma_uint64, long long); } #else { (void)order; return ma_atomic_compare_and_swap_64((volatile ma_uint64*)ptr, 0, 0); } #endif } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(64, ptr, order); } #endif } static MA_INLINE ma_uint8 __stdcall ma_atomic_exchange_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedExchange8, ma_uint8, char); } #else { (void)order; return (ma_uint8)_InterlockedExchange8((volatile char*)dst, (char)src); } #endif } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 __stdcall ma_atomic_exchange_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedExchange16, ma_uint16, short); } #else { (void)order; return (ma_uint16)_InterlockedExchange16((volatile short*)dst, (short)src); } #endif } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 __stdcall ma_atomic_exchange_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedExchange, ma_uint32, long); } #else { (void)order; return (ma_uint32)_InterlockedExchange((volatile long*)dst, (long)src); } #endif } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 __stdcall ma_atomic_exchange_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { #if defined(MA_32BIT) { MA_ATOMIC_EXCHANGE_EXPLICIT_CAS(64, dst, src, order); } #else { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedExchange64, ma_uint64, long long); } #else { (void)order; return (ma_uint64)_InterlockedExchange64((volatile long long*)dst, (long long)src); } #endif } #endif } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 __stdcall ma_atomic_fetch_add_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedExchangeAdd8, ma_uint8, char); } #else { (void)order; return (ma_uint8)_InterlockedExchangeAdd8((volatile char*)dst, (char)src); } #endif } #else { MA_ATOMIC_FETCH_ADD_LOCK(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 __stdcall ma_atomic_fetch_add_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedExchangeAdd16, ma_uint16, short); } #else { (void)order; return (ma_uint16)_InterlockedExchangeAdd16((volatile short*)dst, (short)src); } #endif } #else { MA_ATOMIC_FETCH_ADD_LOCK(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 __stdcall ma_atomic_fetch_add_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedExchangeAdd, ma_uint32, long); } #else { (void)order; return (ma_uint32)_InterlockedExchangeAdd((volatile long*)dst, (long)src); } #endif } #else { MA_ATOMIC_FETCH_ADD_LOCK(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 __stdcall ma_atomic_fetch_add_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { #if defined(MA_32BIT) { MA_ATOMIC_FETCH_ADD_CAS(64, dst, src, order); } #else { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedExchangeAdd64, ma_uint64, long long); } #else { (void)order; return (ma_uint64)_InterlockedExchangeAdd64((volatile long long*)dst, (long long)src); } #endif } #endif } #else { MA_ATOMIC_FETCH_ADD_LOCK(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 __stdcall ma_atomic_fetch_sub_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { return ma_atomic_fetch_add_explicit_8(dst, (ma_uint8)(-(ma_int8)src), order); } static MA_INLINE ma_uint16 __stdcall ma_atomic_fetch_sub_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { return ma_atomic_fetch_add_explicit_16(dst, (ma_uint16)(-(ma_int16)src), order); } static MA_INLINE ma_uint32 __stdcall ma_atomic_fetch_sub_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { return ma_atomic_fetch_add_explicit_32(dst, (ma_uint32)(-(ma_int32)src), order); } static MA_INLINE ma_uint64 __stdcall ma_atomic_fetch_sub_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { return ma_atomic_fetch_add_explicit_64(dst, (ma_uint64)(-(ma_int64)src), order); } static MA_INLINE ma_uint8 __stdcall ma_atomic_fetch_and_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedAnd8, ma_uint8, char); } #else { MA_ATOMIC_FETCH_AND_CAS(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 __stdcall ma_atomic_fetch_and_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedAnd16, ma_uint16, short); } #else { MA_ATOMIC_FETCH_AND_CAS(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 __stdcall ma_atomic_fetch_and_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedAnd, ma_uint32, long); } #else { MA_ATOMIC_FETCH_AND_CAS(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 __stdcall ma_atomic_fetch_and_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedAnd64, ma_uint64, long long); } #else { MA_ATOMIC_FETCH_AND_CAS(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 __stdcall ma_atomic_fetch_or_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedOr8, ma_uint8, char); } #else { MA_ATOMIC_FETCH_OR_CAS(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 __stdcall ma_atomic_fetch_or_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedOr16, ma_uint16, short); } #else { MA_ATOMIC_FETCH_OR_CAS(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 __stdcall ma_atomic_fetch_or_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedOr, ma_uint32, long); } #else { MA_ATOMIC_FETCH_OR_CAS(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 __stdcall ma_atomic_fetch_or_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedOr64, ma_uint64, long long); } #else { MA_ATOMIC_FETCH_OR_CAS(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 __stdcall ma_atomic_fetch_xor_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedXor8, ma_uint8, char); } #else { MA_ATOMIC_FETCH_XOR_CAS(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 __stdcall ma_atomic_fetch_xor_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedXor16, ma_uint16, short); } #else { MA_ATOMIC_FETCH_XOR_CAS(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 __stdcall ma_atomic_fetch_xor_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedXor, ma_uint32, long); } #else { MA_ATOMIC_FETCH_XOR_CAS(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 __stdcall ma_atomic_fetch_xor_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ARM) { MA_ATOMIC_MSVC_ARM_INTRINSIC(dst, src, order, _InterlockedXor64, ma_uint64, long long); } #else { MA_ATOMIC_FETCH_XOR_CAS(64, dst, src, order); } #endif } #define ma_atomic_store_explicit_8( dst, src, order) (void)ma_atomic_exchange_explicit_8 (dst, src, order) #define ma_atomic_store_explicit_16(dst, src, order) (void)ma_atomic_exchange_explicit_16(dst, src, order) #define ma_atomic_store_explicit_32(dst, src, order) (void)ma_atomic_exchange_explicit_32(dst, src, order) #define ma_atomic_store_explicit_64(dst, src, order) (void)ma_atomic_exchange_explicit_64(dst, src, order) #if defined(MA_X64) #define ma_atomic_thread_fence(order) __faststorefence(), (void)order #elif defined(MA_ARM64) #define ma_atomic_thread_fence(order) __dmb(_ARM64_BARRIER_ISH), (void)order #else static MA_INLINE void ma_atomic_thread_fence(ma_atomic_memory_order order) { volatile ma_uint32 barrier = 0; ma_atomic_fetch_add_explicit_32(&barrier, 0, order); } #endif #define ma_atomic_signal_fence(order) _ReadWriteBarrier(), (void)order #endif #if defined(MA_ATOMIC_LEGACY_MSVC_ASM) static MA_INLINE ma_uint8 __stdcall ma_atomic_compare_and_swap_8(volatile ma_uint8* dst, ma_uint8 expected, ma_uint8 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { ma_uint8 result = 0; __asm { mov ecx, dst mov al, expected mov dl, replacement lock cmpxchg [ecx], dl mov result, al } return result; } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(8, dst, expected, replacement); } #endif } static MA_INLINE ma_uint16 __stdcall ma_atomic_compare_and_swap_16(volatile ma_uint16* dst, ma_uint16 expected, ma_uint16 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { ma_uint16 result = 0; __asm { mov ecx, dst mov ax, expected mov dx, replacement lock cmpxchg [ecx], dx mov result, ax } return result; } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(16, dst, expected, replacement); } #endif } static MA_INLINE ma_uint32 __stdcall ma_atomic_compare_and_swap_32(volatile ma_uint32* dst, ma_uint32 expected, ma_uint32 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { ma_uint32 result = 0; __asm { mov ecx, dst mov eax, expected mov edx, replacement lock cmpxchg [ecx], edx mov result, eax } return result; } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(32, dst, expected, replacement); } #endif } static MA_INLINE ma_uint64 __stdcall ma_atomic_compare_and_swap_64(volatile ma_uint64* dst, ma_uint64 expected, ma_uint64 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { ma_uint32 resultEAX = 0; ma_uint32 resultEDX = 0; __asm { mov esi, dst mov eax, dword ptr expected mov edx, dword ptr expected + 4 mov ebx, dword ptr replacement mov ecx, dword ptr replacement + 4 lock cmpxchg8b qword ptr [esi] mov resultEAX, eax mov resultEDX, edx } return ((ma_uint64)resultEDX << 32) | resultEAX; } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(64, dst, expected, replacement); } #endif } static MA_INLINE ma_uint8 ma_atomic_load_explicit_8(volatile const ma_uint8* dst, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { ma_uint8 result = 0; if (order == ma_atomic_memory_order_relaxed) { __asm { mov esi, dst mov al, [esi] mov result, al } } else if (order <= ma_atomic_memory_order_release) { __asm { mov esi, dst mov al, [esi] lock add dword ptr [esp], 0 mov result, al } } else { __asm { lock add dword ptr [esp], 0 mov esi, dst mov al, [esi] mov result, al lock add dword ptr [esp], 0 } } return result; } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(8, dst, order); } #endif } static MA_INLINE ma_uint16 ma_atomic_load_explicit_16(volatile const ma_uint16* dst, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { ma_uint16 result = 0; if (order == ma_atomic_memory_order_relaxed) { __asm { mov esi, dst mov ax, [esi] mov result, ax } } else if (order <= ma_atomic_memory_order_release) { __asm { mov esi, dst mov ax, [esi] lock add dword ptr [esp], 0 mov result, ax } } else { __asm { lock add dword ptr [esp], 0 mov esi, dst mov ax, [esi] mov result, ax lock add dword ptr [esp], 0 } } return result; } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(16, dst, order); } #endif } static MA_INLINE ma_uint32 ma_atomic_load_explicit_32(volatile const ma_uint32* dst, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { ma_uint32 result = 0; if (order == ma_atomic_memory_order_relaxed) { __asm { mov esi, dst mov eax, [esi] mov result, eax } } else if (order <= ma_atomic_memory_order_release) { __asm { mov esi, dst mov eax, [esi] lock add dword ptr [esp], 0 mov result, eax } } else { __asm { lock add dword ptr [esp], 0 mov esi, dst mov eax, [esi] mov result, eax lock add dword ptr [esp], 0 } } return result; } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(32, dst, order); } #endif } static MA_INLINE ma_uint64 ma_atomic_load_explicit_64(volatile const ma_uint64* dst, ma_atomic_memory_order order) { (void)order; return ma_atomic_compare_and_swap_64((volatile ma_uint64*)dst, 0, 0); } static MA_INLINE void __stdcall ma_atomic_store_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { if (order == ma_atomic_memory_order_relaxed) { __asm { mov esi, dst mov al, src mov [esi], al } } else { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { __asm { mov esi, dst mov al, src xchg [esi], al } } #else { MA_ATOMIC_STORE_EXPLICIT_LOCK(8, dst, src, order); } #endif } } static MA_INLINE void __stdcall ma_atomic_store_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { if (order == ma_atomic_memory_order_relaxed) { __asm { mov esi, dst mov ax, src mov [esi], ax } } else { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { __asm { mov esi, dst mov ax, src xchg [esi], ax } } #else { MA_ATOMIC_STORE_EXPLICIT_LOCK(16, dst, src, order); } #endif } } static MA_INLINE void __stdcall ma_atomic_store_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { if (order == ma_atomic_memory_order_relaxed) { __asm { mov esi, dst mov eax, src mov [esi], eax } } else { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { __asm { mov esi, dst mov eax, src xchg [esi], eax } } #else { MA_ATOMIC_STORE_EXPLICIT_LOCK(32, dst, src, order); } #endif } } static MA_INLINE void __stdcall ma_atomic_store_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { MA_ATOMIC_STORE_EXPLICIT_CAS(64, dst, src, order); } #else { MA_ATOMIC_STORE_EXPLICIT_LOCK(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 __stdcall ma_atomic_exchange_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { ma_uint8 result = 0; (void)order; __asm { mov ecx, dst mov al, src lock xchg [ecx], al mov result, al } return result; } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 __stdcall ma_atomic_exchange_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { ma_uint16 result = 0; (void)order; __asm { mov ecx, dst mov ax, src lock xchg [ecx], ax mov result, ax } return result; } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 __stdcall ma_atomic_exchange_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { ma_uint32 result = 0; (void)order; __asm { mov ecx, dst mov eax, src xchg [ecx], eax mov result, eax } return result; } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 __stdcall ma_atomic_exchange_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { MA_ATOMIC_EXCHANGE_EXPLICIT_CAS(64, dst, src, order); } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 __stdcall ma_atomic_fetch_add_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { ma_uint8 result = 0; (void)order; __asm { mov ecx, dst mov al, src lock xadd [ecx], al mov result, al } return result; } #else { MA_ATOMIC_FETCH_ADD_LOCK(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 __stdcall ma_atomic_fetch_add_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { ma_uint16 result = 0; (void)order; __asm { mov ecx, dst mov ax, src lock xadd [ecx], ax mov result, ax } return result; } #else { MA_ATOMIC_FETCH_ADD_LOCK(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 __stdcall ma_atomic_fetch_add_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { ma_uint32 result = 0; (void)order; __asm { mov ecx, dst mov eax, src lock xadd [ecx], eax mov result, eax } return result; } #else { MA_ATOMIC_FETCH_ADD_LOCK(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 __stdcall ma_atomic_fetch_add_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { MA_ATOMIC_FETCH_ADD_CAS(64, dst, src, order); } #else { MA_ATOMIC_FETCH_ADD_LOCK(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 __stdcall ma_atomic_fetch_sub_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { ma_uint8 result = 0; (void)order; __asm { mov ecx, dst mov al, src neg al lock xadd [ecx], al mov result, al } return result; } #else { MA_ATOMIC_FETCH_ADD_LOCK(8, dst, (ma_uint8)(-(ma_int8)src), order); } #endif } static MA_INLINE ma_uint16 __stdcall ma_atomic_fetch_sub_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { ma_uint16 result = 0; (void)order; __asm { mov ecx, dst mov ax, src neg ax lock xadd [ecx], ax mov result, ax } return result; } #else { MA_ATOMIC_FETCH_ADD_LOCK(16, dst, (ma_uint16)(-(ma_int16)src), order); } #endif } static MA_INLINE ma_uint32 __stdcall ma_atomic_fetch_sub_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { ma_uint32 result = 0; (void)order; __asm { mov ecx, dst mov eax, src neg eax lock xadd [ecx], eax mov result, eax } return result; } #else { MA_ATOMIC_FETCH_ADD_LOCK(32, dst, (ma_uint32)(-(ma_int32)src), order); } #endif } static MA_INLINE ma_uint64 __stdcall ma_atomic_fetch_sub_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_ADD_CAS(64, dst, (ma_uint64)(-(ma_int64)src), order); } static MA_INLINE ma_uint8 __stdcall ma_atomic_fetch_and_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_AND_CAS(8, dst, src, order); } static MA_INLINE ma_uint16 __stdcall ma_atomic_fetch_and_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_AND_CAS(16, dst, src, order); } static MA_INLINE ma_uint32 __stdcall ma_atomic_fetch_and_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_AND_CAS(32, dst, src, order); } static MA_INLINE ma_uint64 __stdcall ma_atomic_fetch_and_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_AND_CAS(64, dst, src, order); } static MA_INLINE ma_uint8 __stdcall ma_atomic_fetch_or_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_OR_CAS(8, dst, src, order); } static MA_INLINE ma_uint16 __stdcall ma_atomic_fetch_or_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_OR_CAS(16, dst, src, order); } static MA_INLINE ma_uint32 __stdcall ma_atomic_fetch_or_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_OR_CAS(32, dst, src, order); } static MA_INLINE ma_uint64 __stdcall ma_atomic_fetch_or_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_OR_CAS(64, dst, src, order); } static MA_INLINE ma_uint8 __stdcall ma_atomic_fetch_xor_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_XOR_CAS(8, dst, src, order); } static MA_INLINE ma_uint16 __stdcall ma_atomic_fetch_xor_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_XOR_CAS(16, dst, src, order); } static MA_INLINE ma_uint32 __stdcall ma_atomic_fetch_xor_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_XOR_CAS(32, dst, src, order); } static MA_INLINE ma_uint64 __stdcall ma_atomic_fetch_xor_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_XOR_CAS(64, dst, src, order); } static MA_INLINE void __stdcall ma_atomic_thread_fence(ma_atomic_memory_order order) { (void)order; __asm { lock add dword ptr [esp], 0 } } #define ma_atomic_signal_fence(order) __asm {}; (void)order #endif #if defined(MA_ATOMIC_MODERN_GCC) #define MA_ATOMIC_HAS_NATIVE_COMPARE_EXCHANGE #define ma_atomic_thread_fence(order) __atomic_thread_fence(order) #define ma_atomic_signal_fence(order) __atomic_signal_fence(order) #define ma_atomic_is_lock_free_8(ptr) __atomic_is_lock_free(1, ptr) #define ma_atomic_is_lock_free_16(ptr) __atomic_is_lock_free(2, ptr) #define ma_atomic_is_lock_free_32(ptr) __atomic_is_lock_free(4, ptr) #define ma_atomic_is_lock_free_64(ptr) __atomic_is_lock_free(8, ptr) #define ma_atomic_store_explicit_8( dst, src, order) __atomic_store_n(dst, src, order) #define ma_atomic_store_explicit_16(dst, src, order) __atomic_store_n(dst, src, order) #define ma_atomic_store_explicit_32(dst, src, order) __atomic_store_n(dst, src, order) #define ma_atomic_store_explicit_64(dst, src, order) __atomic_store_n(dst, src, order) #define ma_atomic_load_explicit_8( dst, order) __atomic_load_n(dst, order) #define ma_atomic_load_explicit_16(dst, order) __atomic_load_n(dst, order) #define ma_atomic_load_explicit_32(dst, order) __atomic_load_n(dst, order) #define ma_atomic_load_explicit_64(dst, order) __atomic_load_n(dst, order) #define ma_atomic_exchange_explicit_8( dst, src, order) __atomic_exchange_n(dst, src, order) #define ma_atomic_exchange_explicit_16(dst, src, order) __atomic_exchange_n(dst, src, order) #define ma_atomic_exchange_explicit_32(dst, src, order) __atomic_exchange_n(dst, src, order) #define ma_atomic_exchange_explicit_64(dst, src, order) __atomic_exchange_n(dst, src, order) #define ma_atomic_compare_exchange_strong_explicit_8( dst, expected, replacement, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, replacement, 0, successOrder, failureOrder) #define ma_atomic_compare_exchange_strong_explicit_16(dst, expected, replacement, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, replacement, 0, successOrder, failureOrder) #define ma_atomic_compare_exchange_strong_explicit_32(dst, expected, replacement, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, replacement, 0, successOrder, failureOrder) #define ma_atomic_compare_exchange_strong_explicit_64(dst, expected, replacement, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, replacement, 0, successOrder, failureOrder) #define ma_atomic_compare_exchange_weak_explicit_8( dst, expected, replacement, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, replacement, 1, successOrder, failureOrder) #define ma_atomic_compare_exchange_weak_explicit_16(dst, expected, replacement, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, replacement, 1, successOrder, failureOrder) #define ma_atomic_compare_exchange_weak_explicit_32(dst, expected, replacement, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, replacement, 1, successOrder, failureOrder) #define ma_atomic_compare_exchange_weak_explicit_64(dst, expected, replacement, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, replacement, 1, successOrder, failureOrder) #define ma_atomic_fetch_add_explicit_8( dst, src, order) __atomic_fetch_add(dst, src, order) #define ma_atomic_fetch_add_explicit_16(dst, src, order) __atomic_fetch_add(dst, src, order) #define ma_atomic_fetch_add_explicit_32(dst, src, order) __atomic_fetch_add(dst, src, order) #define ma_atomic_fetch_add_explicit_64(dst, src, order) __atomic_fetch_add(dst, src, order) #define ma_atomic_fetch_sub_explicit_8( dst, src, order) __atomic_fetch_sub(dst, src, order) #define ma_atomic_fetch_sub_explicit_16(dst, src, order) __atomic_fetch_sub(dst, src, order) #define ma_atomic_fetch_sub_explicit_32(dst, src, order) __atomic_fetch_sub(dst, src, order) #define ma_atomic_fetch_sub_explicit_64(dst, src, order) __atomic_fetch_sub(dst, src, order) #define ma_atomic_fetch_or_explicit_8( dst, src, order) __atomic_fetch_or(dst, src, order) #define ma_atomic_fetch_or_explicit_16(dst, src, order) __atomic_fetch_or(dst, src, order) #define ma_atomic_fetch_or_explicit_32(dst, src, order) __atomic_fetch_or(dst, src, order) #define ma_atomic_fetch_or_explicit_64(dst, src, order) __atomic_fetch_or(dst, src, order) #define ma_atomic_fetch_xor_explicit_8( dst, src, order) __atomic_fetch_xor(dst, src, order) #define ma_atomic_fetch_xor_explicit_16(dst, src, order) __atomic_fetch_xor(dst, src, order) #define ma_atomic_fetch_xor_explicit_32(dst, src, order) __atomic_fetch_xor(dst, src, order) #define ma_atomic_fetch_xor_explicit_64(dst, src, order) __atomic_fetch_xor(dst, src, order) #define ma_atomic_fetch_and_explicit_8( dst, src, order) __atomic_fetch_and(dst, src, order) #define ma_atomic_fetch_and_explicit_16(dst, src, order) __atomic_fetch_and(dst, src, order) #define ma_atomic_fetch_and_explicit_32(dst, src, order) __atomic_fetch_and(dst, src, order) #define ma_atomic_fetch_and_explicit_64(dst, src, order) __atomic_fetch_and(dst, src, order) static MA_INLINE ma_uint8 ma_atomic_compare_and_swap_8(volatile ma_uint8* dst, ma_uint8 expected, ma_uint8 replacement) { __atomic_compare_exchange_n(dst, &expected, replacement, 0, __ATOMIC_SEQ_CST, __ATOMIC_SEQ_CST); return expected; } static MA_INLINE ma_uint16 ma_atomic_compare_and_swap_16(volatile ma_uint16* dst, ma_uint16 expected, ma_uint16 replacement) { __atomic_compare_exchange_n(dst, &expected, replacement, 0, __ATOMIC_SEQ_CST, __ATOMIC_SEQ_CST); return expected; } static MA_INLINE ma_uint32 ma_atomic_compare_and_swap_32(volatile ma_uint32* dst, ma_uint32 expected, ma_uint32 replacement) { __atomic_compare_exchange_n(dst, &expected, replacement, 0, __ATOMIC_SEQ_CST, __ATOMIC_SEQ_CST); return expected; } #if defined(__clang__) #pragma clang diagnostic push #if __clang_major__ >= 8 #pragma clang diagnostic ignored "-Watomic-alignment" #endif #endif static MA_INLINE ma_uint64 ma_atomic_compare_and_swap_64(volatile ma_uint64* dst, ma_uint64 expected, ma_uint64 replacement) { __atomic_compare_exchange_n(dst, &expected, replacement, 0, __ATOMIC_SEQ_CST, __ATOMIC_SEQ_CST); return expected; } #if defined(__clang__) #pragma clang diagnostic pop #endif #endif #if defined(MA_ATOMIC_LEGACY_GCC) || defined(MA_ATOMIC_LEGACY_GCC_ASM) #define ma_atomic_signal_fence(order) __asm__ __volatile__("":::"memory") #if defined(MA_ATOMIC_LEGACY_GCC) #define ma_atomic_thread_fence(order) __sync_synchronize(), (void)order static MA_INLINE ma_uint8 ma_atomic_compare_and_swap_8(volatile ma_uint8* dst, ma_uint8 expected, ma_uint8 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { return __sync_val_compare_and_swap(dst, expected, replacement); } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(8, dst, expected, replacement); } #endif } static MA_INLINE ma_uint16 ma_atomic_compare_and_swap_16(volatile ma_uint16* dst, ma_uint16 expected, ma_uint16 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { return __sync_val_compare_and_swap(dst, expected, replacement); } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(16, dst, expected, replacement); } #endif } static MA_INLINE ma_uint32 ma_atomic_compare_and_swap_32(volatile ma_uint32* dst, ma_uint32 expected, ma_uint32 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { return __sync_val_compare_and_swap(dst, expected, replacement); } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(32, dst, expected, replacement); } #endif } static MA_INLINE ma_uint64 ma_atomic_compare_and_swap_64(volatile ma_uint64* dst, ma_uint64 expected, ma_uint64 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { return __sync_val_compare_and_swap(dst, expected, replacement); } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(64, dst, expected, replacement); } #endif } static MA_INLINE ma_uint8 ma_atomic_load_explicit_8(volatile const ma_uint8* ptr, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { (void)order; return ma_atomic_compare_and_swap_8((ma_uint8*)ptr, 0, 0); } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(8, ptr, order); } #endif } static MA_INLINE ma_uint16 ma_atomic_load_explicit_16(volatile const ma_uint16* ptr, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { (void)order; return ma_atomic_compare_and_swap_16((ma_uint16*)ptr, 0, 0); } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(16, ptr, order); } #endif } static MA_INLINE ma_uint32 ma_atomic_load_explicit_32(volatile const ma_uint32* ptr, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { (void)order; return ma_atomic_compare_and_swap_32((ma_uint32*)ptr, 0, 0); } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(32, ptr, order); } #endif } static MA_INLINE ma_uint64 ma_atomic_load_explicit_64(volatile const ma_uint64* ptr, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { (void)order; return ma_atomic_compare_and_swap_64((ma_uint64*)ptr, 0, 0); } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(64, ptr, order); } #endif } static MA_INLINE ma_uint8 ma_atomic_exchange_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { if (order > ma_atomic_memory_order_acquire) { __sync_synchronize(); } return __sync_lock_test_and_set(dst, src); } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 ma_atomic_exchange_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { if (order > ma_atomic_memory_order_acquire) { __sync_synchronize(); } return __sync_lock_test_and_set(dst, src); } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 ma_atomic_exchange_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { if (order > ma_atomic_memory_order_acquire) { __sync_synchronize(); } return __sync_lock_test_and_set(dst, src); } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 ma_atomic_exchange_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { if (order > ma_atomic_memory_order_acquire) { __sync_synchronize(); } return __sync_lock_test_and_set(dst, src); } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(64, dst, src, order); } #endif } #define ma_atomic_store_explicit_8( dst, src, order) (void)ma_atomic_exchange_explicit_8 (dst, src, order) #define ma_atomic_store_explicit_16(dst, src, order) (void)ma_atomic_exchange_explicit_16(dst, src, order) #define ma_atomic_store_explicit_32(dst, src, order) (void)ma_atomic_exchange_explicit_32(dst, src, order) #define ma_atomic_store_explicit_64(dst, src, order) (void)ma_atomic_exchange_explicit_64(dst, src, order) static MA_INLINE ma_uint8 ma_atomic_fetch_add_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { (void)order; return __sync_fetch_and_add(dst, src); } #else { MA_ATOMIC_FETCH_ADD_LOCK(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 ma_atomic_fetch_add_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { (void)order; return __sync_fetch_and_add(dst, src); } #else { MA_ATOMIC_FETCH_ADD_LOCK(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 ma_atomic_fetch_add_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { (void)order; return __sync_fetch_and_add(dst, src); } #else { MA_ATOMIC_FETCH_ADD_LOCK(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 ma_atomic_fetch_add_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { (void)order; return __sync_fetch_and_add(dst, src); } #else { MA_ATOMIC_FETCH_ADD_LOCK(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 ma_atomic_fetch_sub_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { (void)order; return __sync_fetch_and_sub(dst, src); } #else { MA_ATOMIC_FETCH_ADD_LOCK(8, dst, (ma_uint8)(-(ma_int8)src), order); } #endif } static MA_INLINE ma_uint16 ma_atomic_fetch_sub_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { (void)order; return __sync_fetch_and_sub(dst, src); } #else { MA_ATOMIC_FETCH_ADD_LOCK(16, dst, (ma_uint16)(-(ma_int16)src), order); } #endif } static MA_INLINE ma_uint32 ma_atomic_fetch_sub_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { (void)order; return __sync_fetch_and_sub(dst, src); } #else { MA_ATOMIC_FETCH_ADD_LOCK(32, dst, (ma_uint32)(-(ma_int32)src), order); } #endif } static MA_INLINE ma_uint64 ma_atomic_fetch_sub_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { (void)order; return __sync_fetch_and_sub(dst, src); } #else { MA_ATOMIC_FETCH_ADD_LOCK(64, dst, (ma_uint64)(-(ma_int64)src), order); } #endif } static MA_INLINE ma_uint8 ma_atomic_fetch_and_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { (void)order; return __sync_fetch_and_and(dst, src); } #else { MA_ATOMIC_FETCH_AND_CAS(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 ma_atomic_fetch_and_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { (void)order; return __sync_fetch_and_and(dst, src); } #else { MA_ATOMIC_FETCH_AND_CAS(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 ma_atomic_fetch_and_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { (void)order; return __sync_fetch_and_and(dst, src); } #else { MA_ATOMIC_FETCH_AND_CAS(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 ma_atomic_fetch_and_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { (void)order; return __sync_fetch_and_and(dst, src); } #else { MA_ATOMIC_FETCH_AND_CAS(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 ma_atomic_fetch_or_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { (void)order; return __sync_fetch_and_or(dst, src); } #else { MA_ATOMIC_FETCH_OR_CAS(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 ma_atomic_fetch_or_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { (void)order; return __sync_fetch_and_or(dst, src); } #else { MA_ATOMIC_FETCH_OR_CAS(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 ma_atomic_fetch_or_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { (void)order; return __sync_fetch_and_or(dst, src); } #else { MA_ATOMIC_FETCH_OR_CAS(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 ma_atomic_fetch_or_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { (void)order; return __sync_fetch_and_or(dst, src); } #else { MA_ATOMIC_FETCH_OR_CAS(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 ma_atomic_fetch_xor_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) { (void)order; return __sync_fetch_and_xor(dst, src); } #else { MA_ATOMIC_FETCH_XOR_CAS(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 ma_atomic_fetch_xor_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) { (void)order; return __sync_fetch_and_xor(dst, src); } #else { MA_ATOMIC_FETCH_XOR_CAS(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 ma_atomic_fetch_xor_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) { (void)order; return __sync_fetch_and_xor(dst, src); } #else { MA_ATOMIC_FETCH_XOR_CAS(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 ma_atomic_fetch_xor_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) { (void)order; return __sync_fetch_and_xor(dst, src); } #else { MA_ATOMIC_FETCH_XOR_CAS(64, dst, src, order); } #endif } #elif defined(MA_ATOMIC_LEGACY_GCC_ASM) #define MA_ATOMIC_CMPXCHG_GCC_X86(instructionSizeSuffix, result, dst, expected, replacement) \ __asm__ __volatile__( \ "lock; cmpxchg"instructionSizeSuffix" %2, %1" \ : "=a"(result), \ "=m"(*dst) \ : "r"(replacement), \ "0"(expected), \ "m"(*dst) \ : "cc", "memory") #define MA_ATOMIC_XADD_GCC_X86(instructionSizeSuffix, result, dst, src) \ __asm__ __volatile__( \ "lock; xadd"instructionSizeSuffix" %0, %1" \ : "=a"(result), \ "=m"(*dst) \ : "0"(src), \ "m"(*dst) \ : "cc", "memory") static MA_INLINE ma_uint8 ma_atomic_compare_and_swap_8(volatile ma_uint8* dst, ma_uint8 expected, ma_uint8 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) && (defined(MA_X86) || defined(MA_X64)) { ma_uint8 result; #if defined(MA_X86) || defined(MA_X64) { MA_ATOMIC_CMPXCHG_GCC_X86("b", result, dst, expected, replacement); } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(8, dst, expected, replacement); } #endif } static MA_INLINE ma_uint16 ma_atomic_compare_and_swap_16(volatile ma_uint16* dst, ma_uint16 expected, ma_uint16 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) && (defined(MA_X86) || defined(MA_X64)) { ma_uint16 result; #if defined(MA_X86) || defined(MA_X64) { MA_ATOMIC_CMPXCHG_GCC_X86("w", result, dst, expected, replacement); } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(16, dst, expected, replacement); } #endif } static MA_INLINE ma_uint32 ma_atomic_compare_and_swap_32(volatile ma_uint32* dst, ma_uint32 expected, ma_uint32 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) && (defined(MA_X86) || defined(MA_X64)) { ma_uint32 result; #if defined(MA_X86) || defined(MA_X64) { MA_ATOMIC_CMPXCHG_GCC_X86("l", result, dst, expected, replacement); } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(32, dst, expected, replacement); } #endif } static MA_INLINE ma_uint64 ma_atomic_compare_and_swap_64(volatile ma_uint64* dst, ma_uint64 expected, ma_uint64 replacement) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) && (defined(MA_X86) || defined(MA_X64)) { ma_uint64 result; #if defined(MA_X86) { ma_uint32 resultEAX; ma_uint32 resultEDX; __asm__ __volatile__( "pushl %%ebx\n" "movl %4, %%ebx\n" "lock cmpxchg8b (%%edi)\n" "popl %%ebx\n" : "=a"(resultEAX), "=d"(resultEDX) : "a"((ma_uint32)(expected & 0xFFFFFFFF)), "d"((ma_uint32)(expected >> 32)), "r"((ma_uint32)(replacement & 0xFFFFFFFF)), "c"((ma_uint32)(replacement >> 32)), "D"(dst) : "memory", "cc"); result = ((ma_uint64)resultEDX << 32) | resultEAX; } #elif defined(MA_X64) { MA_ATOMIC_CMPXCHG_GCC_X86("q", result, dst, expected, replacement); } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_COMPARE_AND_SWAP_LOCK(64, dst, expected, replacement); } #endif } static MA_INLINE ma_uint8 ma_atomic_load_explicit_8(volatile const ma_uint8* dst, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) && (defined(MA_X86) || defined(MA_X64)) { ma_uint8 result; #if defined(MA_X86) || defined(MA_X64) { if (order == ma_atomic_memory_order_relaxed) { MA_ATOMIC_LOAD_RELAXED_GCC_X86("b", result, dst); } else if (order <= ma_atomic_memory_order_release) { MA_ATOMIC_LOAD_RELEASE_GCC_X86("b", result, dst); } else { MA_ATOMIC_LOAD_SEQ_CST_GCC_X86("b", result, dst); } } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(8, dst, order); } #endif } static MA_INLINE ma_uint16 ma_atomic_load_explicit_16(volatile const ma_uint16* dst, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) && (defined(MA_X86) || defined(MA_X64)) { ma_uint16 result; #if defined(MA_X86) || defined(MA_X64) { if (order == ma_atomic_memory_order_relaxed) { MA_ATOMIC_LOAD_RELAXED_GCC_X86("w", result, dst); } else if (order <= ma_atomic_memory_order_release) { MA_ATOMIC_LOAD_RELEASE_GCC_X86("w", result, dst); } else { MA_ATOMIC_LOAD_SEQ_CST_GCC_X86("w", result, dst); } } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(16, dst, order); } #endif } static MA_INLINE ma_uint32 ma_atomic_load_explicit_32(volatile const ma_uint32* dst, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) && (defined(MA_X86) || defined(MA_X64)) { ma_uint32 result; #if defined(MA_X86) || defined(MA_X64) { if (order == ma_atomic_memory_order_relaxed) { MA_ATOMIC_LOAD_RELAXED_GCC_X86("l", result, dst); } else if (order <= ma_atomic_memory_order_release) { MA_ATOMIC_LOAD_RELEASE_GCC_X86("l", result, dst); } else { MA_ATOMIC_LOAD_SEQ_CST_GCC_X86("l", result, dst); } } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(32, dst, order); } #endif } static MA_INLINE ma_uint64 ma_atomic_load_explicit_64(volatile const ma_uint64* dst, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) && (defined(MA_X86) || defined(MA_X64)) { ma_uint64 result; #if defined(MA_X64) { if (order == ma_atomic_memory_order_relaxed) { MA_ATOMIC_LOAD_RELAXED_GCC_X86("q", result, dst); } else if (order <= ma_atomic_memory_order_release) { MA_ATOMIC_LOAD_RELEASE_GCC_X86("q", result, dst); } else { MA_ATOMIC_LOAD_SEQ_CST_GCC_X86("q", result, dst); } } #elif defined(MA_X86) { (void)order; return ma_atomic_compare_and_swap_64((volatile ma_uint64*)dst, 0, 0); } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_LOAD_EXPLICIT_LOCK(64, dst, order); } #endif } static MA_INLINE ma_uint8 ma_atomic_exchange_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) && (defined(MA_X86) || defined(MA_X64)) { ma_uint8 result; (void)order; #if defined(MA_X86) || defined(MA_X64) { MA_ATOMIC_XCHG_GCC_X86("b", result, dst, src); } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 ma_atomic_exchange_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) && (defined(MA_X86) || defined(MA_X64)) { ma_uint16 result; (void)order; #if defined(MA_X86) || defined(MA_X64) { MA_ATOMIC_XCHG_GCC_X86("w", result, dst, src); } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 ma_atomic_exchange_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) && (defined(MA_X86) || defined(MA_X64)) { ma_uint32 result; (void)order; #if defined(MA_X86) || defined(MA_X64) { MA_ATOMIC_XCHG_GCC_X86("l", result, dst, src); } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 ma_atomic_exchange_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) && (defined(MA_X86) || defined(MA_X64)) { ma_uint64 result; (void)order; #if defined(MA_X86) { MA_ATOMIC_EXCHANGE_EXPLICIT_CAS(64, dst, src, order); } #elif defined(MA_X64) { MA_ATOMIC_XCHG_GCC_X86("q", result, dst, src); } #else { #error Unsupported architecture. } #endif return result; } #else { MA_ATOMIC_EXCHANGE_EXPLICIT_LOCK(64, dst, src, order); } #endif } static MA_INLINE void ma_atomic_store_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) && (defined(MA_X86) || defined(MA_X64)) { #if defined(MA_X86) || defined(MA_X64) { if (order == ma_atomic_memory_order_relaxed) { __asm__ __volatile__ ( "movb %1, %0" : "=m"(*dst) : "r"(src) ); } else { __asm__ __volatile__ ( "xchgb %1, %0" : "=m"(*dst) : "r"(src) : "memory" ); } } #else { #error Unsupported architecture. } #endif } #else { MA_ATOMIC_STORE_EXPLICIT_LOCK(8, dst, src, order); } #endif } static MA_INLINE void ma_atomic_store_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) && (defined(MA_X86) || defined(MA_X64)) { #if defined(MA_X86) || defined(MA_X64) { if (order == ma_atomic_memory_order_relaxed) { __asm__ __volatile__ ( "movw %1, %0" : "=m"(*dst) : "r"(src) ); } else { __asm__ __volatile__ ( "xchgw %1, %0" : "=m"(*dst) : "r"(src) : "memory" ); } } #else { #error Unsupported architecture. } #endif } #else { MA_ATOMIC_STORE_EXPLICIT_LOCK(16, dst, src, order); } #endif } static MA_INLINE void ma_atomic_store_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) && (defined(MA_X86) || defined(MA_X64)) { #if defined(MA_X86) || defined(MA_X64) { if (order == ma_atomic_memory_order_relaxed) { __asm__ __volatile__ ( "movl %1, %0" : "=m"(*dst) : "r"(src) ); } else { __asm__ __volatile__ ( "xchgl %1, %0" : "=m"(*dst) : "r"(src) : "memory" ); } } #else { #error Unsupported architecture. } #endif } #else { MA_ATOMIC_STORE_EXPLICIT_LOCK(32, dst, src, order); } #endif } static MA_INLINE void ma_atomic_store_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) && (defined(MA_X86) || defined(MA_X64)) { #if defined(MA_X64) { if (order == ma_atomic_memory_order_relaxed) { __asm__ __volatile__ ( "movq %1, %0" : "=m"(*dst) : "r"(src) ); } else { __asm__ __volatile__ ( "xchgq %1, %0" : "=m"(*dst) : "r"(src) : "memory" ); } } #else { MA_ATOMIC_STORE_EXPLICIT_CAS(64, dst, src, order); } #endif } #else { MA_ATOMIC_STORE_EXPLICIT_LOCK(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 ma_atomic_fetch_add_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_8) && (defined(MA_X86) || defined(MA_X64)) { #if defined(MA_X86) || defined(MA_X64) { ma_uint8 result; (void)order; MA_ATOMIC_XADD_GCC_X86("b", result, dst, src); return result; } #else { #error Unsupported architecture. } #endif } #else { MA_ATOMIC_FETCH_ADD_LOCK(8, dst, src, order); } #endif } static MA_INLINE ma_uint16 ma_atomic_fetch_add_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_16) && (defined(MA_X86) || defined(MA_X64)) { #if defined(MA_X86) || defined(MA_X64) { ma_uint16 result; (void)order; MA_ATOMIC_XADD_GCC_X86("w", result, dst, src); return result; } #else { #error Unsupported architecture. } #endif } #else { MA_ATOMIC_FETCH_ADD_LOCK(16, dst, src, order); } #endif } static MA_INLINE ma_uint32 ma_atomic_fetch_add_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_32) && (defined(MA_X86) || defined(MA_X64)) { #if defined(MA_X86) || defined(MA_X64) { ma_uint32 result; (void)order; MA_ATOMIC_XADD_GCC_X86("l", result, dst, src); return result; } #else { #error Unsupported architecture. } #endif } #else { MA_ATOMIC_FETCH_ADD_LOCK(32, dst, src, order); } #endif } static MA_INLINE ma_uint64 ma_atomic_fetch_add_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { #if defined(MA_ATOMIC_IS_LOCK_FREE_64) && (defined(MA_X86) || defined(MA_X64)) { #if defined(MA_X86) { MA_ATOMIC_FETCH_ADD_CAS(64, dst, src, order); } #elif defined(MA_X64) { ma_uint64 result; MA_ATOMIC_XADD_GCC_X86("q", result, dst, src); (void)order; return result; } #else { #error Unsupported architecture. } #endif } #else { MA_ATOMIC_FETCH_ADD_LOCK(64, dst, src, order); } #endif } static MA_INLINE ma_uint8 ma_atomic_fetch_sub_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { return ma_atomic_fetch_add_explicit_8(dst, (ma_uint8)(-(ma_int8)src), order); } static MA_INLINE ma_uint16 ma_atomic_fetch_sub_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { return ma_atomic_fetch_add_explicit_16(dst, (ma_uint16)(-(ma_int16)src), order); } static MA_INLINE ma_uint32 ma_atomic_fetch_sub_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { return ma_atomic_fetch_add_explicit_32(dst, (ma_uint32)(-(ma_int32)src), order); } static MA_INLINE ma_uint64 ma_atomic_fetch_sub_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { return ma_atomic_fetch_add_explicit_64(dst, (ma_uint64)(-(ma_int64)src), order); } static MA_INLINE ma_uint8 ma_atomic_fetch_and_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_AND_CAS(8, dst, src, order); } static MA_INLINE ma_uint16 ma_atomic_fetch_and_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_AND_CAS(16, dst, src, order); } static MA_INLINE ma_uint32 ma_atomic_fetch_and_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_AND_CAS(32, dst, src, order); } static MA_INLINE ma_uint64 ma_atomic_fetch_and_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_AND_CAS(64, dst, src, order); } static MA_INLINE ma_uint8 ma_atomic_fetch_or_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_OR_CAS(8, dst, src, order); } static MA_INLINE ma_uint16 ma_atomic_fetch_or_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_OR_CAS(16, dst, src, order); } static MA_INLINE ma_uint32 ma_atomic_fetch_or_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_OR_CAS(32, dst, src, order); } static MA_INLINE ma_uint64 ma_atomic_fetch_or_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_OR_CAS(64, dst, src, order); } static MA_INLINE ma_uint8 ma_atomic_fetch_xor_explicit_8(volatile ma_uint8* dst, ma_uint8 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_XOR_CAS(8, dst, src, order); } static MA_INLINE ma_uint16 ma_atomic_fetch_xor_explicit_16(volatile ma_uint16* dst, ma_uint16 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_XOR_CAS(16, dst, src, order); } static MA_INLINE ma_uint32 ma_atomic_fetch_xor_explicit_32(volatile ma_uint32* dst, ma_uint32 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_XOR_CAS(32, dst, src, order); } static MA_INLINE ma_uint64 ma_atomic_fetch_xor_explicit_64(volatile ma_uint64* dst, ma_uint64 src, ma_atomic_memory_order order) { MA_ATOMIC_FETCH_XOR_CAS(64, dst, src, order); } #else #error Unsupported compiler. #endif #endif #if !defined(MA_ATOMIC_HAS_NATIVE_COMPARE_EXCHANGE) static MA_INLINE ma_bool32 ma_atomic_compare_exchange_strong_explicit_8(volatile ma_uint8* dst, ma_uint8* expected, ma_uint8 replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { ma_uint8 result; (void)successOrder; (void)failureOrder; result = ma_atomic_compare_and_swap_8(dst, *expected, replacement); if (result == *expected) { return 1; } else { *expected = result; return 0; } } static MA_INLINE ma_bool32 ma_atomic_compare_exchange_strong_explicit_16(volatile ma_uint16* dst, ma_uint16* expected, ma_uint16 replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { ma_uint16 result; (void)successOrder; (void)failureOrder; result = ma_atomic_compare_and_swap_16(dst, *expected, replacement); if (result == *expected) { return 1; } else { *expected = result; return 0; } } static MA_INLINE ma_bool32 ma_atomic_compare_exchange_strong_explicit_32(volatile ma_uint32* dst, ma_uint32* expected, ma_uint32 replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { ma_uint32 result; (void)successOrder; (void)failureOrder; result = ma_atomic_compare_and_swap_32(dst, *expected, replacement); if (result == *expected) { return 1; } else { *expected = result; return 0; } } static MA_INLINE ma_bool32 ma_atomic_compare_exchange_strong_explicit_64(volatile ma_uint64* dst, volatile ma_uint64* expected, ma_uint64 replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { ma_uint64 result; (void)successOrder; (void)failureOrder; result = ma_atomic_compare_and_swap_64(dst, *expected, replacement); if (result == *expected) { return 1; } else { *expected = result; return 0; } } #define ma_atomic_compare_exchange_weak_explicit_8( dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_strong_explicit_8 (dst, expected, replacement, successOrder, failureOrder) #define ma_atomic_compare_exchange_weak_explicit_16(dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_strong_explicit_16(dst, expected, replacement, successOrder, failureOrder) #define ma_atomic_compare_exchange_weak_explicit_32(dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_strong_explicit_32(dst, expected, replacement, successOrder, failureOrder) #define ma_atomic_compare_exchange_weak_explicit_64(dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_strong_explicit_64(dst, expected, replacement, successOrder, failureOrder) #endif #if defined(MA_64BIT) static MA_INLINE ma_bool32 ma_atomic_is_lock_free_ptr(volatile void** ptr) { return ma_atomic_is_lock_free_64((volatile ma_uint64*)ptr); } static MA_INLINE void* ma_atomic_load_explicit_ptr(volatile void** ptr, ma_atomic_memory_order order) { return (void*)ma_atomic_load_explicit_64((volatile ma_uint64*)ptr, order); } static MA_INLINE void ma_atomic_store_explicit_ptr(volatile void** dst, void* src, ma_atomic_memory_order order) { ma_atomic_store_explicit_64((volatile ma_uint64*)dst, (ma_uint64)src, order); } static MA_INLINE void* ma_atomic_exchange_explicit_ptr(volatile void** dst, void* src, ma_atomic_memory_order order) { return (void*)ma_atomic_exchange_explicit_64((volatile ma_uint64*)dst, (ma_uint64)src, order); } static MA_INLINE ma_bool32 ma_atomic_compare_exchange_strong_explicit_ptr(volatile void** dst, void** expected, void* replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { return ma_atomic_compare_exchange_strong_explicit_64((volatile ma_uint64*)dst, (ma_uint64*)expected, (ma_uint64)replacement, successOrder, failureOrder); } static MA_INLINE ma_bool32 ma_atomic_compare_exchange_weak_explicit_ptr(volatile void** dst, void** expected, void* replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { return ma_atomic_compare_exchange_weak_explicit_64((volatile ma_uint64*)dst, (ma_uint64*)expected, (ma_uint64)replacement, successOrder, failureOrder); } static MA_INLINE void* ma_atomic_compare_and_swap_ptr(volatile void** dst, void* expected, void* replacement) { return (void*)ma_atomic_compare_and_swap_64((volatile ma_uint64*)dst, (ma_uint64)expected, (ma_uint64)replacement); } #elif defined(MA_32BIT) static MA_INLINE ma_bool32 ma_atomic_is_lock_free_ptr(volatile void** ptr) { return ma_atomic_is_lock_free_32((volatile ma_uint32*)ptr); } static MA_INLINE void* ma_atomic_load_explicit_ptr(volatile void** ptr, ma_atomic_memory_order order) { return (void*)ma_atomic_load_explicit_32((volatile ma_uint32*)ptr, order); } static MA_INLINE void ma_atomic_store_explicit_ptr(volatile void** dst, void* src, ma_atomic_memory_order order) { ma_atomic_store_explicit_32((volatile ma_uint32*)dst, (ma_uint32)src, order); } static MA_INLINE void* ma_atomic_exchange_explicit_ptr(volatile void** dst, void* src, ma_atomic_memory_order order) { return (void*)ma_atomic_exchange_explicit_32((volatile ma_uint32*)dst, (ma_uint32)src, order); } static MA_INLINE ma_bool32 ma_atomic_compare_exchange_strong_explicit_ptr(volatile void** dst, void** expected, void* replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { return ma_atomic_compare_exchange_strong_explicit_32((volatile ma_uint32*)dst, (ma_uint32*)expected, (ma_uint32)replacement, successOrder, failureOrder); } static MA_INLINE ma_bool32 ma_atomic_compare_exchange_weak_explicit_ptr(volatile void** dst, void** expected, void* replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { return ma_atomic_compare_exchange_weak_explicit_32((volatile ma_uint32*)dst, (ma_uint32*)expected, (ma_uint32)replacement, successOrder, failureOrder); } static MA_INLINE void* ma_atomic_compare_and_swap_ptr(volatile void** dst, void* expected, void* replacement) { return (void*)ma_atomic_compare_and_swap_32((volatile ma_uint32*)dst, (ma_uint32)expected, (ma_uint32)replacement); } #else #error Unsupported architecture. #endif #define ma_atomic_store_ptr(dst, src) ma_atomic_store_explicit_ptr((volatile void**)dst, (void*)src, ma_atomic_memory_order_seq_cst) #define ma_atomic_load_ptr(ptr) ma_atomic_load_explicit_ptr((volatile void**)ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_exchange_ptr(dst, src) ma_atomic_exchange_explicit_ptr((volatile void**)dst, (void*)src, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_strong_ptr(dst, expected, replacement) ma_atomic_compare_exchange_strong_explicit_ptr((volatile void**)dst, (void**)expected, (void*)replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_weak_ptr(dst, expected, replacement) ma_atomic_compare_exchange_weak_explicit_ptr((volatile void**)dst, (void**)expected, (void*)replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_store_8( dst, src) ma_atomic_store_explicit_8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_store_16(dst, src) ma_atomic_store_explicit_16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_store_32(dst, src) ma_atomic_store_explicit_32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_store_64(dst, src) ma_atomic_store_explicit_64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_load_8( ptr) ma_atomic_load_explicit_8( ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_load_16(ptr) ma_atomic_load_explicit_16(ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_load_32(ptr) ma_atomic_load_explicit_32(ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_load_64(ptr) ma_atomic_load_explicit_64(ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_exchange_8( dst, src) ma_atomic_exchange_explicit_8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_exchange_16(dst, src) ma_atomic_exchange_explicit_16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_exchange_32(dst, src) ma_atomic_exchange_explicit_32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_exchange_64(dst, src) ma_atomic_exchange_explicit_64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_strong_8( dst, expected, replacement) ma_atomic_compare_exchange_strong_explicit_8( dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_strong_16(dst, expected, replacement) ma_atomic_compare_exchange_strong_explicit_16(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_strong_32(dst, expected, replacement) ma_atomic_compare_exchange_strong_explicit_32(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_strong_64(dst, expected, replacement) ma_atomic_compare_exchange_strong_explicit_64(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_weak_8( dst, expected, replacement) ma_atomic_compare_exchange_weak_explicit_8( dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_weak_16( dst, expected, replacement) ma_atomic_compare_exchange_weak_explicit_16(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_weak_32( dst, expected, replacement) ma_atomic_compare_exchange_weak_explicit_32(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_weak_64( dst, expected, replacement) ma_atomic_compare_exchange_weak_explicit_64(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_add_8( dst, src) ma_atomic_fetch_add_explicit_8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_add_16(dst, src) ma_atomic_fetch_add_explicit_16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_add_32(dst, src) ma_atomic_fetch_add_explicit_32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_add_64(dst, src) ma_atomic_fetch_add_explicit_64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_sub_8( dst, src) ma_atomic_fetch_sub_explicit_8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_sub_16(dst, src) ma_atomic_fetch_sub_explicit_16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_sub_32(dst, src) ma_atomic_fetch_sub_explicit_32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_sub_64(dst, src) ma_atomic_fetch_sub_explicit_64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_or_8( dst, src) ma_atomic_fetch_or_explicit_8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_or_16(dst, src) ma_atomic_fetch_or_explicit_16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_or_32(dst, src) ma_atomic_fetch_or_explicit_32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_or_64(dst, src) ma_atomic_fetch_or_explicit_64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_xor_8( dst, src) ma_atomic_fetch_xor_explicit_8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_xor_16(dst, src) ma_atomic_fetch_xor_explicit_16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_xor_32(dst, src) ma_atomic_fetch_xor_explicit_32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_xor_64(dst, src) ma_atomic_fetch_xor_explicit_64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_and_8( dst, src) ma_atomic_fetch_and_explicit_8 (dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_and_16(dst, src) ma_atomic_fetch_and_explicit_16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_and_32(dst, src) ma_atomic_fetch_and_explicit_32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_and_64(dst, src) ma_atomic_fetch_and_explicit_64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_store_explicit_i8( dst, src, order) ma_atomic_store_explicit_8( (ma_uint8* )dst, (ma_uint8 )src, order) #define ma_atomic_store_explicit_i16(dst, src, order) ma_atomic_store_explicit_16((ma_uint16*)dst, (ma_uint16)src, order) #define ma_atomic_store_explicit_i32(dst, src, order) ma_atomic_store_explicit_32((ma_uint32*)dst, (ma_uint32)src, order) #define ma_atomic_store_explicit_i64(dst, src, order) ma_atomic_store_explicit_64((ma_uint64*)dst, (ma_uint64)src, order) #define ma_atomic_load_explicit_i8( ptr, order) (ma_int8 )ma_atomic_load_explicit_8( (ma_uint8* )ptr, order) #define ma_atomic_load_explicit_i16(ptr, order) (ma_int16)ma_atomic_load_explicit_16((ma_uint16*)ptr, order) #define ma_atomic_load_explicit_i32(ptr, order) (ma_int32)ma_atomic_load_explicit_32((ma_uint32*)ptr, order) #define ma_atomic_load_explicit_i64(ptr, order) (ma_int64)ma_atomic_load_explicit_64((ma_uint64*)ptr, order) #define ma_atomic_exchange_explicit_i8( dst, src, order) (ma_int8 )ma_atomic_exchange_explicit_8 ((ma_uint8* )dst, (ma_uint8 )src, order) #define ma_atomic_exchange_explicit_i16(dst, src, order) (ma_int16)ma_atomic_exchange_explicit_16((ma_uint16*)dst, (ma_uint16)src, order) #define ma_atomic_exchange_explicit_i32(dst, src, order) (ma_int32)ma_atomic_exchange_explicit_32((ma_uint32*)dst, (ma_uint32)src, order) #define ma_atomic_exchange_explicit_i64(dst, src, order) (ma_int64)ma_atomic_exchange_explicit_64((ma_uint64*)dst, (ma_uint64)src, order) #define ma_atomic_compare_exchange_strong_explicit_i8( dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_strong_explicit_8( (ma_uint8* )dst, (ma_uint8* )expected, (ma_uint8 )replacement, successOrder, failureOrder) #define ma_atomic_compare_exchange_strong_explicit_i16(dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_strong_explicit_16((ma_uint16*)dst, (ma_uint16*)expected, (ma_uint16)replacement, successOrder, failureOrder) #define ma_atomic_compare_exchange_strong_explicit_i32(dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_strong_explicit_32((ma_uint32*)dst, (ma_uint32*)expected, (ma_uint32)replacement, successOrder, failureOrder) #define ma_atomic_compare_exchange_strong_explicit_i64(dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_strong_explicit_64((ma_uint64*)dst, (ma_uint64*)expected, (ma_uint64)replacement, successOrder, failureOrder) #define ma_atomic_compare_exchange_weak_explicit_i8( dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_weak_explicit_8( (ma_uint8* )dst, (ma_uint8* )expected, (ma_uint8 )replacement, successOrder, failureOrder) #define ma_atomic_compare_exchange_weak_explicit_i16(dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_weak_explicit_16((ma_uint16*)dst, (ma_uint16*)expected, (ma_uint16)replacement, successOrder, failureOrder) #define ma_atomic_compare_exchange_weak_explicit_i32(dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_weak_explicit_32((ma_uint32*)dst, (ma_uint32*)expected, (ma_uint32)replacement, successOrder, failureOrder) #define ma_atomic_compare_exchange_weak_explicit_i64(dst, expected, replacement, successOrder, failureOrder) ma_atomic_compare_exchange_weak_explicit_64((ma_uint64*)dst, (ma_uint64*)expected, (ma_uint64)replacement, successOrder, failureOrder) #define ma_atomic_fetch_add_explicit_i8( dst, src, order) (ma_int8 )ma_atomic_fetch_add_explicit_8( (ma_uint8* )dst, (ma_uint8 )src, order) #define ma_atomic_fetch_add_explicit_i16(dst, src, order) (ma_int16)ma_atomic_fetch_add_explicit_16((ma_uint16*)dst, (ma_uint16)src, order) #define ma_atomic_fetch_add_explicit_i32(dst, src, order) (ma_int32)ma_atomic_fetch_add_explicit_32((ma_uint32*)dst, (ma_uint32)src, order) #define ma_atomic_fetch_add_explicit_i64(dst, src, order) (ma_int64)ma_atomic_fetch_add_explicit_64((ma_uint64*)dst, (ma_uint64)src, order) #define ma_atomic_fetch_sub_explicit_i8( dst, src, order) (ma_int8 )ma_atomic_fetch_sub_explicit_8( (ma_uint8* )dst, (ma_uint8 )src, order) #define ma_atomic_fetch_sub_explicit_i16(dst, src, order) (ma_int16)ma_atomic_fetch_sub_explicit_16((ma_uint16*)dst, (ma_uint16)src, order) #define ma_atomic_fetch_sub_explicit_i32(dst, src, order) (ma_int32)ma_atomic_fetch_sub_explicit_32((ma_uint32*)dst, (ma_uint32)src, order) #define ma_atomic_fetch_sub_explicit_i64(dst, src, order) (ma_int64)ma_atomic_fetch_sub_explicit_64((ma_uint64*)dst, (ma_uint64)src, order) #define ma_atomic_fetch_or_explicit_i8( dst, src, order) (ma_int8 )ma_atomic_fetch_or_explicit_8( (ma_uint8* )dst, (ma_uint8 )src, order) #define ma_atomic_fetch_or_explicit_i16(dst, src, order) (ma_int16)ma_atomic_fetch_or_explicit_16((ma_uint16*)dst, (ma_uint16)src, order) #define ma_atomic_fetch_or_explicit_i32(dst, src, order) (ma_int32)ma_atomic_fetch_or_explicit_32((ma_uint32*)dst, (ma_uint32)src, order) #define ma_atomic_fetch_or_explicit_i64(dst, src, order) (ma_int64)ma_atomic_fetch_or_explicit_64((ma_uint64*)dst, (ma_uint64)src, order) #define ma_atomic_fetch_xor_explicit_i8( dst, src, order) (ma_int8 )ma_atomic_fetch_xor_explicit_8( (ma_uint8* )dst, (ma_uint8 )src, order) #define ma_atomic_fetch_xor_explicit_i16(dst, src, order) (ma_int16)ma_atomic_fetch_xor_explicit_16((ma_uint16*)dst, (ma_uint16)src, order) #define ma_atomic_fetch_xor_explicit_i32(dst, src, order) (ma_int32)ma_atomic_fetch_xor_explicit_32((ma_uint32*)dst, (ma_uint32)src, order) #define ma_atomic_fetch_xor_explicit_i64(dst, src, order) (ma_int64)ma_atomic_fetch_xor_explicit_64((ma_uint64*)dst, (ma_uint64)src, order) #define ma_atomic_fetch_and_explicit_i8( dst, src, order) (ma_int8 )ma_atomic_fetch_and_explicit_8( (ma_uint8* )dst, (ma_uint8 )src, order) #define ma_atomic_fetch_and_explicit_i16(dst, src, order) (ma_int16)ma_atomic_fetch_and_explicit_16((ma_uint16*)dst, (ma_uint16)src, order) #define ma_atomic_fetch_and_explicit_i32(dst, src, order) (ma_int32)ma_atomic_fetch_and_explicit_32((ma_uint32*)dst, (ma_uint32)src, order) #define ma_atomic_fetch_and_explicit_i64(dst, src, order) (ma_int64)ma_atomic_fetch_and_explicit_64((ma_uint64*)dst, (ma_uint64)src, order) #define ma_atomic_store_i8( dst, src) ma_atomic_store_explicit_i8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_store_i16(dst, src) ma_atomic_store_explicit_i16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_store_i32(dst, src) ma_atomic_store_explicit_i32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_store_i64(dst, src) ma_atomic_store_explicit_i64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_load_i8( ptr) ma_atomic_load_explicit_i8( ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_load_i16(ptr) ma_atomic_load_explicit_i16(ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_load_i32(ptr) ma_atomic_load_explicit_i32(ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_load_i64(ptr) ma_atomic_load_explicit_i64(ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_exchange_i8( dst, src) ma_atomic_exchange_explicit_i8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_exchange_i16(dst, src) ma_atomic_exchange_explicit_i16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_exchange_i32(dst, src) ma_atomic_exchange_explicit_i32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_exchange_i64(dst, src) ma_atomic_exchange_explicit_i64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_strong_i8( dst, expected, replacement) ma_atomic_compare_exchange_strong_explicit_i8( dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_strong_i16(dst, expected, replacement) ma_atomic_compare_exchange_strong_explicit_i16(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_strong_i32(dst, expected, replacement) ma_atomic_compare_exchange_strong_explicit_i32(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_strong_i64(dst, expected, replacement) ma_atomic_compare_exchange_strong_explicit_i64(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_weak_i8( dst, expected, replacement) ma_atomic_compare_exchange_weak_explicit_i8( dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_weak_i16(dst, expected, replacement) ma_atomic_compare_exchange_weak_explicit_i16(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_weak_i32(dst, expected, replacement) ma_atomic_compare_exchange_weak_explicit_i32(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_weak_i64(dst, expected, replacement) ma_atomic_compare_exchange_weak_explicit_i64(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_add_i8( dst, src) ma_atomic_fetch_add_explicit_i8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_add_i16(dst, src) ma_atomic_fetch_add_explicit_i16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_add_i32(dst, src) ma_atomic_fetch_add_explicit_i32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_add_i64(dst, src) ma_atomic_fetch_add_explicit_i64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_sub_i8( dst, src) ma_atomic_fetch_sub_explicit_i8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_sub_i16(dst, src) ma_atomic_fetch_sub_explicit_i16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_sub_i32(dst, src) ma_atomic_fetch_sub_explicit_i32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_sub_i64(dst, src) ma_atomic_fetch_sub_explicit_i64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_or_i8( dst, src) ma_atomic_fetch_or_explicit_i8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_or_i16(dst, src) ma_atomic_fetch_or_explicit_i16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_or_i32(dst, src) ma_atomic_fetch_or_explicit_i32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_or_i64(dst, src) ma_atomic_fetch_or_explicit_i64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_xor_i8( dst, src) ma_atomic_fetch_xor_explicit_i8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_xor_i16(dst, src) ma_atomic_fetch_xor_explicit_i16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_xor_i32(dst, src) ma_atomic_fetch_xor_explicit_i32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_xor_i64(dst, src) ma_atomic_fetch_xor_explicit_i64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_and_i8( dst, src) ma_atomic_fetch_and_explicit_i8( dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_and_i16(dst, src) ma_atomic_fetch_and_explicit_i16(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_and_i32(dst, src) ma_atomic_fetch_and_explicit_i32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_and_i64(dst, src) ma_atomic_fetch_and_explicit_i64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_and_swap_i8( dst, expected, dedsired) (ma_int8 )ma_atomic_compare_and_swap_8( (ma_uint8* )dst, (ma_uint8 )expected, (ma_uint8 )dedsired) #define ma_atomic_compare_and_swap_i16(dst, expected, dedsired) (ma_int16)ma_atomic_compare_and_swap_16((ma_uint16*)dst, (ma_uint16)expected, (ma_uint16)dedsired) #define ma_atomic_compare_and_swap_i32(dst, expected, dedsired) (ma_int32)ma_atomic_compare_and_swap_32((ma_uint32*)dst, (ma_uint32)expected, (ma_uint32)dedsired) #define ma_atomic_compare_and_swap_i64(dst, expected, dedsired) (ma_int64)ma_atomic_compare_and_swap_64((ma_uint64*)dst, (ma_uint64)expected, (ma_uint64)dedsired) typedef union { ma_uint32 i; float f; } ma_atomic_if32; typedef union { ma_uint64 i; double f; } ma_atomic_if64; #define ma_atomic_clear_explicit_f32(ptr, order) ma_atomic_clear_explicit_32((ma_uint32*)ptr, order) #define ma_atomic_clear_explicit_f64(ptr, order) ma_atomic_clear_explicit_64((ma_uint64*)ptr, order) static MA_INLINE void ma_atomic_store_explicit_f32(volatile float* dst, float src, ma_atomic_memory_order order) { ma_atomic_if32 x; x.f = src; ma_atomic_store_explicit_32((volatile ma_uint32*)dst, x.i, order); } static MA_INLINE void ma_atomic_store_explicit_f64(volatile double* dst, double src, ma_atomic_memory_order order) { ma_atomic_if64 x; x.f = src; ma_atomic_store_explicit_64((volatile ma_uint64*)dst, x.i, order); } static MA_INLINE float ma_atomic_load_explicit_f32(volatile const float* ptr, ma_atomic_memory_order order) { ma_atomic_if32 r; r.i = ma_atomic_load_explicit_32((volatile const ma_uint32*)ptr, order); return r.f; } static MA_INLINE double ma_atomic_load_explicit_f64(volatile const double* ptr, ma_atomic_memory_order order) { ma_atomic_if64 r; r.i = ma_atomic_load_explicit_64((volatile const ma_uint64*)ptr, order); return r.f; } static MA_INLINE float ma_atomic_exchange_explicit_f32(volatile float* dst, float src, ma_atomic_memory_order order) { ma_atomic_if32 r; ma_atomic_if32 x; x.f = src; r.i = ma_atomic_exchange_explicit_32((volatile ma_uint32*)dst, x.i, order); return r.f; } static MA_INLINE double ma_atomic_exchange_explicit_f64(volatile double* dst, double src, ma_atomic_memory_order order) { ma_atomic_if64 r; ma_atomic_if64 x; x.f = src; r.i = ma_atomic_exchange_explicit_64((volatile ma_uint64*)dst, x.i, order); return r.f; } static MA_INLINE ma_bool32 ma_atomic_compare_exchange_strong_explicit_f32(volatile float* dst, float* expected, float replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { ma_atomic_if32 d; d.f = replacement; return ma_atomic_compare_exchange_strong_explicit_32((volatile ma_uint32*)dst, (ma_uint32*)expected, d.i, successOrder, failureOrder); } static MA_INLINE ma_bool32 ma_atomic_compare_exchange_strong_explicit_f64(volatile double* dst, double* expected, double replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { ma_atomic_if64 d; d.f = replacement; return ma_atomic_compare_exchange_strong_explicit_64((volatile ma_uint64*)dst, (ma_uint64*)expected, d.i, successOrder, failureOrder); } static MA_INLINE ma_bool32 ma_atomic_compare_exchange_weak_explicit_f32(volatile float* dst, float* expected, float replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { ma_atomic_if32 d; d.f = replacement; return ma_atomic_compare_exchange_weak_explicit_32((volatile ma_uint32*)dst, (ma_uint32*)expected, d.i, successOrder, failureOrder); } static MA_INLINE ma_bool32 ma_atomic_compare_exchange_weak_explicit_f64(volatile double* dst, double* expected, double replacement, ma_atomic_memory_order successOrder, ma_atomic_memory_order failureOrder) { ma_atomic_if64 d; d.f = replacement; return ma_atomic_compare_exchange_weak_explicit_64((volatile ma_uint64*)dst, (ma_uint64*)expected, d.i, successOrder, failureOrder); } static MA_INLINE float ma_atomic_fetch_add_explicit_f32(volatile float* dst, float src, ma_atomic_memory_order order) { ma_atomic_if32 r; ma_atomic_if32 x; x.f = src; r.i = ma_atomic_fetch_add_explicit_32((volatile ma_uint32*)dst, x.i, order); return r.f; } static MA_INLINE double ma_atomic_fetch_add_explicit_f64(volatile double* dst, double src, ma_atomic_memory_order order) { ma_atomic_if64 r; ma_atomic_if64 x; x.f = src; r.i = ma_atomic_fetch_add_explicit_64((volatile ma_uint64*)dst, x.i, order); return r.f; } static MA_INLINE float ma_atomic_fetch_sub_explicit_f32(volatile float* dst, float src, ma_atomic_memory_order order) { ma_atomic_if32 r; ma_atomic_if32 x; x.f = src; r.i = ma_atomic_fetch_sub_explicit_32((volatile ma_uint32*)dst, x.i, order); return r.f; } static MA_INLINE double ma_atomic_fetch_sub_explicit_f64(volatile double* dst, double src, ma_atomic_memory_order order) { ma_atomic_if64 r; ma_atomic_if64 x; x.f = src; r.i = ma_atomic_fetch_sub_explicit_64((volatile ma_uint64*)dst, x.i, order); return r.f; } static MA_INLINE float ma_atomic_fetch_or_explicit_f32(volatile float* dst, float src, ma_atomic_memory_order order) { ma_atomic_if32 r; ma_atomic_if32 x; x.f = src; r.i = ma_atomic_fetch_or_explicit_32((volatile ma_uint32*)dst, x.i, order); return r.f; } static MA_INLINE double ma_atomic_fetch_or_explicit_f64(volatile double* dst, double src, ma_atomic_memory_order order) { ma_atomic_if64 r; ma_atomic_if64 x; x.f = src; r.i = ma_atomic_fetch_or_explicit_64((volatile ma_uint64*)dst, x.i, order); return r.f; } static MA_INLINE float ma_atomic_fetch_xor_explicit_f32(volatile float* dst, float src, ma_atomic_memory_order order) { ma_atomic_if32 r; ma_atomic_if32 x; x.f = src; r.i = ma_atomic_fetch_xor_explicit_32((volatile ma_uint32*)dst, x.i, order); return r.f; } static MA_INLINE double ma_atomic_fetch_xor_explicit_f64(volatile double* dst, double src, ma_atomic_memory_order order) { ma_atomic_if64 r; ma_atomic_if64 x; x.f = src; r.i = ma_atomic_fetch_xor_explicit_64((volatile ma_uint64*)dst, x.i, order); return r.f; } static MA_INLINE float ma_atomic_fetch_and_explicit_f32(volatile float* dst, float src, ma_atomic_memory_order order) { ma_atomic_if32 r; ma_atomic_if32 x; x.f = src; r.i = ma_atomic_fetch_and_explicit_32((volatile ma_uint32*)dst, x.i, order); return r.f; } static MA_INLINE double ma_atomic_fetch_and_explicit_f64(volatile double* dst, double src, ma_atomic_memory_order order) { ma_atomic_if64 r; ma_atomic_if64 x; x.f = src; r.i = ma_atomic_fetch_and_explicit_64((volatile ma_uint64*)dst, x.i, order); return r.f; } #define ma_atomic_clear_f32(ptr) (float )ma_atomic_clear_explicit_f32(ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_clear_f64(ptr) (double)ma_atomic_clear_explicit_f64(ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_store_f32(dst, src) ma_atomic_store_explicit_f32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_store_f64(dst, src) ma_atomic_store_explicit_f64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_load_f32(ptr) (float )ma_atomic_load_explicit_f32(ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_load_f64(ptr) (double)ma_atomic_load_explicit_f64(ptr, ma_atomic_memory_order_seq_cst) #define ma_atomic_exchange_f32(dst, src) (float )ma_atomic_exchange_explicit_f32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_exchange_f64(dst, src) (double)ma_atomic_exchange_explicit_f64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_strong_f32(dst, expected, replacement) ma_atomic_compare_exchange_strong_explicit_f32(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_strong_f64(dst, expected, replacement) ma_atomic_compare_exchange_strong_explicit_f64(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_weak_f32(dst, expected, replacement) ma_atomic_compare_exchange_weak_explicit_f32(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_compare_exchange_weak_f64(dst, expected, replacement) ma_atomic_compare_exchange_weak_explicit_f64(dst, expected, replacement, ma_atomic_memory_order_seq_cst, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_add_f32(dst, src) ma_atomic_fetch_add_explicit_f32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_add_f64(dst, src) ma_atomic_fetch_add_explicit_f64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_sub_f32(dst, src) ma_atomic_fetch_sub_explicit_f32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_sub_f64(dst, src) ma_atomic_fetch_sub_explicit_f64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_or_f32(dst, src) ma_atomic_fetch_or_explicit_f32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_or_f64(dst, src) ma_atomic_fetch_or_explicit_f64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_xor_f32(dst, src) ma_atomic_fetch_xor_explicit_f32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_xor_f64(dst, src) ma_atomic_fetch_xor_explicit_f64(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_and_f32(dst, src) ma_atomic_fetch_and_explicit_f32(dst, src, ma_atomic_memory_order_seq_cst) #define ma_atomic_fetch_and_f64(dst, src) ma_atomic_fetch_and_explicit_f64(dst, src, ma_atomic_memory_order_seq_cst) static MA_INLINE float ma_atomic_compare_and_swap_f32(volatile float* dst, float expected, float replacement) { ma_atomic_if32 r; ma_atomic_if32 e, d; e.f = expected; d.f = replacement; r.i = ma_atomic_compare_and_swap_32((volatile ma_uint32*)dst, e.i, d.i); return r.f; } static MA_INLINE double ma_atomic_compare_and_swap_f64(volatile double* dst, double expected, double replacement) { ma_atomic_if64 r; ma_atomic_if64 e, d; e.f = expected; d.f = replacement; r.i = ma_atomic_compare_and_swap_64((volatile ma_uint64*)dst, e.i, d.i); return r.f; } #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) #pragma GCC diagnostic pop #endif #if defined(__cplusplus) } #endif #endif /* c89atomic.h end */ #define MA_ATOMIC_SAFE_TYPE_IMPL(c89TypeExtension, type) \ static MA_INLINE ma_##type ma_atomic_##type##_get(ma_atomic_##type* x) \ { \ return (ma_##type)ma_atomic_load_##c89TypeExtension(&x->value); \ } \ static MA_INLINE void ma_atomic_##type##_set(ma_atomic_##type* x, ma_##type value) \ { \ ma_atomic_store_##c89TypeExtension(&x->value, value); \ } \ static MA_INLINE ma_##type ma_atomic_##type##_exchange(ma_atomic_##type* x, ma_##type value) \ { \ return (ma_##type)ma_atomic_exchange_##c89TypeExtension(&x->value, value); \ } \ static MA_INLINE ma_bool32 ma_atomic_##type##_compare_exchange(ma_atomic_##type* x, ma_##type* expected, ma_##type desired) \ { \ return ma_atomic_compare_exchange_weak_##c89TypeExtension(&x->value, expected, desired); \ } \ static MA_INLINE ma_##type ma_atomic_##type##_fetch_add(ma_atomic_##type* x, ma_##type y) \ { \ return (ma_##type)ma_atomic_fetch_add_##c89TypeExtension(&x->value, y); \ } \ static MA_INLINE ma_##type ma_atomic_##type##_fetch_sub(ma_atomic_##type* x, ma_##type y) \ { \ return (ma_##type)ma_atomic_fetch_sub_##c89TypeExtension(&x->value, y); \ } \ static MA_INLINE ma_##type ma_atomic_##type##_fetch_or(ma_atomic_##type* x, ma_##type y) \ { \ return (ma_##type)ma_atomic_fetch_or_##c89TypeExtension(&x->value, y); \ } \ static MA_INLINE ma_##type ma_atomic_##type##_fetch_xor(ma_atomic_##type* x, ma_##type y) \ { \ return (ma_##type)ma_atomic_fetch_xor_##c89TypeExtension(&x->value, y); \ } \ static MA_INLINE ma_##type ma_atomic_##type##_fetch_and(ma_atomic_##type* x, ma_##type y) \ { \ return (ma_##type)ma_atomic_fetch_and_##c89TypeExtension(&x->value, y); \ } \ static MA_INLINE ma_##type ma_atomic_##type##_compare_and_swap(ma_atomic_##type* x, ma_##type expected, ma_##type desired) \ { \ return (ma_##type)ma_atomic_compare_and_swap_##c89TypeExtension(&x->value, expected, desired); \ } \ #define MA_ATOMIC_SAFE_TYPE_IMPL_PTR(type) \ static MA_INLINE ma_##type* ma_atomic_ptr_##type##_get(ma_atomic_ptr_##type* x) \ { \ return ma_atomic_load_ptr((void**)&x->value); \ } \ static MA_INLINE void ma_atomic_ptr_##type##_set(ma_atomic_ptr_##type* x, ma_##type* value) \ { \ ma_atomic_store_ptr((void**)&x->value, (void*)value); \ } \ static MA_INLINE ma_##type* ma_atomic_ptr_##type##_exchange(ma_atomic_ptr_##type* x, ma_##type* value) \ { \ return ma_atomic_exchange_ptr((void**)&x->value, (void*)value); \ } \ static MA_INLINE ma_bool32 ma_atomic_ptr_##type##_compare_exchange(ma_atomic_ptr_##type* x, ma_##type** expected, ma_##type* desired) \ { \ return ma_atomic_compare_exchange_weak_ptr((void**)&x->value, (void*)expected, (void*)desired); \ } \ static MA_INLINE ma_##type* ma_atomic_ptr_##type##_compare_and_swap(ma_atomic_ptr_##type* x, ma_##type* expected, ma_##type* desired) \ { \ return (ma_##type*)ma_atomic_compare_and_swap_ptr((void**)&x->value, (void*)expected, (void*)desired); \ } \ MA_ATOMIC_SAFE_TYPE_IMPL(32, uint32) MA_ATOMIC_SAFE_TYPE_IMPL(i32, int32) MA_ATOMIC_SAFE_TYPE_IMPL(64, uint64) MA_ATOMIC_SAFE_TYPE_IMPL(f32, float) MA_ATOMIC_SAFE_TYPE_IMPL(32, bool32) #if !defined(MA_NO_DEVICE_IO) MA_ATOMIC_SAFE_TYPE_IMPL(i32, device_state) #endif MA_API ma_uint64 ma_calculate_frame_count_after_resampling(ma_uint32 sampleRateOut, ma_uint32 sampleRateIn, ma_uint64 frameCountIn) { /* This is based on the calculation in ma_linear_resampler_get_expected_output_frame_count(). */ ma_uint64 outputFrameCount; ma_uint64 preliminaryInputFrameCountFromFrac; ma_uint64 preliminaryInputFrameCount; if (sampleRateIn == 0 || sampleRateOut == 0 || frameCountIn == 0) { return 0; } if (sampleRateOut == sampleRateIn) { return frameCountIn; } outputFrameCount = (frameCountIn * sampleRateOut) / sampleRateIn; preliminaryInputFrameCountFromFrac = (outputFrameCount * (sampleRateIn / sampleRateOut)) / sampleRateOut; preliminaryInputFrameCount = (outputFrameCount * (sampleRateIn % sampleRateOut)) + preliminaryInputFrameCountFromFrac; if (preliminaryInputFrameCount <= frameCountIn) { outputFrameCount += 1; } return outputFrameCount; } #ifndef MA_DATA_CONVERTER_STACK_BUFFER_SIZE #define MA_DATA_CONVERTER_STACK_BUFFER_SIZE 4096 #endif #if defined(MA_WIN32) static ma_result ma_result_from_GetLastError(DWORD error) { switch (error) { case ERROR_SUCCESS: return MA_SUCCESS; case ERROR_PATH_NOT_FOUND: return MA_DOES_NOT_EXIST; case ERROR_TOO_MANY_OPEN_FILES: return MA_TOO_MANY_OPEN_FILES; case ERROR_NOT_ENOUGH_MEMORY: return MA_OUT_OF_MEMORY; case ERROR_DISK_FULL: return MA_NO_SPACE; case ERROR_HANDLE_EOF: return MA_AT_END; case ERROR_NEGATIVE_SEEK: return MA_BAD_SEEK; case ERROR_INVALID_PARAMETER: return MA_INVALID_ARGS; case ERROR_ACCESS_DENIED: return MA_ACCESS_DENIED; case ERROR_SEM_TIMEOUT: return MA_TIMEOUT; case ERROR_FILE_NOT_FOUND: return MA_DOES_NOT_EXIST; default: break; } return MA_ERROR; } #endif /* MA_WIN32 */ /******************************************************************************* Threading *******************************************************************************/ static MA_INLINE ma_result ma_spinlock_lock_ex(volatile ma_spinlock* pSpinlock, ma_bool32 yield) { if (pSpinlock == NULL) { return MA_INVALID_ARGS; } for (;;) { if (ma_atomic_exchange_explicit_32(pSpinlock, 1, ma_atomic_memory_order_acquire) == 0) { break; } while (ma_atomic_load_explicit_32(pSpinlock, ma_atomic_memory_order_relaxed) == 1) { if (yield) { ma_yield(); } } } return MA_SUCCESS; } MA_API ma_result ma_spinlock_lock(volatile ma_spinlock* pSpinlock) { return ma_spinlock_lock_ex(pSpinlock, MA_TRUE); } MA_API ma_result ma_spinlock_lock_noyield(volatile ma_spinlock* pSpinlock) { return ma_spinlock_lock_ex(pSpinlock, MA_FALSE); } MA_API ma_result ma_spinlock_unlock(volatile ma_spinlock* pSpinlock) { if (pSpinlock == NULL) { return MA_INVALID_ARGS; } ma_atomic_store_explicit_32(pSpinlock, 0, ma_atomic_memory_order_release); return MA_SUCCESS; } #ifndef MA_NO_THREADING #if defined(MA_POSIX) #define MA_THREADCALL typedef void* ma_thread_result; #elif defined(MA_WIN32) #define MA_THREADCALL WINAPI typedef unsigned long ma_thread_result; #endif typedef ma_thread_result (MA_THREADCALL * ma_thread_entry_proc)(void* pData); #ifdef MA_POSIX static ma_result ma_thread_create__posix(ma_thread* pThread, ma_thread_priority priority, size_t stackSize, ma_thread_entry_proc entryProc, void* pData) { int result; pthread_attr_t* pAttr = NULL; #if !defined(MA_EMSCRIPTEN) && !defined(MA_3DS) && !defined(MA_SWITCH) /* Try setting the thread priority. It's not critical if anything fails here. */ pthread_attr_t attr; if (pthread_attr_init(&attr) == 0) { int scheduler = -1; /* We successfully initialized our attributes object so we can assign the pointer so it's passed into pthread_create(). */ pAttr = &attr; /* We need to set the scheduler policy. Only do this if the OS supports pthread_attr_setschedpolicy() */ #if !defined(MA_BEOS) { if (priority == ma_thread_priority_idle) { #ifdef SCHED_IDLE if (pthread_attr_setschedpolicy(&attr, SCHED_IDLE) == 0) { scheduler = SCHED_IDLE; } #endif } else if (priority == ma_thread_priority_realtime) { #ifdef SCHED_FIFO if (pthread_attr_setschedpolicy(&attr, SCHED_FIFO) == 0) { scheduler = SCHED_FIFO; } #endif #ifdef MA_LINUX } else { scheduler = sched_getscheduler(0); #endif } } #endif #if defined(_POSIX_THREAD_ATTR_STACKSIZE) && _POSIX_THREAD_ATTR_STACKSIZE >= 0 { if (stackSize > 0) { pthread_attr_setstacksize(&attr, stackSize); } } #else { (void)stackSize; /* Suppress unused parameter warning. */ } #endif if (scheduler != -1) { int priorityMin = sched_get_priority_min(scheduler); int priorityMax = sched_get_priority_max(scheduler); int priorityStep = (priorityMax - priorityMin) / 7; /* 7 = number of priorities supported by miniaudio. */ struct sched_param sched; if (priorityMin != -1 && priorityMax != -1 && pthread_attr_getschedparam(&attr, &sched) == 0) { if (priority == ma_thread_priority_idle) { sched.sched_priority = priorityMin; } else if (priority == ma_thread_priority_realtime) { #if defined(MA_PTHREAD_REALTIME_THREAD_PRIORITY) { sched.sched_priority = MA_PTHREAD_REALTIME_THREAD_PRIORITY; } #else { sched.sched_priority = priorityMax; } #endif } else { sched.sched_priority += ((int)priority + 5) * priorityStep; /* +5 because the lowest priority is -5. */ } if (sched.sched_priority < priorityMin) { sched.sched_priority = priorityMin; } if (sched.sched_priority > priorityMax) { sched.sched_priority = priorityMax; } /* I'm not treating a failure of setting the priority as a critical error so not aborting on failure here. */ if (pthread_attr_setschedparam(&attr, &sched) == 0) { #if !defined(MA_ANDROID) || (defined(__ANDROID_API__) && __ANDROID_API__ >= 28) { pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED); } #endif } } } } #else /* It's the emscripten build. We'll have a few unused parameters. */ (void)priority; (void)stackSize; #endif result = pthread_create((pthread_t*)pThread, pAttr, entryProc, pData); /* The thread attributes object is no longer required. */ if (pAttr != NULL) { pthread_attr_destroy(pAttr); } if (result != 0) { /* There have been reports that attempting to create a realtime thread can sometimes fail. In this case, fall back to a normal priority thread. I'm including a compile-time option here to disable this functionality for those who have a hard requirement on realtime threads and would rather an explicit failure. */ #ifndef MA_NO_PTHREAD_REALTIME_PRIORITY_FALLBACK { if(result == EPERM && priority == ma_thread_priority_realtime) { return ma_thread_create__posix(pThread, ma_thread_priority_normal, stackSize, entryProc, pData); } } #endif return ma_result_from_errno(result); } return MA_SUCCESS; } static void ma_thread_wait__posix(ma_thread* pThread) { pthread_join((pthread_t)*pThread, NULL); } static ma_result ma_mutex_init__posix(ma_mutex* pMutex) { int result; if (pMutex == NULL) { return MA_INVALID_ARGS; } MA_ZERO_OBJECT(pMutex); result = pthread_mutex_init((pthread_mutex_t*)pMutex, NULL); if (result != 0) { return ma_result_from_errno(result); } return MA_SUCCESS; } static void ma_mutex_uninit__posix(ma_mutex* pMutex) { pthread_mutex_destroy((pthread_mutex_t*)pMutex); } static void ma_mutex_lock__posix(ma_mutex* pMutex) { pthread_mutex_lock((pthread_mutex_t*)pMutex); } static void ma_mutex_unlock__posix(ma_mutex* pMutex) { pthread_mutex_unlock((pthread_mutex_t*)pMutex); } static ma_result ma_event_init__posix(ma_event* pEvent) { int result; result = pthread_mutex_init((pthread_mutex_t*)&pEvent->lock, NULL); if (result != 0) { return ma_result_from_errno(result); } result = pthread_cond_init((pthread_cond_t*)&pEvent->cond, NULL); if (result != 0) { pthread_mutex_destroy((pthread_mutex_t*)&pEvent->lock); return ma_result_from_errno(result); } pEvent->value = 0; return MA_SUCCESS; } static void ma_event_uninit__posix(ma_event* pEvent) { pthread_cond_destroy((pthread_cond_t*)&pEvent->cond); pthread_mutex_destroy((pthread_mutex_t*)&pEvent->lock); } static ma_result ma_event_wait__posix(ma_event* pEvent) { pthread_mutex_lock((pthread_mutex_t*)&pEvent->lock); { while (pEvent->value == 0) { pthread_cond_wait((pthread_cond_t*)&pEvent->cond, (pthread_mutex_t*)&pEvent->lock); } pEvent->value = 0; /* Auto-reset. */ } pthread_mutex_unlock((pthread_mutex_t*)&pEvent->lock); return MA_SUCCESS; } static ma_result ma_event_signal__posix(ma_event* pEvent) { pthread_mutex_lock((pthread_mutex_t*)&pEvent->lock); { pEvent->value = 1; pthread_cond_signal((pthread_cond_t*)&pEvent->cond); } pthread_mutex_unlock((pthread_mutex_t*)&pEvent->lock); return MA_SUCCESS; } static ma_result ma_semaphore_init__posix(int initialValue, ma_semaphore* pSemaphore) { int result; if (pSemaphore == NULL) { return MA_INVALID_ARGS; } pSemaphore->value = initialValue; result = pthread_mutex_init((pthread_mutex_t*)&pSemaphore->lock, NULL); if (result != 0) { return ma_result_from_errno(result); /* Failed to create mutex. */ } result = pthread_cond_init((pthread_cond_t*)&pSemaphore->cond, NULL); if (result != 0) { pthread_mutex_destroy((pthread_mutex_t*)&pSemaphore->lock); return ma_result_from_errno(result); /* Failed to create condition variable. */ } return MA_SUCCESS; } static void ma_semaphore_uninit__posix(ma_semaphore* pSemaphore) { if (pSemaphore == NULL) { return; } pthread_cond_destroy((pthread_cond_t*)&pSemaphore->cond); pthread_mutex_destroy((pthread_mutex_t*)&pSemaphore->lock); } static ma_result ma_semaphore_wait__posix(ma_semaphore* pSemaphore) { if (pSemaphore == NULL) { return MA_INVALID_ARGS; } pthread_mutex_lock((pthread_mutex_t*)&pSemaphore->lock); { /* We need to wait on a condition variable before escaping. We can't return from this function until the semaphore has been signaled. */ while (pSemaphore->value == 0) { pthread_cond_wait((pthread_cond_t*)&pSemaphore->cond, (pthread_mutex_t*)&pSemaphore->lock); } pSemaphore->value -= 1; } pthread_mutex_unlock((pthread_mutex_t*)&pSemaphore->lock); return MA_SUCCESS; } static ma_result ma_semaphore_release__posix(ma_semaphore* pSemaphore) { if (pSemaphore == NULL) { return MA_INVALID_ARGS; } pthread_mutex_lock((pthread_mutex_t*)&pSemaphore->lock); { pSemaphore->value += 1; pthread_cond_signal((pthread_cond_t*)&pSemaphore->cond); } pthread_mutex_unlock((pthread_mutex_t*)&pSemaphore->lock); return MA_SUCCESS; } #elif defined(MA_WIN32) static int ma_thread_priority_to_win32(ma_thread_priority priority) { switch (priority) { case ma_thread_priority_idle: return THREAD_PRIORITY_IDLE; case ma_thread_priority_lowest: return THREAD_PRIORITY_LOWEST; case ma_thread_priority_low: return THREAD_PRIORITY_BELOW_NORMAL; case ma_thread_priority_normal: return THREAD_PRIORITY_NORMAL; case ma_thread_priority_high: return THREAD_PRIORITY_ABOVE_NORMAL; case ma_thread_priority_highest: return THREAD_PRIORITY_HIGHEST; case ma_thread_priority_realtime: return THREAD_PRIORITY_TIME_CRITICAL; default: return THREAD_PRIORITY_NORMAL; } } static ma_result ma_thread_create__win32(ma_thread* pThread, ma_thread_priority priority, size_t stackSize, ma_thread_entry_proc entryProc, void* pData) { DWORD threadID; /* Not used. Only used for passing into CreateThread() so it doesn't fail on Windows 98. */ *pThread = CreateThread(NULL, stackSize, entryProc, pData, 0, &threadID); if (*pThread == NULL) { return ma_result_from_GetLastError(GetLastError()); } SetThreadPriority((HANDLE)*pThread, ma_thread_priority_to_win32(priority)); return MA_SUCCESS; } static void ma_thread_wait__win32(ma_thread* pThread) { WaitForSingleObject((HANDLE)*pThread, INFINITE); CloseHandle((HANDLE)*pThread); } static ma_result ma_mutex_init__win32(ma_mutex* pMutex) { *pMutex = CreateEventA(NULL, FALSE, TRUE, NULL); if (*pMutex == NULL) { return ma_result_from_GetLastError(GetLastError()); } return MA_SUCCESS; } static void ma_mutex_uninit__win32(ma_mutex* pMutex) { CloseHandle((HANDLE)*pMutex); } static void ma_mutex_lock__win32(ma_mutex* pMutex) { WaitForSingleObject((HANDLE)*pMutex, INFINITE); } static void ma_mutex_unlock__win32(ma_mutex* pMutex) { SetEvent((HANDLE)*pMutex); } static ma_result ma_event_init__win32(ma_event* pEvent) { *pEvent = CreateEventA(NULL, FALSE, FALSE, NULL); if (*pEvent == NULL) { return ma_result_from_GetLastError(GetLastError()); } return MA_SUCCESS; } static void ma_event_uninit__win32(ma_event* pEvent) { CloseHandle((HANDLE)*pEvent); } static ma_result ma_event_wait__win32(ma_event* pEvent) { DWORD result = WaitForSingleObject((HANDLE)*pEvent, INFINITE); if (result == WAIT_OBJECT_0) { return MA_SUCCESS; } if (result == WAIT_TIMEOUT) { return MA_TIMEOUT; } return ma_result_from_GetLastError(GetLastError()); } static ma_result ma_event_signal__win32(ma_event* pEvent) { BOOL result = SetEvent((HANDLE)*pEvent); if (result == 0) { return ma_result_from_GetLastError(GetLastError()); } return MA_SUCCESS; } static ma_result ma_semaphore_init__win32(int initialValue, ma_semaphore* pSemaphore) { *pSemaphore = CreateSemaphore(NULL, (LONG)initialValue, LONG_MAX, NULL); if (*pSemaphore == NULL) { return ma_result_from_GetLastError(GetLastError()); } return MA_SUCCESS; } static void ma_semaphore_uninit__win32(ma_semaphore* pSemaphore) { CloseHandle((HANDLE)*pSemaphore); } static ma_result ma_semaphore_wait__win32(ma_semaphore* pSemaphore) { DWORD result = WaitForSingleObject((HANDLE)*pSemaphore, INFINITE); if (result == WAIT_OBJECT_0) { return MA_SUCCESS; } if (result == WAIT_TIMEOUT) { return MA_TIMEOUT; } return ma_result_from_GetLastError(GetLastError()); } static ma_result ma_semaphore_release__win32(ma_semaphore* pSemaphore) { BOOL result = ReleaseSemaphore((HANDLE)*pSemaphore, 1, NULL); if (result == 0) { return ma_result_from_GetLastError(GetLastError()); } return MA_SUCCESS; } #endif typedef struct { ma_thread_entry_proc entryProc; void* pData; ma_allocation_callbacks allocationCallbacks; } ma_thread_proxy_data; static ma_thread_result MA_THREADCALL ma_thread_entry_proxy(void* pData) { ma_thread_proxy_data* pProxyData = (ma_thread_proxy_data*)pData; ma_thread_entry_proc entryProc; void* pEntryProcData; ma_thread_result result; #if defined(MA_ON_THREAD_ENTRY) MA_ON_THREAD_ENTRY #endif entryProc = pProxyData->entryProc; pEntryProcData = pProxyData->pData; /* Free the proxy data before getting into the real thread entry proc. */ ma_free(pProxyData, &pProxyData->allocationCallbacks); result = entryProc(pEntryProcData); #if defined(MA_ON_THREAD_EXIT) MA_ON_THREAD_EXIT #endif return result; } static ma_result ma_thread_create(ma_thread* pThread, ma_thread_priority priority, size_t stackSize, ma_thread_entry_proc entryProc, void* pData, const ma_allocation_callbacks* pAllocationCallbacks) { ma_result result; ma_thread_proxy_data* pProxyData; if (pThread == NULL || entryProc == NULL) { return MA_INVALID_ARGS; } pProxyData = (ma_thread_proxy_data*)ma_malloc(sizeof(*pProxyData), pAllocationCallbacks); /* Will be freed by the proxy entry proc. */ if (pProxyData == NULL) { return MA_OUT_OF_MEMORY; } #if defined(MA_THREAD_DEFAULT_STACK_SIZE) if (stackSize == 0) { stackSize = MA_THREAD_DEFAULT_STACK_SIZE; } #endif pProxyData->entryProc = entryProc; pProxyData->pData = pData; ma_allocation_callbacks_init_copy(&pProxyData->allocationCallbacks, pAllocationCallbacks); #if defined(MA_POSIX) result = ma_thread_create__posix(pThread, priority, stackSize, ma_thread_entry_proxy, pProxyData); #elif defined(MA_WIN32) result = ma_thread_create__win32(pThread, priority, stackSize, ma_thread_entry_proxy, pProxyData); #endif if (result != MA_SUCCESS) { ma_free(pProxyData, pAllocationCallbacks); return result; } return MA_SUCCESS; } static void ma_thread_wait(ma_thread* pThread) { if (pThread == NULL) { return; } #if defined(MA_POSIX) ma_thread_wait__posix(pThread); #elif defined(MA_WIN32) ma_thread_wait__win32(pThread); #endif } MA_API ma_result ma_mutex_init(ma_mutex* pMutex) { if (pMutex == NULL) { MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */ return MA_INVALID_ARGS; } #if defined(MA_POSIX) return ma_mutex_init__posix(pMutex); #elif defined(MA_WIN32) return ma_mutex_init__win32(pMutex); #endif } MA_API void ma_mutex_uninit(ma_mutex* pMutex) { if (pMutex == NULL) { return; } #if defined(MA_POSIX) ma_mutex_uninit__posix(pMutex); #elif defined(MA_WIN32) ma_mutex_uninit__win32(pMutex); #endif } MA_API void ma_mutex_lock(ma_mutex* pMutex) { if (pMutex == NULL) { MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */ return; } #if defined(MA_POSIX) ma_mutex_lock__posix(pMutex); #elif defined(MA_WIN32) ma_mutex_lock__win32(pMutex); #endif } MA_API void ma_mutex_unlock(ma_mutex* pMutex) { if (pMutex == NULL) { MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */ return; } #if defined(MA_POSIX) ma_mutex_unlock__posix(pMutex); #elif defined(MA_WIN32) ma_mutex_unlock__win32(pMutex); #endif } MA_API ma_result ma_event_init(ma_event* pEvent) { if (pEvent == NULL) { MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */ return MA_INVALID_ARGS; } #if defined(MA_POSIX) return ma_event_init__posix(pEvent); #elif defined(MA_WIN32) return ma_event_init__win32(pEvent); #endif } #if 0 static ma_result ma_event_alloc_and_init(ma_event** ppEvent, ma_allocation_callbacks* pAllocationCallbacks) { ma_result result; ma_event* pEvent; if (ppEvent == NULL) { return MA_INVALID_ARGS; } *ppEvent = NULL; pEvent = ma_malloc(sizeof(*pEvent), pAllocationCallbacks); if (pEvent == NULL) { return MA_OUT_OF_MEMORY; } result = ma_event_init(pEvent); if (result != MA_SUCCESS) { ma_free(pEvent, pAllocationCallbacks); return result; } *ppEvent = pEvent; return result; } #endif MA_API void ma_event_uninit(ma_event* pEvent) { if (pEvent == NULL) { return; } #if defined(MA_POSIX) ma_event_uninit__posix(pEvent); #elif defined(MA_WIN32) ma_event_uninit__win32(pEvent); #endif } #if 0 static void ma_event_uninit_and_free(ma_event* pEvent, ma_allocation_callbacks* pAllocationCallbacks) { if (pEvent == NULL) { return; } ma_event_uninit(pEvent); ma_free(pEvent, pAllocationCallbacks); } #endif MA_API ma_result ma_event_wait(ma_event* pEvent) { if (pEvent == NULL) { MA_ASSERT(MA_FALSE); /* Fire an assert to the caller is aware of this bug. */ return MA_INVALID_ARGS; } #if defined(MA_POSIX) return ma_event_wait__posix(pEvent); #elif defined(MA_WIN32) return ma_event_wait__win32(pEvent); #endif } MA_API ma_result ma_event_signal(ma_event* pEvent) { if (pEvent == NULL) { MA_ASSERT(MA_FALSE); /* Fire an assert to the caller is aware of this bug. */ return MA_INVALID_ARGS; } #if defined(MA_POSIX) return ma_event_signal__posix(pEvent); #elif defined(MA_WIN32) return ma_event_signal__win32(pEvent); #endif } MA_API ma_result ma_semaphore_init(int initialValue, ma_semaphore* pSemaphore) { if (pSemaphore == NULL) { MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */ return MA_INVALID_ARGS; } #if defined(MA_POSIX) return ma_semaphore_init__posix(initialValue, pSemaphore); #elif defined(MA_WIN32) return ma_semaphore_init__win32(initialValue, pSemaphore); #endif } MA_API void ma_semaphore_uninit(ma_semaphore* pSemaphore) { if (pSemaphore == NULL) { MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */ return; } #if defined(MA_POSIX) ma_semaphore_uninit__posix(pSemaphore); #elif defined(MA_WIN32) ma_semaphore_uninit__win32(pSemaphore); #endif } MA_API ma_result ma_semaphore_wait(ma_semaphore* pSemaphore) { if (pSemaphore == NULL) { MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */ return MA_INVALID_ARGS; } #if defined(MA_POSIX) return ma_semaphore_wait__posix(pSemaphore); #elif defined(MA_WIN32) return ma_semaphore_wait__win32(pSemaphore); #endif } MA_API ma_result ma_semaphore_release(ma_semaphore* pSemaphore) { if (pSemaphore == NULL) { MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */ return MA_INVALID_ARGS; } #if defined(MA_POSIX) return ma_semaphore_release__posix(pSemaphore); #elif defined(MA_WIN32) return ma_semaphore_release__win32(pSemaphore); #endif } #else /* MA_NO_THREADING is set which means threading is disabled. Threading is required by some API families. If any of these are enabled we need to throw an error. */ #ifndef MA_NO_DEVICE_IO #error "MA_NO_THREADING cannot be used without MA_NO_DEVICE_IO"; #endif #endif /* MA_NO_THREADING */ #define MA_FENCE_COUNTER_MAX 0x7FFFFFFF MA_API ma_result ma_fence_init(ma_fence* pFence) { if (pFence == NULL) { return MA_INVALID_ARGS; } MA_ZERO_OBJECT(pFence); pFence->counter = 0; #ifndef MA_NO_THREADING { ma_result result; result = ma_event_init(&pFence->e); if (result != MA_SUCCESS) { return result; } } #endif return MA_SUCCESS; } MA_API void ma_fence_uninit(ma_fence* pFence) { if (pFence == NULL) { return; } #ifndef MA_NO_THREADING { ma_event_uninit(&pFence->e); } #endif MA_ZERO_OBJECT(pFence); } MA_API ma_result ma_fence_acquire(ma_fence* pFence) { if (pFence == NULL) { return MA_INVALID_ARGS; } for (;;) { ma_uint32 oldCounter = ma_atomic_load_32(&pFence->counter); ma_uint32 newCounter = oldCounter + 1; /* Make sure we're not about to exceed our maximum value. */ if (newCounter > MA_FENCE_COUNTER_MAX) { MA_ASSERT(MA_FALSE); return MA_OUT_OF_RANGE; } if (ma_atomic_compare_exchange_weak_32(&pFence->counter, &oldCounter, newCounter)) { return MA_SUCCESS; } else { if (oldCounter == MA_FENCE_COUNTER_MAX) { MA_ASSERT(MA_FALSE); return MA_OUT_OF_RANGE; /* The other thread took the last available slot. Abort. */ } } } /* Should never get here. */ /*return MA_SUCCESS;*/ } MA_API ma_result ma_fence_release(ma_fence* pFence) { if (pFence == NULL) { return MA_INVALID_ARGS; } for (;;) { ma_uint32 oldCounter = ma_atomic_load_32(&pFence->counter); ma_uint32 newCounter = oldCounter - 1; if (oldCounter == 0) { MA_ASSERT(MA_FALSE); return MA_INVALID_OPERATION; /* Acquire/release mismatch. */ } if (ma_atomic_compare_exchange_weak_32(&pFence->counter, &oldCounter, newCounter)) { #ifndef MA_NO_THREADING { if (newCounter == 0) { ma_event_signal(&pFence->e); /* <-- ma_fence_wait() will be waiting on this. */ } } #endif return MA_SUCCESS; } else { if (oldCounter == 0) { MA_ASSERT(MA_FALSE); return MA_INVALID_OPERATION; /* Another thread has taken the 0 slot. Acquire/release mismatch. */ } } } /* Should never get here. */ /*return MA_SUCCESS;*/ } MA_API ma_result ma_fence_wait(ma_fence* pFence) { if (pFence == NULL) { return MA_INVALID_ARGS; } for (;;) { ma_uint32 counter; counter = ma_atomic_load_32(&pFence->counter); if (counter == 0) { /* Counter has hit zero. By the time we get here some other thread may have acquired the fence again, but that is where the caller needs to take care with how they se the fence. */ return MA_SUCCESS; } /* Getting here means the counter is > 0. We'll need to wait for something to happen. */ #ifndef MA_NO_THREADING { ma_result result; result = ma_event_wait(&pFence->e); if (result != MA_SUCCESS) { return result; } } #endif } /* Should never get here. */ /*return MA_INVALID_OPERATION;*/ } MA_API ma_result ma_async_notification_signal(ma_async_notification* pNotification) { ma_async_notification_callbacks* pNotificationCallbacks = (ma_async_notification_callbacks*)pNotification; if (pNotification == NULL) { return MA_INVALID_ARGS; } if (pNotificationCallbacks->onSignal == NULL) { return MA_NOT_IMPLEMENTED; } pNotificationCallbacks->onSignal(pNotification); return MA_INVALID_ARGS; } static void ma_async_notification_poll__on_signal(ma_async_notification* pNotification) { ((ma_async_notification_poll*)pNotification)->signalled = MA_TRUE; } MA_API ma_result ma_async_notification_poll_init(ma_async_notification_poll* pNotificationPoll) { if (pNotificationPoll == NULL) { return MA_INVALID_ARGS; } pNotificationPoll->cb.onSignal = ma_async_notification_poll__on_signal; pNotificationPoll->signalled = MA_FALSE; return MA_SUCCESS; } MA_API ma_bool32 ma_async_notification_poll_is_signalled(const ma_async_notification_poll* pNotificationPoll) { if (pNotificationPoll == NULL) { return MA_FALSE; } return pNotificationPoll->signalled; } static void ma_async_notification_event__on_signal(ma_async_notification* pNotification) { ma_async_notification_event_signal((ma_async_notification_event*)pNotification); } MA_API ma_result ma_async_notification_event_init(ma_async_notification_event* pNotificationEvent) { if (pNotificationEvent == NULL) { return MA_INVALID_ARGS; } pNotificationEvent->cb.onSignal = ma_async_notification_event__on_signal; #ifndef MA_NO_THREADING { ma_result result; result = ma_event_init(&pNotificationEvent->e); if (result != MA_SUCCESS) { return result; } return MA_SUCCESS; } #else { return MA_NOT_IMPLEMENTED; /* Threading is disabled. */ } #endif } MA_API ma_result ma_async_notification_event_uninit(ma_async_notification_event* pNotificationEvent) { if (pNotificationEvent == NULL) { return MA_INVALID_ARGS; } #ifndef MA_NO_THREADING { ma_event_uninit(&pNotificationEvent->e); return MA_SUCCESS; } #else { return MA_NOT_IMPLEMENTED; /* Threading is disabled. */ } #endif } MA_API ma_result ma_async_notification_event_wait(ma_async_notification_event* pNotificationEvent) { if (pNotificationEvent == NULL) { return MA_INVALID_ARGS; } #ifndef MA_NO_THREADING { return ma_event_wait(&pNotificationEvent->e); } #else { return MA_NOT_IMPLEMENTED; /* Threading is disabled. */ } #endif } MA_API ma_result ma_async_notification_event_signal(ma_async_notification_event* pNotificationEvent) { if (pNotificationEvent == NULL) { return MA_INVALID_ARGS; } #ifndef MA_NO_THREADING { return ma_event_signal(&pNotificationEvent->e); } #else { return MA_NOT_IMPLEMENTED; /* Threading is disabled. */ } #endif } /************************************************************************************************************************************************************ Job Queue ************************************************************************************************************************************************************/ MA_API ma_slot_allocator_config ma_slot_allocator_config_init(ma_uint32 capacity) { ma_slot_allocator_config config; MA_ZERO_OBJECT(&config); config.capacity = capacity; return config; } static MA_INLINE ma_uint32 ma_slot_allocator_calculate_group_capacity(ma_uint32 slotCapacity) { ma_uint32 cap = slotCapacity / 32; if ((slotCapacity % 32) != 0) { cap += 1; } return cap; } static MA_INLINE ma_uint32 ma_slot_allocator_group_capacity(const ma_slot_allocator* pAllocator) { return ma_slot_allocator_calculate_group_capacity(pAllocator->capacity); } typedef struct { size_t sizeInBytes; size_t groupsOffset; size_t slotsOffset; } ma_slot_allocator_heap_layout; static ma_result ma_slot_allocator_get_heap_layout(const ma_slot_allocator_config* pConfig, ma_slot_allocator_heap_layout* pHeapLayout) { MA_ASSERT(pHeapLayout != NULL); MA_ZERO_OBJECT(pHeapLayout); if (pConfig == NULL) { return MA_INVALID_ARGS; } if (pConfig->capacity == 0) { return MA_INVALID_ARGS; } pHeapLayout->sizeInBytes = 0; /* Groups. */ pHeapLayout->groupsOffset = pHeapLayout->sizeInBytes; pHeapLayout->sizeInBytes += ma_align_64(ma_slot_allocator_calculate_group_capacity(pConfig->capacity) * sizeof(ma_slot_allocator_group)); /* Slots. */ pHeapLayout->slotsOffset = pHeapLayout->sizeInBytes; pHeapLayout->sizeInBytes += ma_align_64(pConfig->capacity * sizeof(ma_uint32)); return MA_SUCCESS; } MA_API ma_result ma_slot_allocator_get_heap_size(const ma_slot_allocator_config* pConfig, size_t* pHeapSizeInBytes) { ma_result result; ma_slot_allocator_heap_layout layout; if (pHeapSizeInBytes == NULL) { return MA_INVALID_ARGS; } *pHeapSizeInBytes = 0; result = ma_slot_allocator_get_heap_layout(pConfig, &layout); if (result != MA_SUCCESS) { return result; } *pHeapSizeInBytes = layout.sizeInBytes; return result; } MA_API ma_result ma_slot_allocator_init_preallocated(const ma_slot_allocator_config* pConfig, void* pHeap, ma_slot_allocator* pAllocator) { ma_result result; ma_slot_allocator_heap_layout heapLayout; if (pAllocator == NULL) { return MA_INVALID_ARGS; } MA_ZERO_OBJECT(pAllocator); if (pHeap == NULL) { return MA_INVALID_ARGS; } result = ma_slot_allocator_get_heap_layout(pConfig, &heapLayout); if (result != MA_SUCCESS) { return result; } pAllocator->_pHeap = pHeap; MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes); pAllocator->pGroups = (ma_slot_allocator_group*)ma_offset_ptr(pHeap, heapLayout.groupsOffset); pAllocator->pSlots = (ma_uint32*)ma_offset_ptr(pHeap, heapLayout.slotsOffset); pAllocator->capacity = pConfig->capacity; return MA_SUCCESS; } MA_API ma_result ma_slot_allocator_init(const ma_slot_allocator_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_slot_allocator* pAllocator) { ma_result result; size_t heapSizeInBytes; void* pHeap; result = ma_slot_allocator_get_heap_size(pConfig, &heapSizeInBytes); if (result != MA_SUCCESS) { return result; /* Failed to retrieve the size of the heap allocation. */ } if (heapSizeInBytes > 0) { pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks); if (pHeap == NULL) { return MA_OUT_OF_MEMORY; } } else { pHeap = NULL; } result = ma_slot_allocator_init_preallocated(pConfig, pHeap, pAllocator); if (result != MA_SUCCESS) { ma_free(pHeap, pAllocationCallbacks); return result; } pAllocator->_ownsHeap = MA_TRUE; return MA_SUCCESS; } MA_API void ma_slot_allocator_uninit(ma_slot_allocator* pAllocator, const ma_allocation_callbacks* pAllocationCallbacks) { if (pAllocator == NULL) { return; } if (pAllocator->_ownsHeap) { ma_free(pAllocator->_pHeap, pAllocationCallbacks); } } MA_API ma_result ma_slot_allocator_alloc(ma_slot_allocator* pAllocator, ma_uint64* pSlot) { ma_uint32 iAttempt; const ma_uint32 maxAttempts = 2; /* The number of iterations to perform until returning MA_OUT_OF_MEMORY if no slots can be found. */ if (pAllocator == NULL || pSlot == NULL) { return MA_INVALID_ARGS; } for (iAttempt = 0; iAttempt < maxAttempts; iAttempt += 1) { /* We need to acquire a suitable bitfield first. This is a bitfield that's got an available slot within it. */ ma_uint32 iGroup; for (iGroup = 0; iGroup < ma_slot_allocator_group_capacity(pAllocator); iGroup += 1) { /* CAS */ for (;;) { ma_uint32 oldBitfield; ma_uint32 newBitfield; ma_uint32 bitOffset; oldBitfield = ma_atomic_load_32(&pAllocator->pGroups[iGroup].bitfield); /* <-- This copy must happen. The compiler must not optimize this away. */ /* Fast check to see if anything is available. */ if (oldBitfield == 0xFFFFFFFF) { break; /* No available bits in this bitfield. */ } bitOffset = ma_ffs_32(~oldBitfield); MA_ASSERT(bitOffset < 32); newBitfield = oldBitfield | (1 << bitOffset); if (ma_atomic_compare_and_swap_32(&pAllocator->pGroups[iGroup].bitfield, oldBitfield, newBitfield) == oldBitfield) { ma_uint32 slotIndex; /* Increment the counter as soon as possible to have other threads report out-of-memory sooner than later. */ ma_atomic_fetch_add_32(&pAllocator->count, 1); /* The slot index is required for constructing the output value. */ slotIndex = (iGroup << 5) + bitOffset; /* iGroup << 5 = iGroup * 32 */ if (slotIndex >= pAllocator->capacity) { return MA_OUT_OF_MEMORY; } /* Increment the reference count before constructing the output value. */ pAllocator->pSlots[slotIndex] += 1; /* Construct the output value. */ *pSlot = (((ma_uint64)pAllocator->pSlots[slotIndex] << 32) | slotIndex); return MA_SUCCESS; } } } /* We weren't able to find a slot. If it's because we've reached our capacity we need to return MA_OUT_OF_MEMORY. Otherwise we need to do another iteration and try again. */ if (pAllocator->count < pAllocator->capacity) { ma_yield(); } else { return MA_OUT_OF_MEMORY; } } /* We couldn't find a slot within the maximum number of attempts. */ return MA_OUT_OF_MEMORY; } MA_API ma_result ma_slot_allocator_free(ma_slot_allocator* pAllocator, ma_uint64 slot) { ma_uint32 iGroup; ma_uint32 iBit; if (pAllocator == NULL) { return MA_INVALID_ARGS; } iGroup = (ma_uint32)((slot & 0xFFFFFFFF) >> 5); /* slot / 32 */ iBit = (ma_uint32)((slot & 0xFFFFFFFF) & 31); /* slot % 32 */ if (iGroup >= ma_slot_allocator_group_capacity(pAllocator)) { return MA_INVALID_ARGS; } MA_ASSERT(iBit < 32); /* This must be true due to the logic we used to actually calculate it. */ while (ma_atomic_load_32(&pAllocator->count) > 0) { /* CAS */ ma_uint32 oldBitfield; ma_uint32 newBitfield; oldBitfield = ma_atomic_load_32(&pAllocator->pGroups[iGroup].bitfield); /* <-- This copy must happen. The compiler must not optimize this away. */ newBitfield = oldBitfield & ~(1 << iBit); /* Debugging for checking for double-frees. */ #if defined(MA_DEBUG_OUTPUT) { if ((oldBitfield & (1 << iBit)) == 0) { MA_ASSERT(MA_FALSE); /* Double free detected.*/ } } #endif if (ma_atomic_compare_and_swap_32(&pAllocator->pGroups[iGroup].bitfield, oldBitfield, newBitfield) == oldBitfield) { ma_atomic_fetch_sub_32(&pAllocator->count, 1); return MA_SUCCESS; } } /* Getting here means there are no allocations available for freeing. */ return MA_INVALID_OPERATION; } #define MA_JOB_ID_NONE ~((ma_uint64)0) #define MA_JOB_SLOT_NONE (ma_uint16)(~0) static MA_INLINE ma_uint32 ma_job_extract_refcount(ma_uint64 toc) { return (ma_uint32)(toc >> 32); } static MA_INLINE ma_uint16 ma_job_extract_slot(ma_uint64 toc) { return (ma_uint16)(toc & 0x0000FFFF); } #if 0 /* Currently unused, but might make use of this later. */ static MA_INLINE ma_uint16 ma_job_extract_code(ma_uint64 toc) { return (ma_uint16)((toc & 0xFFFF0000) >> 16); } #endif static MA_INLINE ma_uint64 ma_job_toc_to_allocation(ma_uint64 toc) { return ((ma_uint64)ma_job_extract_refcount(toc) << 32) | (ma_uint64)ma_job_extract_slot(toc); } static MA_INLINE ma_uint64 ma_job_set_refcount(ma_uint64 toc, ma_uint32 refcount) { /* Clear the reference count first. */ toc = toc & ~((ma_uint64)0xFFFFFFFF << 32); toc = toc | ((ma_uint64)refcount << 32); return toc; } MA_API ma_job ma_job_init(ma_uint16 code) { ma_job job; MA_ZERO_OBJECT(&job); job.toc.breakup.code = code; job.toc.breakup.slot = MA_JOB_SLOT_NONE; /* Temp value. Will be allocated when posted to a queue. */ job.next = MA_JOB_ID_NONE; return job; } static ma_result ma_job_process__noop(ma_job* pJob); static ma_result ma_job_process__quit(ma_job* pJob); static ma_result ma_job_process__custom(ma_job* pJob); static ma_result ma_job_process__resource_manager__load_data_buffer_node(ma_job* pJob); static ma_result ma_job_process__resource_manager__free_data_buffer_node(ma_job* pJob); static ma_result ma_job_process__resource_manager__page_data_buffer_node(ma_job* pJob); static ma_result ma_job_process__resource_manager__load_data_buffer(ma_job* pJob); static ma_result ma_job_process__resource_manager__free_data_buffer(ma_job* pJob); static ma_result ma_job_process__resource_manager__load_data_stream(ma_job* pJob); static ma_result ma_job_process__resource_manager__free_data_stream(ma_job* pJob); static ma_result ma_job_process__resource_manager__page_data_stream(ma_job* pJob); static ma_result ma_job_process__resource_manager__seek_data_stream(ma_job* pJob); #if !defined(MA_NO_DEVICE_IO) static ma_result ma_job_process__device__aaudio_reroute(ma_job* pJob); #endif static ma_job_proc g_jobVTable[MA_JOB_TYPE_COUNT] = { /* Miscellaneous. */ ma_job_process__quit, /* MA_JOB_TYPE_QUIT */ ma_job_process__custom, /* MA_JOB_TYPE_CUSTOM */ /* Resource Manager. */ ma_job_process__resource_manager__load_data_buffer_node, /* MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER_NODE */ ma_job_process__resource_manager__free_data_buffer_node, /* MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER_NODE */ ma_job_process__resource_manager__page_data_buffer_node, /* MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE */ ma_job_process__resource_manager__load_data_buffer, /* MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER */ ma_job_process__resource_manager__free_data_buffer, /* MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER */ ma_job_process__resource_manager__load_data_stream, /* MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_STREAM */ ma_job_process__resource_manager__free_data_stream, /* MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_STREAM */ ma_job_process__resource_manager__page_data_stream, /* MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_STREAM */ ma_job_process__resource_manager__seek_data_stream, /* MA_JOB_TYPE_RESOURCE_MANAGER_SEEK_DATA_STREAM */ /* Device. */ #if !defined(MA_NO_DEVICE_IO) ma_job_process__device__aaudio_reroute /* MA_JOB_TYPE_DEVICE_AAUDIO_REROUTE */ #endif }; MA_API ma_result ma_job_process(ma_job* pJob) { if (pJob == NULL) { return MA_INVALID_ARGS; } if (pJob->toc.breakup.code >= MA_JOB_TYPE_COUNT) { return MA_INVALID_OPERATION; } return g_jobVTable[pJob->toc.breakup.code](pJob); } static ma_result ma_job_process__noop(ma_job* pJob) { MA_ASSERT(pJob != NULL); /* No-op. */ (void)pJob; return MA_SUCCESS; } static ma_result ma_job_process__quit(ma_job* pJob) { return ma_job_process__noop(pJob); } static ma_result ma_job_process__custom(ma_job* pJob) { MA_ASSERT(pJob != NULL); /* No-op if there's no callback. */ if (pJob->data.custom.proc == NULL) { return MA_SUCCESS; } return pJob->data.custom.proc(pJob); } MA_API ma_job_queue_config ma_job_queue_config_init(ma_uint32 flags, ma_uint32 capacity) { ma_job_queue_config config; config.flags = flags; config.capacity = capacity; return config; } typedef struct { size_t sizeInBytes; size_t allocatorOffset; size_t jobsOffset; } ma_job_queue_heap_layout; static ma_result ma_job_queue_get_heap_layout(const ma_job_queue_config* pConfig, ma_job_queue_heap_layout* pHeapLayout) { ma_result result; MA_ASSERT(pHeapLayout != NULL); MA_ZERO_OBJECT(pHeapLayout); if (pConfig == NULL) { return MA_INVALID_ARGS; } if (pConfig->capacity == 0) { return MA_INVALID_ARGS; } pHeapLayout->sizeInBytes = 0; /* Allocator. */ { ma_slot_allocator_config allocatorConfig; size_t allocatorHeapSizeInBytes; allocatorConfig = ma_slot_allocator_config_init(pConfig->capacity); result = ma_slot_allocator_get_heap_size(&allocatorConfig, &allocatorHeapSizeInBytes); if (result != MA_SUCCESS) { return result; } pHeapLayout->allocatorOffset = pHeapLayout->sizeInBytes; pHeapLayout->sizeInBytes += allocatorHeapSizeInBytes; } /* Jobs. */ pHeapLayout->jobsOffset = pHeapLayout->sizeInBytes; pHeapLayout->sizeInBytes += ma_align_64(pConfig->capacity * sizeof(ma_job)); return MA_SUCCESS; } MA_API ma_result ma_job_queue_get_heap_size(const ma_job_queue_config* pConfig, size_t* pHeapSizeInBytes) { ma_result result; ma_job_queue_heap_layout layout; if (pHeapSizeInBytes == NULL) { return MA_INVALID_ARGS; } *pHeapSizeInBytes = 0; result = ma_job_queue_get_heap_layout(pConfig, &layout); if (result != MA_SUCCESS) { return result; } *pHeapSizeInBytes = layout.sizeInBytes; return MA_SUCCESS; } MA_API ma_result ma_job_queue_init_preallocated(const ma_job_queue_config* pConfig, void* pHeap, ma_job_queue* pQueue) { ma_result result; ma_job_queue_heap_layout heapLayout; ma_slot_allocator_config allocatorConfig; if (pQueue == NULL) { return MA_INVALID_ARGS; } MA_ZERO_OBJECT(pQueue); result = ma_job_queue_get_heap_layout(pConfig, &heapLayout); if (result != MA_SUCCESS) { return result; } pQueue->_pHeap = pHeap; MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes); pQueue->flags = pConfig->flags; pQueue->capacity = pConfig->capacity; pQueue->pJobs = (ma_job*)ma_offset_ptr(pHeap, heapLayout.jobsOffset); allocatorConfig = ma_slot_allocator_config_init(pConfig->capacity); result = ma_slot_allocator_init_preallocated(&allocatorConfig, ma_offset_ptr(pHeap, heapLayout.allocatorOffset), &pQueue->allocator); if (result != MA_SUCCESS) { return result; } /* We need a semaphore if we're running in non-blocking mode. If threading is disabled we need to return an error. */ if ((pQueue->flags & MA_JOB_QUEUE_FLAG_NON_BLOCKING) == 0) { #ifndef MA_NO_THREADING { ma_semaphore_init(0, &pQueue->sem); } #else { /* Threading is disabled and we've requested non-blocking mode. */ return MA_INVALID_OPERATION; } #endif } /* Our queue needs to be initialized with a free standing node. This should always be slot 0. Required for the lock free algorithm. The first job in the queue is just a dummy item for giving us the first item in the list which is stored in the "next" member. */ ma_slot_allocator_alloc(&pQueue->allocator, &pQueue->head); /* Will never fail. */ pQueue->pJobs[ma_job_extract_slot(pQueue->head)].next = MA_JOB_ID_NONE; pQueue->tail = pQueue->head; return MA_SUCCESS; } MA_API ma_result ma_job_queue_init(const ma_job_queue_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_job_queue* pQueue) { ma_result result; size_t heapSizeInBytes; void* pHeap; result = ma_job_queue_get_heap_size(pConfig, &heapSizeInBytes); if (result != MA_SUCCESS) { return result; } if (heapSizeInBytes > 0) { pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks); if (pHeap == NULL) { return MA_OUT_OF_MEMORY; } } else { pHeap = NULL; } result = ma_job_queue_init_preallocated(pConfig, pHeap, pQueue); if (result != MA_SUCCESS) { ma_free(pHeap, pAllocationCallbacks); return result; } pQueue->_ownsHeap = MA_TRUE; return MA_SUCCESS; } MA_API void ma_job_queue_uninit(ma_job_queue* pQueue, const ma_allocation_callbacks* pAllocationCallbacks) { if (pQueue == NULL) { return; } /* All we need to do is uninitialize the semaphore. */ if ((pQueue->flags & MA_JOB_QUEUE_FLAG_NON_BLOCKING) == 0) { #ifndef MA_NO_THREADING { ma_semaphore_uninit(&pQueue->sem); } #else { MA_ASSERT(MA_FALSE); /* Should never get here. Should have been checked at initialization time. */ } #endif } ma_slot_allocator_uninit(&pQueue->allocator, pAllocationCallbacks); if (pQueue->_ownsHeap) { ma_free(pQueue->_pHeap, pAllocationCallbacks); } } static ma_bool32 ma_job_queue_cas(volatile ma_uint64* dst, ma_uint64 expected, ma_uint64 desired) { /* The new counter is taken from the expected value. */ return ma_atomic_compare_and_swap_64(dst, expected, ma_job_set_refcount(desired, ma_job_extract_refcount(expected) + 1)) == expected; } MA_API ma_result ma_job_queue_post(ma_job_queue* pQueue, const ma_job* pJob) { /* Lock free queue implementation based on the paper by Michael and Scott: Nonblocking Algorithms and Preemption-Safe Locking on Multiprogrammed Shared Memory Multiprocessors */ ma_result result; ma_uint64 slot; ma_uint64 tail; ma_uint64 next; if (pQueue == NULL || pJob == NULL) { return MA_INVALID_ARGS; } /* We need a new slot. */ result = ma_slot_allocator_alloc(&pQueue->allocator, &slot); if (result != MA_SUCCESS) { return result; /* Probably ran out of slots. If so, MA_OUT_OF_MEMORY will be returned. */ } /* At this point we should have a slot to place the job. */ MA_ASSERT(ma_job_extract_slot(slot) < pQueue->capacity); /* We need to put the job into memory before we do anything. */ pQueue->pJobs[ma_job_extract_slot(slot)] = *pJob; pQueue->pJobs[ma_job_extract_slot(slot)].toc.allocation = slot; /* This will overwrite the job code. */ pQueue->pJobs[ma_job_extract_slot(slot)].toc.breakup.code = pJob->toc.breakup.code; /* The job code needs to be applied again because the line above overwrote it. */ pQueue->pJobs[ma_job_extract_slot(slot)].next = MA_JOB_ID_NONE; /* Reset for safety. */ #ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE ma_spinlock_lock(&pQueue->lock); #endif { /* The job is stored in memory so now we need to add it to our linked list. We only ever add items to the end of the list. */ for (;;) { tail = ma_atomic_load_64(&pQueue->tail); next = ma_atomic_load_64(&pQueue->pJobs[ma_job_extract_slot(tail)].next); if (ma_job_toc_to_allocation(tail) == ma_job_toc_to_allocation(ma_atomic_load_64(&pQueue->tail))) { if (ma_job_extract_slot(next) == 0xFFFF) { if (ma_job_queue_cas(&pQueue->pJobs[ma_job_extract_slot(tail)].next, next, slot)) { break; } } else { ma_job_queue_cas(&pQueue->tail, tail, ma_job_extract_slot(next)); } } } ma_job_queue_cas(&pQueue->tail, tail, slot); } #ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE ma_spinlock_unlock(&pQueue->lock); #endif /* Signal the semaphore as the last step if we're using synchronous mode. */ if ((pQueue->flags & MA_JOB_QUEUE_FLAG_NON_BLOCKING) == 0) { #ifndef MA_NO_THREADING { ma_semaphore_release(&pQueue->sem); } #else { MA_ASSERT(MA_FALSE); /* Should never get here. Should have been checked at initialization time. */ } #endif } return MA_SUCCESS; } MA_API ma_result ma_job_queue_next(ma_job_queue* pQueue, ma_job* pJob) { ma_uint64 head; ma_uint64 tail; ma_uint64 next; if (pQueue == NULL || pJob == NULL) { return MA_INVALID_ARGS; } /* If we're running in synchronous mode we'll need to wait on a semaphore. */ if ((pQueue->flags & MA_JOB_QUEUE_FLAG_NON_BLOCKING) == 0) { #ifndef MA_NO_THREADING { ma_semaphore_wait(&pQueue->sem); } #else { MA_ASSERT(MA_FALSE); /* Should never get here. Should have been checked at initialization time. */ } #endif } #ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE ma_spinlock_lock(&pQueue->lock); #endif { /* BUG: In lock-free mode, multiple threads can be in this section of code. The "head" variable in the loop below is stored. One thread can fall through to the freeing of this item while another is still using "head" for the retrieval of the "next" variable. The slot allocator might need to make use of some reference counting to ensure it's only truly freed when there are no more references to the item. This must be fixed before removing these locks. */ /* Now we need to remove the root item from the list. */ for (;;) { head = ma_atomic_load_64(&pQueue->head); tail = ma_atomic_load_64(&pQueue->tail); next = ma_atomic_load_64(&pQueue->pJobs[ma_job_extract_slot(head)].next); if (ma_job_toc_to_allocation(head) == ma_job_toc_to_allocation(ma_atomic_load_64(&pQueue->head))) { if (ma_job_extract_slot(head) == ma_job_extract_slot(tail)) { if (ma_job_extract_slot(next) == 0xFFFF) { #ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE ma_spinlock_unlock(&pQueue->lock); #endif return MA_NO_DATA_AVAILABLE; } ma_job_queue_cas(&pQueue->tail, tail, ma_job_extract_slot(next)); } else { *pJob = pQueue->pJobs[ma_job_extract_slot(next)]; if (ma_job_queue_cas(&pQueue->head, head, ma_job_extract_slot(next))) { break; } } } } } #ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE ma_spinlock_unlock(&pQueue->lock); #endif ma_slot_allocator_free(&pQueue->allocator, head); /* If it's a quit job make sure it's put back on the queue to ensure other threads have an opportunity to detect it and terminate naturally. We could instead just leave it on the queue, but that would involve fiddling with the lock-free code above and I want to keep that as simple as possible. */ if (pJob->toc.breakup.code == MA_JOB_TYPE_QUIT) { ma_job_queue_post(pQueue, pJob); return MA_CANCELLED; /* Return a cancelled status just in case the thread is checking return codes and not properly checking for a quit job. */ } return MA_SUCCESS; } /******************************************************************************* Dynamic Linking *******************************************************************************/ /* Disable run-time linking on certain backends and platforms. */ #ifndef MA_NO_RUNTIME_LINKING #if defined(MA_EMSCRIPTEN) || defined(MA_ORBIS) || defined(MA_PROSPERO) || defined(MA_SWITCH) || defined(MA_DOS) #define MA_NO_RUNTIME_LINKING #endif #endif #ifdef MA_POSIX /* No need for dlfcn.h if we're not using runtime linking. */ #ifndef MA_NO_RUNTIME_LINKING #include <dlfcn.h> #endif #endif MA_API ma_handle ma_dlopen(ma_log* pLog, const char* filename) { #ifndef MA_NO_RUNTIME_LINKING { ma_handle handle; ma_log_postf(pLog, MA_LOG_LEVEL_DEBUG, "Loading library: %s\n", filename); #ifdef MA_WIN32 /* From MSDN: Desktop applications cannot use LoadPackagedLibrary; if a desktop application calls this function it fails with APPMODEL_ERROR_NO_PACKAGE.*/ #if !defined(MA_WIN32_UWP) || !(defined(WINAPI_FAMILY) && ((defined(WINAPI_FAMILY_PHONE_APP) && WINAPI_FAMILY == WINAPI_FAMILY_PHONE_APP))) handle = (ma_handle)LoadLibraryA(filename); #else /* *sigh* It appears there is no ANSI version of LoadPackagedLibrary()... */ WCHAR filenameW[4096]; if (MultiByteToWideChar(CP_UTF8, 0, filename, -1, filenameW, sizeof(filenameW)) == 0) { handle = NULL; } else { handle = (ma_handle)LoadPackagedLibrary(filenameW, 0); } #endif #else handle = (ma_handle)dlopen(filename, RTLD_NOW); #endif /* I'm not considering failure to load a library an error nor a warning because seamlessly falling through to a lower-priority backend is a deliberate design choice. Instead I'm logging it as an informational message. */ if (handle == NULL) { ma_log_postf(pLog, MA_LOG_LEVEL_INFO, "Failed to load library: %s\n", filename); } return handle; } #else { /* Runtime linking is disabled. */ (void)pLog; (void)filename; return NULL; } #endif } MA_API void ma_dlclose(ma_log* pLog, ma_handle handle) { #ifndef MA_NO_RUNTIME_LINKING { #ifdef MA_WIN32 { FreeLibrary((HMODULE)handle); } #else { /* Hack for Android bug (see https://github.com/android/ndk/issues/360). Calling dlclose() pre-API 28 may segfault. */ #if !defined(MA_ANDROID) || (defined(__ANDROID_API__) && __ANDROID_API__ >= 28) { dlclose((void*)handle); } #else { (void)handle; } #endif } #endif (void)pLog; } #else { /* Runtime linking is disabled. */ (void)pLog; (void)handle; } #endif } MA_API ma_proc ma_dlsym(ma_log* pLog, ma_handle handle, const char* symbol) { #ifndef MA_NO_RUNTIME_LINKING { ma_proc proc; ma_log_postf(pLog, MA_LOG_LEVEL_DEBUG, "Loading symbol: %s\n", symbol); #ifdef _WIN32 { proc = (ma_proc)GetProcAddress((HMODULE)handle, symbol); } #else { #if (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))) || defined(__clang__) #pragma GCC diagnostic push #pragma GCC diagnostic ignored "-Wpedantic" #endif proc = (ma_proc)dlsym((void*)handle, symbol); #if (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))) || defined(__clang__) #pragma GCC diagnostic pop #endif } #endif if (proc == NULL) { ma_log_postf(pLog, MA_LOG_LEVEL_WARNING, "Failed to load symbol: %s\n", symbol); } (void)pLog; /* It's possible for pContext to be unused. */ return proc; } #else { /* Runtime linking is disabled. */ (void)pLog; (void)handle; (void)symbol; return NULL; } #endif } /************************************************************************************************************************************************************ ************************************************************************************************************************************************************* DEVICE I/O ========== ************************************************************************************************************************************************************* ************************************************************************************************************************************************************/ #ifdef MA_APPLE #include <AvailabilityMacros.h> #endif #ifndef MA_NO_DEVICE_IO #if defined(MA_APPLE) && (MAC_OS_X_VERSION_MIN_REQUIRED < 101200) #include <mach/mach_time.h> /* For mach_absolute_time() */ #endif #ifdef MA_POSIX #include <sys/types.h> #endif /* This must be set to at least 26. */ #ifndef MA_AAUDIO_MIN_ANDROID_SDK_VERSION #define MA_AAUDIO_MIN_ANDROID_SDK_VERSION 27 #endif MA_API void ma_device_info_add_native_data_format(ma_device_info* pDeviceInfo, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 flags) { if (pDeviceInfo == NULL) { return; } if (pDeviceInfo->nativeDataFormatCount < ma_countof(pDeviceInfo->nativeDataFormats)) { pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = flags; pDeviceInfo->nativeDataFormatCount += 1; } } typedef struct { ma_backend backend; const char* pName; } ma_backend_info; static ma_backend_info gBackendInfo[] = /* Indexed by the backend enum. Must be in the order backends are declared in the ma_backend enum. */ { {ma_backend_wasapi, "WASAPI"}, {ma_backend_dsound, "DirectSound"}, {ma_backend_winmm, "WinMM"}, {ma_backend_coreaudio, "Core Audio"}, {ma_backend_sndio, "sndio"}, {ma_backend_audio4, "audio(4)"}, {ma_backend_oss, "OSS"}, {ma_backend_pulseaudio, "PulseAudio"}, {ma_backend_alsa, "ALSA"}, {ma_backend_jack, "JACK"}, {ma_backend_aaudio, "AAudio"}, {ma_backend_opensl, "OpenSL|ES"}, {ma_backend_webaudio, "Web Audio"}, {ma_backend_custom, "Custom"}, {ma_backend_null, "Null"} }; MA_API const char* ma_get_backend_name(ma_backend backend) { if (backend < 0 || backend >= (int)ma_countof(gBackendInfo)) { return "Unknown"; } return gBackendInfo[backend].pName; } MA_API ma_result ma_get_backend_from_name(const char* pBackendName, ma_backend* pBackend) { size_t iBackend; if (pBackendName == NULL) { return MA_INVALID_ARGS; } for (iBackend = 0; iBackend < ma_countof(gBackendInfo); iBackend += 1) { if (ma_strcmp(pBackendName, gBackendInfo[iBackend].pName) == 0) { if (pBackend != NULL) { *pBackend = gBackendInfo[iBackend].backend; } return MA_SUCCESS; } } /* Getting here means the backend name is unknown. */ return MA_INVALID_ARGS; } MA_API ma_bool32 ma_is_backend_enabled(ma_backend backend) { /* This looks a little bit gross, but we want all backends to be included in the switch to avoid warnings on some compilers about some enums not being handled by the switch statement. */ switch (backend) { case ma_backend_wasapi: #if defined(MA_HAS_WASAPI) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_dsound: #if defined(MA_HAS_DSOUND) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_winmm: #if defined(MA_HAS_WINMM) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_coreaudio: #if defined(MA_HAS_COREAUDIO) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_sndio: #if defined(MA_HAS_SNDIO) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_audio4: #if defined(MA_HAS_AUDIO4) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_oss: #if defined(MA_HAS_OSS) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_pulseaudio: #if defined(MA_HAS_PULSEAUDIO) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_alsa: #if defined(MA_HAS_ALSA) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_jack: #if defined(MA_HAS_JACK) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_aaudio: #if defined(MA_HAS_AAUDIO) #if defined(MA_ANDROID) { return ma_android_sdk_version() >= MA_AAUDIO_MIN_ANDROID_SDK_VERSION; } #else return MA_FALSE; #endif #else return MA_FALSE; #endif case ma_backend_opensl: #if defined(MA_HAS_OPENSL) #if defined(MA_ANDROID) { return ma_android_sdk_version() >= 9; } #else return MA_TRUE; #endif #else return MA_FALSE; #endif case ma_backend_webaudio: #if defined(MA_HAS_WEBAUDIO) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_custom: #if defined(MA_HAS_CUSTOM) return MA_TRUE; #else return MA_FALSE; #endif case ma_backend_null: #if defined(MA_HAS_NULL) return MA_TRUE; #else return MA_FALSE; #endif default: return MA_FALSE; } } MA_API ma_result ma_get_enabled_backends(ma_backend* pBackends, size_t backendCap, size_t* pBackendCount) { size_t backendCount; size_t iBackend; ma_result result = MA_SUCCESS; if (pBackendCount == NULL) { return MA_INVALID_ARGS; } backendCount = 0; for (iBackend = 0; iBackend <= ma_backend_null; iBackend += 1) { ma_backend backend = (ma_backend)iBackend; if (ma_is_backend_enabled(backend)) { /* The backend is enabled. Try adding it to the list. If there's no room, MA_NO_SPACE needs to be returned. */ if (backendCount == backendCap) { result = MA_NO_SPACE; break; } else { pBackends[backendCount] = backend; backendCount += 1; } } } if (pBackendCount != NULL) { *pBackendCount = backendCount; } return result; } MA_API ma_bool32 ma_is_loopback_supported(ma_backend backend) { switch (backend) { case ma_backend_wasapi: return MA_TRUE; case ma_backend_dsound: return MA_FALSE; case ma_backend_winmm: return MA_FALSE; case ma_backend_coreaudio: return MA_FALSE; case ma_backend_sndio: return MA_FALSE; case ma_backend_audio4: return MA_FALSE; case ma_backend_oss: return MA_FALSE; case ma_backend_pulseaudio: return MA_FALSE; case ma_backend_alsa: return MA_FALSE; case ma_backend_jack: return MA_FALSE; case ma_backend_aaudio: return MA_FALSE; case ma_backend_opensl: return MA_FALSE; case ma_backend_webaudio: return MA_FALSE; case ma_backend_custom: return MA_FALSE; /* <-- Will depend on the implementation of the backend. */ case ma_backend_null: return MA_FALSE; default: return MA_FALSE; } } #if defined(MA_WIN32) && !defined(MA_XBOX) /* WASAPI error codes. */ #define MA_AUDCLNT_E_NOT_INITIALIZED ((HRESULT)0x88890001) #define MA_AUDCLNT_E_ALREADY_INITIALIZED ((HRESULT)0x88890002) #define MA_AUDCLNT_E_WRONG_ENDPOINT_TYPE ((HRESULT)0x88890003) #define MA_AUDCLNT_E_DEVICE_INVALIDATED ((HRESULT)0x88890004) #define MA_AUDCLNT_E_NOT_STOPPED ((HRESULT)0x88890005) #define MA_AUDCLNT_E_BUFFER_TOO_LARGE ((HRESULT)0x88890006) #define MA_AUDCLNT_E_OUT_OF_ORDER ((HRESULT)0x88890007) #define MA_AUDCLNT_E_UNSUPPORTED_FORMAT ((HRESULT)0x88890008) #define MA_AUDCLNT_E_INVALID_SIZE ((HRESULT)0x88890009) #define MA_AUDCLNT_E_DEVICE_IN_USE ((HRESULT)0x8889000A) #define MA_AUDCLNT_E_BUFFER_OPERATION_PENDING ((HRESULT)0x8889000B) #define MA_AUDCLNT_E_THREAD_NOT_REGISTERED ((HRESULT)0x8889000C) #define MA_AUDCLNT_E_NO_SINGLE_PROCESS ((HRESULT)0x8889000D) #define MA_AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED ((HRESULT)0x8889000E) #define MA_AUDCLNT_E_ENDPOINT_CREATE_FAILED ((HRESULT)0x8889000F) #define MA_AUDCLNT_E_SERVICE_NOT_RUNNING ((HRESULT)0x88890010) #define MA_AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED ((HRESULT)0x88890011) #define MA_AUDCLNT_E_EXCLUSIVE_MODE_ONLY ((HRESULT)0x88890012) #define MA_AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL ((HRESULT)0x88890013) #define MA_AUDCLNT_E_EVENTHANDLE_NOT_SET ((HRESULT)0x88890014) #define MA_AUDCLNT_E_INCORRECT_BUFFER_SIZE ((HRESULT)0x88890015) #define MA_AUDCLNT_E_BUFFER_SIZE_ERROR ((HRESULT)0x88890016) #define MA_AUDCLNT_E_CPUUSAGE_EXCEEDED ((HRESULT)0x88890017) #define MA_AUDCLNT_E_BUFFER_ERROR ((HRESULT)0x88890018) #define MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED ((HRESULT)0x88890019) #define MA_AUDCLNT_E_INVALID_DEVICE_PERIOD ((HRESULT)0x88890020) #define MA_AUDCLNT_E_INVALID_STREAM_FLAG ((HRESULT)0x88890021) #define MA_AUDCLNT_E_ENDPOINT_OFFLOAD_NOT_CAPABLE ((HRESULT)0x88890022) #define MA_AUDCLNT_E_OUT_OF_OFFLOAD_RESOURCES ((HRESULT)0x88890023) #define MA_AUDCLNT_E_OFFLOAD_MODE_ONLY ((HRESULT)0x88890024) #define MA_AUDCLNT_E_NONOFFLOAD_MODE_ONLY ((HRESULT)0x88890025) #define MA_AUDCLNT_E_RESOURCES_INVALIDATED ((HRESULT)0x88890026) #define MA_AUDCLNT_E_RAW_MODE_UNSUPPORTED ((HRESULT)0x88890027) #define MA_AUDCLNT_E_ENGINE_PERIODICITY_LOCKED ((HRESULT)0x88890028) #define MA_AUDCLNT_E_ENGINE_FORMAT_LOCKED ((HRESULT)0x88890029) #define MA_AUDCLNT_E_HEADTRACKING_ENABLED ((HRESULT)0x88890030) #define MA_AUDCLNT_E_HEADTRACKING_UNSUPPORTED ((HRESULT)0x88890040) #define MA_AUDCLNT_S_BUFFER_EMPTY ((HRESULT)0x08890001) #define MA_AUDCLNT_S_THREAD_ALREADY_REGISTERED ((HRESULT)0x08890002) #define MA_AUDCLNT_S_POSITION_STALLED ((HRESULT)0x08890003) #define MA_DS_OK ((HRESULT)0) #define MA_DS_NO_VIRTUALIZATION ((HRESULT)0x0878000A) #define MA_DSERR_ALLOCATED ((HRESULT)0x8878000A) #define MA_DSERR_CONTROLUNAVAIL ((HRESULT)0x8878001E) #define MA_DSERR_INVALIDPARAM ((HRESULT)0x80070057) /*E_INVALIDARG*/ #define MA_DSERR_INVALIDCALL ((HRESULT)0x88780032) #define MA_DSERR_GENERIC ((HRESULT)0x80004005) /*E_FAIL*/ #define MA_DSERR_PRIOLEVELNEEDED ((HRESULT)0x88780046) #define MA_DSERR_OUTOFMEMORY ((HRESULT)0x8007000E) /*E_OUTOFMEMORY*/ #define MA_DSERR_BADFORMAT ((HRESULT)0x88780064) #define MA_DSERR_UNSUPPORTED ((HRESULT)0x80004001) /*E_NOTIMPL*/ #define MA_DSERR_NODRIVER ((HRESULT)0x88780078) #define MA_DSERR_ALREADYINITIALIZED ((HRESULT)0x88780082) #define MA_DSERR_NOAGGREGATION ((HRESULT)0x80040110) /*CLASS_E_NOAGGREGATION*/ #define MA_DSERR_BUFFERLOST ((HRESULT)0x88780096) #define MA_DSERR_OTHERAPPHASPRIO ((HRESULT)0x887800A0) #define MA_DSERR_UNINITIALIZED ((HRESULT)0x887800AA) #define MA_DSERR_NOINTERFACE ((HRESULT)0x80004002) /*E_NOINTERFACE*/ #define MA_DSERR_ACCESSDENIED ((HRESULT)0x80070005) /*E_ACCESSDENIED*/ #define MA_DSERR_BUFFERTOOSMALL ((HRESULT)0x887800B4) #define MA_DSERR_DS8_REQUIRED ((HRESULT)0x887800BE) #define MA_DSERR_SENDLOOP ((HRESULT)0x887800C8) #define MA_DSERR_BADSENDBUFFERGUID ((HRESULT)0x887800D2) #define MA_DSERR_OBJECTNOTFOUND ((HRESULT)0x88781161) #define MA_DSERR_FXUNAVAILABLE ((HRESULT)0x887800DC) static ma_result ma_result_from_HRESULT(HRESULT hr) { switch (hr) { case NOERROR: return MA_SUCCESS; /*case S_OK: return MA_SUCCESS;*/ case E_POINTER: return MA_INVALID_ARGS; case E_UNEXPECTED: return MA_ERROR; case E_NOTIMPL: return MA_NOT_IMPLEMENTED; case E_OUTOFMEMORY: return MA_OUT_OF_MEMORY; case E_INVALIDARG: return MA_INVALID_ARGS; case E_NOINTERFACE: return MA_API_NOT_FOUND; case E_HANDLE: return MA_INVALID_ARGS; case E_ABORT: return MA_ERROR; case E_FAIL: return MA_ERROR; case E_ACCESSDENIED: return MA_ACCESS_DENIED; /* WASAPI */ case MA_AUDCLNT_E_NOT_INITIALIZED: return MA_DEVICE_NOT_INITIALIZED; case MA_AUDCLNT_E_ALREADY_INITIALIZED: return MA_DEVICE_ALREADY_INITIALIZED; case MA_AUDCLNT_E_WRONG_ENDPOINT_TYPE: return MA_INVALID_ARGS; case MA_AUDCLNT_E_DEVICE_INVALIDATED: return MA_UNAVAILABLE; case MA_AUDCLNT_E_NOT_STOPPED: return MA_DEVICE_NOT_STOPPED; case MA_AUDCLNT_E_BUFFER_TOO_LARGE: return MA_TOO_BIG; case MA_AUDCLNT_E_OUT_OF_ORDER: return MA_INVALID_OPERATION; case MA_AUDCLNT_E_UNSUPPORTED_FORMAT: return MA_FORMAT_NOT_SUPPORTED; case MA_AUDCLNT_E_INVALID_SIZE: return MA_INVALID_ARGS; case MA_AUDCLNT_E_DEVICE_IN_USE: return MA_BUSY; case MA_AUDCLNT_E_BUFFER_OPERATION_PENDING: return MA_INVALID_OPERATION; case MA_AUDCLNT_E_THREAD_NOT_REGISTERED: return MA_DOES_NOT_EXIST; case MA_AUDCLNT_E_NO_SINGLE_PROCESS: return MA_INVALID_OPERATION; case MA_AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED: return MA_SHARE_MODE_NOT_SUPPORTED; case MA_AUDCLNT_E_ENDPOINT_CREATE_FAILED: return MA_FAILED_TO_OPEN_BACKEND_DEVICE; case MA_AUDCLNT_E_SERVICE_NOT_RUNNING: return MA_NOT_CONNECTED; case MA_AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED: return MA_INVALID_ARGS; case MA_AUDCLNT_E_EXCLUSIVE_MODE_ONLY: return MA_SHARE_MODE_NOT_SUPPORTED; case MA_AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL: return MA_INVALID_ARGS; case MA_AUDCLNT_E_EVENTHANDLE_NOT_SET: return MA_INVALID_ARGS; case MA_AUDCLNT_E_INCORRECT_BUFFER_SIZE: return MA_INVALID_ARGS; case MA_AUDCLNT_E_BUFFER_SIZE_ERROR: return MA_INVALID_ARGS; case MA_AUDCLNT_E_CPUUSAGE_EXCEEDED: return MA_ERROR; case MA_AUDCLNT_E_BUFFER_ERROR: return MA_ERROR; case MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED: return MA_INVALID_ARGS; case MA_AUDCLNT_E_INVALID_DEVICE_PERIOD: return MA_INVALID_ARGS; case MA_AUDCLNT_E_INVALID_STREAM_FLAG: return MA_INVALID_ARGS; case MA_AUDCLNT_E_ENDPOINT_OFFLOAD_NOT_CAPABLE: return MA_INVALID_OPERATION; case MA_AUDCLNT_E_OUT_OF_OFFLOAD_RESOURCES: return MA_OUT_OF_MEMORY; case MA_AUDCLNT_E_OFFLOAD_MODE_ONLY: return MA_INVALID_OPERATION; case MA_AUDCLNT_E_NONOFFLOAD_MODE_ONLY: return MA_INVALID_OPERATION; case MA_AUDCLNT_E_RESOURCES_INVALIDATED: return MA_INVALID_DATA; case MA_AUDCLNT_E_RAW_MODE_UNSUPPORTED: return MA_INVALID_OPERATION; case MA_AUDCLNT_E_ENGINE_PERIODICITY_LOCKED: return MA_INVALID_OPERATION; case MA_AUDCLNT_E_ENGINE_FORMAT_LOCKED: return MA_INVALID_OPERATION; case MA_AUDCLNT_E_HEADTRACKING_ENABLED: return MA_INVALID_OPERATION; case MA_AUDCLNT_E_HEADTRACKING_UNSUPPORTED: return MA_INVALID_OPERATION; case MA_AUDCLNT_S_BUFFER_EMPTY: return MA_NO_SPACE; case MA_AUDCLNT_S_THREAD_ALREADY_REGISTERED: return MA_ALREADY_EXISTS; case MA_AUDCLNT_S_POSITION_STALLED: return MA_ERROR; /* DirectSound */ /*case MA_DS_OK: return MA_SUCCESS;*/ /* S_OK */ case MA_DS_NO_VIRTUALIZATION: return MA_SUCCESS; case MA_DSERR_ALLOCATED: return MA_ALREADY_IN_USE; case MA_DSERR_CONTROLUNAVAIL: return MA_INVALID_OPERATION; /*case MA_DSERR_INVALIDPARAM: return MA_INVALID_ARGS;*/ /* E_INVALIDARG */ case MA_DSERR_INVALIDCALL: return MA_INVALID_OPERATION; /*case MA_DSERR_GENERIC: return MA_ERROR;*/ /* E_FAIL */ case MA_DSERR_PRIOLEVELNEEDED: return MA_INVALID_OPERATION; /*case MA_DSERR_OUTOFMEMORY: return MA_OUT_OF_MEMORY;*/ /* E_OUTOFMEMORY */ case MA_DSERR_BADFORMAT: return MA_FORMAT_NOT_SUPPORTED; /*case MA_DSERR_UNSUPPORTED: return MA_NOT_IMPLEMENTED;*/ /* E_NOTIMPL */ case MA_DSERR_NODRIVER: return MA_FAILED_TO_INIT_BACKEND; case MA_DSERR_ALREADYINITIALIZED: return MA_DEVICE_ALREADY_INITIALIZED; case MA_DSERR_NOAGGREGATION: return MA_ERROR; case MA_DSERR_BUFFERLOST: return MA_UNAVAILABLE; case MA_DSERR_OTHERAPPHASPRIO: return MA_ACCESS_DENIED; case MA_DSERR_UNINITIALIZED: return MA_DEVICE_NOT_INITIALIZED; /*case MA_DSERR_NOINTERFACE: return MA_API_NOT_FOUND;*/ /* E_NOINTERFACE */ /*case MA_DSERR_ACCESSDENIED: return MA_ACCESS_DENIED;*/ /* E_ACCESSDENIED */ case MA_DSERR_BUFFERTOOSMALL: return MA_NO_SPACE; case MA_DSERR_DS8_REQUIRED: return MA_INVALID_OPERATION; case MA_DSERR_SENDLOOP: return MA_DEADLOCK; case MA_DSERR_BADSENDBUFFERGUID: return MA_INVALID_ARGS; case MA_DSERR_OBJECTNOTFOUND: return MA_NO_DEVICE; case MA_DSERR_FXUNAVAILABLE: return MA_UNAVAILABLE; default: return MA_ERROR; } } /* PROPVARIANT */ #define MA_VT_LPWSTR 31 #define MA_VT_BLOB 65 #if defined(_MSC_VER) && !defined(__clang__) #pragma warning(push) #pragma warning(disable:4201) /* nonstandard extension used: nameless struct/union */ #elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))) #pragma GCC diagnostic push #pragma GCC diagnostic ignored "-Wpedantic" /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */ #if defined(__clang__) #pragma GCC diagnostic ignored "-Wc11-extensions" /* anonymous unions are a C11 extension */ #endif #endif typedef struct { WORD vt; WORD wReserved1; WORD wReserved2; WORD wReserved3; union { struct { ULONG cbSize; BYTE* pBlobData; } blob; WCHAR* pwszVal; char pad[16]; /* Just to ensure the size of the struct matches the official version. */ }; } MA_PROPVARIANT; #if defined(_MSC_VER) && !defined(__clang__) #pragma warning(pop) #elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))) #pragma GCC diagnostic pop #endif typedef HRESULT (WINAPI * MA_PFN_CoInitialize)(void* pvReserved); typedef HRESULT (WINAPI * MA_PFN_CoInitializeEx)(void* pvReserved, DWORD dwCoInit); typedef void (WINAPI * MA_PFN_CoUninitialize)(void); typedef HRESULT (WINAPI * MA_PFN_CoCreateInstance)(const IID* rclsid, void* pUnkOuter, DWORD dwClsContext, const IID* riid, void* ppv); typedef void (WINAPI * MA_PFN_CoTaskMemFree)(void* pv); typedef HRESULT (WINAPI * MA_PFN_PropVariantClear)(MA_PROPVARIANT *pvar); typedef int (WINAPI * MA_PFN_StringFromGUID2)(const GUID* const rguid, WCHAR* lpsz, int cchMax); typedef HWND (WINAPI * MA_PFN_GetForegroundWindow)(void); typedef HWND (WINAPI * MA_PFN_GetDesktopWindow)(void); #if defined(MA_WIN32_DESKTOP) /* Microsoft documents these APIs as returning LSTATUS, but the Win32 API shipping with some compilers do not define it. It's just a LONG. */ typedef LONG (WINAPI * MA_PFN_RegOpenKeyExA)(HKEY hKey, const char* lpSubKey, DWORD ulOptions, DWORD samDesired, HKEY* phkResult); typedef LONG (WINAPI * MA_PFN_RegCloseKey)(HKEY hKey); typedef LONG (WINAPI * MA_PFN_RegQueryValueExA)(HKEY hKey, const char* lpValueName, DWORD* lpReserved, DWORD* lpType, BYTE* lpData, DWORD* lpcbData); #endif /* MA_WIN32_DESKTOP */ static GUID MA_GUID_KSDATAFORMAT_SUBTYPE_PCM = {0x00000001, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}}; static GUID MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = {0x00000003, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}}; /*static GUID MA_GUID_KSDATAFORMAT_SUBTYPE_ALAW = {0x00000006, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/ /*static GUID MA_GUID_KSDATAFORMAT_SUBTYPE_MULAW = {0x00000007, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/ MA_API size_t ma_strlen_WCHAR(const WCHAR* str) { size_t len = 0; while (str[len] != '\0') { len += 1; } return len; } MA_API int ma_strcmp_WCHAR(const WCHAR *s1, const WCHAR *s2) { while (*s1 != '\0' && *s1 == *s2) { s1 += 1; s2 += 1; } return *s1 - *s2; } MA_API int ma_strcpy_s_WCHAR(WCHAR* dst, size_t dstCap, const WCHAR* src) { size_t i; if (dst == 0) { return 22; } if (dstCap == 0) { return 34; } if (src == 0) { dst[0] = '\0'; return 22; } for (i = 0; i < dstCap && src[i] != '\0'; ++i) { dst[i] = src[i]; } if (i < dstCap) { dst[i] = '\0'; return 0; } dst[0] = '\0'; return 34; } #endif /* MA_WIN32 */ #define MA_DEFAULT_PLAYBACK_DEVICE_NAME "Default Playback Device" #define MA_DEFAULT_CAPTURE_DEVICE_NAME "Default Capture Device" /******************************************************************************* Timing *******************************************************************************/ #if defined(MA_WIN32) && !defined(MA_POSIX) static LARGE_INTEGER g_ma_TimerFrequency; /* <-- Initialized to zero since it's static. */ static MA_INLINE void ma_timer_init(ma_timer* pTimer) { LARGE_INTEGER counter; if (g_ma_TimerFrequency.QuadPart == 0) { QueryPerformanceFrequency(&g_ma_TimerFrequency); } QueryPerformanceCounter(&counter); pTimer->counter = counter.QuadPart; } static MA_INLINE double ma_timer_get_time_in_seconds(ma_timer* pTimer) { LARGE_INTEGER counter; if (!QueryPerformanceCounter(&counter)) { return 0; } return (double)(counter.QuadPart - pTimer->counter) / g_ma_TimerFrequency.QuadPart; } #elif defined(MA_APPLE) && (MAC_OS_X_VERSION_MIN_REQUIRED < 101200) static ma_uint64 g_ma_TimerFrequency = 0; static MA_INLINE void ma_timer_init(ma_timer* pTimer) { mach_timebase_info_data_t baseTime; mach_timebase_info(&baseTime); g_ma_TimerFrequency = (baseTime.denom * 1e9) / baseTime.numer; pTimer->counter = mach_absolute_time(); } static MA_INLINE double ma_timer_get_time_in_seconds(ma_timer* pTimer) { ma_uint64 newTimeCounter = mach_absolute_time(); ma_uint64 oldTimeCounter = pTimer->counter; return (newTimeCounter - oldTimeCounter) / g_ma_TimerFrequency; } #elif defined(MA_EMSCRIPTEN) static MA_INLINE void ma_timer_init(ma_timer* pTimer) { pTimer->counterD = emscripten_get_now(); } static MA_INLINE double ma_timer_get_time_in_seconds(ma_timer* pTimer) { return (emscripten_get_now() - pTimer->counterD) / 1000; /* Emscripten is in milliseconds. */ } #else #if defined(_POSIX_C_SOURCE) && _POSIX_C_SOURCE >= 199309L #if defined(CLOCK_MONOTONIC) #define MA_CLOCK_ID CLOCK_MONOTONIC #else #define MA_CLOCK_ID CLOCK_REALTIME #endif static MA_INLINE void ma_timer_init(ma_timer* pTimer) { struct timespec newTime; clock_gettime(MA_CLOCK_ID, &newTime); pTimer->counter = ((ma_int64)newTime.tv_sec * 1000000000) + newTime.tv_nsec; } static MA_INLINE double ma_timer_get_time_in_seconds(ma_timer* pTimer) { ma_uint64 newTimeCounter; ma_uint64 oldTimeCounter; struct timespec newTime; clock_gettime(MA_CLOCK_ID, &newTime); newTimeCounter = ((ma_uint64)newTime.tv_sec * 1000000000) + newTime.tv_nsec; oldTimeCounter = pTimer->counter; return (newTimeCounter - oldTimeCounter) / 1000000000.0; } #else static MA_INLINE void ma_timer_init(ma_timer* pTimer) { struct timeval newTime; gettimeofday(&newTime, NULL); pTimer->counter = ((ma_int64)newTime.tv_sec * 1000000) + newTime.tv_usec; } static MA_INLINE double ma_timer_get_time_in_seconds(ma_timer* pTimer) { ma_uint64 newTimeCounter; ma_uint64 oldTimeCounter; struct timeval newTime; gettimeofday(&newTime, NULL); newTimeCounter = ((ma_uint64)newTime.tv_sec * 1000000) + newTime.tv_usec; oldTimeCounter = pTimer->counter; return (newTimeCounter - oldTimeCounter) / 1000000.0; } #endif #endif #if 0 static ma_uint32 ma_get_closest_standard_sample_rate(ma_uint32 sampleRateIn) { ma_uint32 closestRate = 0; ma_uint32 closestDiff = 0xFFFFFFFF; size_t iStandardRate; for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) { ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate]; ma_uint32 diff; if (sampleRateIn > standardRate) { diff = sampleRateIn - standardRate; } else { diff = standardRate - sampleRateIn; } if (diff == 0) { return standardRate; /* The input sample rate is a standard rate. */ } if (closestDiff > diff) { closestDiff = diff; closestRate = standardRate; } } return closestRate; } #endif static MA_INLINE unsigned int ma_device_disable_denormals(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (!pDevice->noDisableDenormals) { return ma_disable_denormals(); } else { return 0; } } static MA_INLINE void ma_device_restore_denormals(ma_device* pDevice, unsigned int prevState) { MA_ASSERT(pDevice != NULL); if (!pDevice->noDisableDenormals) { ma_restore_denormals(prevState); } else { /* Do nothing. */ (void)prevState; } } static ma_device_notification ma_device_notification_init(ma_device* pDevice, ma_device_notification_type type) { ma_device_notification notification; MA_ZERO_OBJECT(¬ification); notification.pDevice = pDevice; notification.type = type; return notification; } static void ma_device__on_notification(ma_device_notification notification) { MA_ASSERT(notification.pDevice != NULL); if (notification.pDevice->onNotification != NULL) { notification.pDevice->onNotification(¬ification); } /* TEMP FOR COMPATIBILITY: If it's a stopped notification, fire the onStop callback as well. This is only for backwards compatibility and will be removed. */ if (notification.pDevice->onStop != NULL && notification.type == ma_device_notification_type_stopped) { notification.pDevice->onStop(notification.pDevice); } } static void ma_device__on_notification_started(ma_device* pDevice) { ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_started)); } static void ma_device__on_notification_stopped(ma_device* pDevice) { ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_stopped)); } /* Not all platforms support reroute notifications. */ #if !defined(MA_EMSCRIPTEN) static void ma_device__on_notification_rerouted(ma_device* pDevice) { ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_rerouted)); } #endif #if defined(MA_EMSCRIPTEN) #ifdef __cplusplus extern "C" { #endif void EMSCRIPTEN_KEEPALIVE ma_device__on_notification_unlocked(ma_device* pDevice) { ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_unlocked)); } #ifdef __cplusplus } #endif #endif static void ma_device__on_data_inner(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount) { MA_ASSERT(pDevice != NULL); MA_ASSERT(pDevice->onData != NULL); if (!pDevice->noPreSilencedOutputBuffer && pFramesOut != NULL) { ma_silence_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels); } pDevice->onData(pDevice, pFramesOut, pFramesIn, frameCount); } static void ma_device__on_data(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount) { MA_ASSERT(pDevice != NULL); /* Don't read more data from the client if we're in the process of stopping. */ if (ma_device_get_state(pDevice) == ma_device_state_stopping) { return; } if (pDevice->noFixedSizedCallback) { /* Fast path. Not using a fixed sized callback. Process directly from the specified buffers. */ ma_device__on_data_inner(pDevice, pFramesOut, pFramesIn, frameCount); } else { /* Slow path. Using a fixed sized callback. Need to use the intermediary buffer. */ ma_uint32 totalFramesProcessed = 0; while (totalFramesProcessed < frameCount) { ma_uint32 totalFramesRemaining = frameCount - totalFramesProcessed; ma_uint32 framesToProcessThisIteration = 0; if (pFramesIn != NULL) { /* Capturing. Write to the intermediary buffer. If there's no room, fire the callback to empty it. */ if (pDevice->capture.intermediaryBufferLen < pDevice->capture.intermediaryBufferCap) { /* There's some room left in the intermediary buffer. Write to it without firing the callback. */ framesToProcessThisIteration = totalFramesRemaining; if (framesToProcessThisIteration > pDevice->capture.intermediaryBufferCap - pDevice->capture.intermediaryBufferLen) { framesToProcessThisIteration = pDevice->capture.intermediaryBufferCap - pDevice->capture.intermediaryBufferLen; } ma_copy_pcm_frames( ma_offset_pcm_frames_ptr(pDevice->capture.pIntermediaryBuffer, pDevice->capture.intermediaryBufferLen, pDevice->capture.format, pDevice->capture.channels), ma_offset_pcm_frames_const_ptr(pFramesIn, totalFramesProcessed, pDevice->capture.format, pDevice->capture.channels), framesToProcessThisIteration, pDevice->capture.format, pDevice->capture.channels); pDevice->capture.intermediaryBufferLen += framesToProcessThisIteration; } if (pDevice->capture.intermediaryBufferLen == pDevice->capture.intermediaryBufferCap) { /* No room left in the intermediary buffer. Fire the data callback. */ if (pDevice->type == ma_device_type_duplex) { /* We'll do the duplex data callback later after we've processed the playback data. */ } else { ma_device__on_data_inner(pDevice, NULL, pDevice->capture.pIntermediaryBuffer, pDevice->capture.intermediaryBufferCap); /* The intermediary buffer has just been drained. */ pDevice->capture.intermediaryBufferLen = 0; } } } if (pFramesOut != NULL) { /* Playing back. Read from the intermediary buffer. If there's nothing in it, fire the callback to fill it. */ if (pDevice->playback.intermediaryBufferLen > 0) { /* There's some content in the intermediary buffer. Read from that without firing the callback. */ if (pDevice->type == ma_device_type_duplex) { /* The frames processed this iteration for a duplex device will always be based on the capture side. Leave it unmodified. */ } else { framesToProcessThisIteration = totalFramesRemaining; if (framesToProcessThisIteration > pDevice->playback.intermediaryBufferLen) { framesToProcessThisIteration = pDevice->playback.intermediaryBufferLen; } } ma_copy_pcm_frames( ma_offset_pcm_frames_ptr(pFramesOut, totalFramesProcessed, pDevice->playback.format, pDevice->playback.channels), ma_offset_pcm_frames_ptr(pDevice->playback.pIntermediaryBuffer, pDevice->playback.intermediaryBufferCap - pDevice->playback.intermediaryBufferLen, pDevice->playback.format, pDevice->playback.channels), framesToProcessThisIteration, pDevice->playback.format, pDevice->playback.channels); pDevice->playback.intermediaryBufferLen -= framesToProcessThisIteration; } if (pDevice->playback.intermediaryBufferLen == 0) { /* There's nothing in the intermediary buffer. Fire the data callback to fill it. */ if (pDevice->type == ma_device_type_duplex) { /* In duplex mode, the data callback will be fired later. Nothing to do here. */ } else { ma_device__on_data_inner(pDevice, pDevice->playback.pIntermediaryBuffer, NULL, pDevice->playback.intermediaryBufferCap); /* The intermediary buffer has just been filled. */ pDevice->playback.intermediaryBufferLen = pDevice->playback.intermediaryBufferCap; } } } /* If we're in duplex mode we might need to do a refill of the data. */ if (pDevice->type == ma_device_type_duplex) { if (pDevice->capture.intermediaryBufferLen == pDevice->capture.intermediaryBufferCap) { ma_device__on_data_inner(pDevice, pDevice->playback.pIntermediaryBuffer, pDevice->capture.pIntermediaryBuffer, pDevice->capture.intermediaryBufferCap); pDevice->playback.intermediaryBufferLen = pDevice->playback.intermediaryBufferCap; /* The playback buffer will have just been filled. */ pDevice->capture.intermediaryBufferLen = 0; /* The intermediary buffer has just been drained. */ } } /* Make sure this is only incremented once in the duplex case. */ totalFramesProcessed += framesToProcessThisIteration; } } } static void ma_device__handle_data_callback(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount) { float masterVolumeFactor; ma_device_get_master_volume(pDevice, &masterVolumeFactor); /* Use ma_device_get_master_volume() to ensure the volume is loaded atomically. */ if (pDevice->onData) { unsigned int prevDenormalState = ma_device_disable_denormals(pDevice); { /* Volume control of input makes things a bit awkward because the input buffer is read-only. We'll need to use a temp buffer and loop in this case. */ if (pFramesIn != NULL && masterVolumeFactor != 1) { ma_uint8 tempFramesIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; ma_uint32 bpfCapture = ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); ma_uint32 bpfPlayback = ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); ma_uint32 totalFramesProcessed = 0; while (totalFramesProcessed < frameCount) { ma_uint32 framesToProcessThisIteration = frameCount - totalFramesProcessed; if (framesToProcessThisIteration > sizeof(tempFramesIn)/bpfCapture) { framesToProcessThisIteration = sizeof(tempFramesIn)/bpfCapture; } ma_copy_and_apply_volume_factor_pcm_frames(tempFramesIn, ma_offset_ptr(pFramesIn, totalFramesProcessed*bpfCapture), framesToProcessThisIteration, pDevice->capture.format, pDevice->capture.channels, masterVolumeFactor); ma_device__on_data(pDevice, ma_offset_ptr(pFramesOut, totalFramesProcessed*bpfPlayback), tempFramesIn, framesToProcessThisIteration); totalFramesProcessed += framesToProcessThisIteration; } } else { ma_device__on_data(pDevice, pFramesOut, pFramesIn, frameCount); } /* Volume control and clipping for playback devices. */ if (pFramesOut != NULL) { if (masterVolumeFactor != 1) { if (pFramesIn == NULL) { /* <-- In full-duplex situations, the volume will have been applied to the input samples before the data callback. Applying it again post-callback will incorrectly compound it. */ ma_apply_volume_factor_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels, masterVolumeFactor); } } if (!pDevice->noClip && pDevice->playback.format == ma_format_f32) { ma_clip_samples_f32((float*)pFramesOut, (const float*)pFramesOut, frameCount * pDevice->playback.channels); /* Intentionally specifying the same pointer for both input and output for in-place processing. */ } } } ma_device_restore_denormals(pDevice, prevDenormalState); } else { /* No data callback. Just silence the output. */ if (pFramesOut != NULL) { ma_silence_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels); } } } /* A helper function for reading sample data from the client. */ static void ma_device__read_frames_from_client(ma_device* pDevice, ma_uint32 frameCount, void* pFramesOut) { MA_ASSERT(pDevice != NULL); MA_ASSERT(frameCount > 0); MA_ASSERT(pFramesOut != NULL); if (pDevice->playback.converter.isPassthrough) { ma_device__handle_data_callback(pDevice, pFramesOut, NULL, frameCount); } else { ma_result result; ma_uint64 totalFramesReadOut; void* pRunningFramesOut; totalFramesReadOut = 0; pRunningFramesOut = pFramesOut; /* We run slightly different logic depending on whether or not we're using a heap-allocated buffer for caching input data. This will be the case if the data converter does not have the ability to retrieve the required input frame count for a given output frame count. */ if (pDevice->playback.pInputCache != NULL) { while (totalFramesReadOut < frameCount) { ma_uint64 framesToReadThisIterationIn; ma_uint64 framesToReadThisIterationOut; /* If there's any data available in the cache, that needs to get processed first. */ if (pDevice->playback.inputCacheRemaining > 0) { framesToReadThisIterationOut = (frameCount - totalFramesReadOut); framesToReadThisIterationIn = framesToReadThisIterationOut; if (framesToReadThisIterationIn > pDevice->playback.inputCacheRemaining) { framesToReadThisIterationIn = pDevice->playback.inputCacheRemaining; } result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, ma_offset_pcm_frames_ptr(pDevice->playback.pInputCache, pDevice->playback.inputCacheConsumed, pDevice->playback.format, pDevice->playback.channels), &framesToReadThisIterationIn, pRunningFramesOut, &framesToReadThisIterationOut); if (result != MA_SUCCESS) { break; } pDevice->playback.inputCacheConsumed += framesToReadThisIterationIn; pDevice->playback.inputCacheRemaining -= framesToReadThisIterationIn; totalFramesReadOut += framesToReadThisIterationOut; pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesToReadThisIterationOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); if (framesToReadThisIterationIn == 0 && framesToReadThisIterationOut == 0) { break; /* We're done. */ } } /* Getting here means there's no data in the cache and we need to fill it up with data from the client. */ if (pDevice->playback.inputCacheRemaining == 0) { ma_device__handle_data_callback(pDevice, pDevice->playback.pInputCache, NULL, (ma_uint32)pDevice->playback.inputCacheCap); pDevice->playback.inputCacheConsumed = 0; pDevice->playback.inputCacheRemaining = pDevice->playback.inputCacheCap; } } } else { while (totalFramesReadOut < frameCount) { ma_uint8 pIntermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In client format. */ ma_uint64 intermediaryBufferCap = sizeof(pIntermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); ma_uint64 framesToReadThisIterationIn; ma_uint64 framesReadThisIterationIn; ma_uint64 framesToReadThisIterationOut; ma_uint64 framesReadThisIterationOut; ma_uint64 requiredInputFrameCount; framesToReadThisIterationOut = (frameCount - totalFramesReadOut); framesToReadThisIterationIn = framesToReadThisIterationOut; if (framesToReadThisIterationIn > intermediaryBufferCap) { framesToReadThisIterationIn = intermediaryBufferCap; } ma_data_converter_get_required_input_frame_count(&pDevice->playback.converter, framesToReadThisIterationOut, &requiredInputFrameCount); if (framesToReadThisIterationIn > requiredInputFrameCount) { framesToReadThisIterationIn = requiredInputFrameCount; } ma_device__handle_data_callback(pDevice, pIntermediaryBuffer, NULL, (ma_uint32)framesToReadThisIterationIn); /* At this point we have our decoded data in input format and now we need to convert to output format. Note that even if we didn't read any input frames, we still want to try processing frames because there may some output frames generated from cached input data. */ framesReadThisIterationIn = framesToReadThisIterationIn; framesReadThisIterationOut = framesToReadThisIterationOut; result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pIntermediaryBuffer, &framesReadThisIterationIn, pRunningFramesOut, &framesReadThisIterationOut); if (result != MA_SUCCESS) { break; } totalFramesReadOut += framesReadThisIterationOut; pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesReadThisIterationOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); if (framesReadThisIterationIn == 0 && framesReadThisIterationOut == 0) { break; /* We're done. */ } } } } } /* A helper for sending sample data to the client. */ static void ma_device__send_frames_to_client(ma_device* pDevice, ma_uint32 frameCountInDeviceFormat, const void* pFramesInDeviceFormat) { MA_ASSERT(pDevice != NULL); MA_ASSERT(frameCountInDeviceFormat > 0); MA_ASSERT(pFramesInDeviceFormat != NULL); if (pDevice->capture.converter.isPassthrough) { ma_device__handle_data_callback(pDevice, NULL, pFramesInDeviceFormat, frameCountInDeviceFormat); } else { ma_result result; ma_uint8 pFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; ma_uint64 framesInClientFormatCap = sizeof(pFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); ma_uint64 totalDeviceFramesProcessed = 0; ma_uint64 totalClientFramesProcessed = 0; const void* pRunningFramesInDeviceFormat = pFramesInDeviceFormat; /* We just keep going until we've exhausted all of our input frames and cannot generate any more output frames. */ for (;;) { ma_uint64 deviceFramesProcessedThisIteration; ma_uint64 clientFramesProcessedThisIteration; deviceFramesProcessedThisIteration = (frameCountInDeviceFormat - totalDeviceFramesProcessed); clientFramesProcessedThisIteration = framesInClientFormatCap; result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningFramesInDeviceFormat, &deviceFramesProcessedThisIteration, pFramesInClientFormat, &clientFramesProcessedThisIteration); if (result != MA_SUCCESS) { break; } if (clientFramesProcessedThisIteration > 0) { ma_device__handle_data_callback(pDevice, NULL, pFramesInClientFormat, (ma_uint32)clientFramesProcessedThisIteration); /* Safe cast. */ } pRunningFramesInDeviceFormat = ma_offset_ptr(pRunningFramesInDeviceFormat, deviceFramesProcessedThisIteration * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); totalDeviceFramesProcessed += deviceFramesProcessedThisIteration; totalClientFramesProcessed += clientFramesProcessedThisIteration; /* This is just to silence a warning. I might want to use this variable later so leaving in place for now. */ (void)totalClientFramesProcessed; if (deviceFramesProcessedThisIteration == 0 && clientFramesProcessedThisIteration == 0) { break; /* We're done. */ } } } } static ma_result ma_device__handle_duplex_callback_capture(ma_device* pDevice, ma_uint32 frameCountInDeviceFormat, const void* pFramesInDeviceFormat, ma_pcm_rb* pRB) { ma_result result; ma_uint32 totalDeviceFramesProcessed = 0; const void* pRunningFramesInDeviceFormat = pFramesInDeviceFormat; MA_ASSERT(pDevice != NULL); MA_ASSERT(frameCountInDeviceFormat > 0); MA_ASSERT(pFramesInDeviceFormat != NULL); MA_ASSERT(pRB != NULL); /* Write to the ring buffer. The ring buffer is in the client format which means we need to convert. */ for (;;) { ma_uint32 framesToProcessInDeviceFormat = (frameCountInDeviceFormat - totalDeviceFramesProcessed); ma_uint32 framesToProcessInClientFormat = MA_DATA_CONVERTER_STACK_BUFFER_SIZE / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); ma_uint64 framesProcessedInDeviceFormat; ma_uint64 framesProcessedInClientFormat; void* pFramesInClientFormat; result = ma_pcm_rb_acquire_write(pRB, &framesToProcessInClientFormat, &pFramesInClientFormat); if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "Failed to acquire capture PCM frames from ring buffer."); break; } if (framesToProcessInClientFormat == 0) { if (ma_pcm_rb_pointer_distance(pRB) == (ma_int32)ma_pcm_rb_get_subbuffer_size(pRB)) { break; /* Overrun. Not enough room in the ring buffer for input frame. Excess frames are dropped. */ } } /* Convert. */ framesProcessedInDeviceFormat = framesToProcessInDeviceFormat; framesProcessedInClientFormat = framesToProcessInClientFormat; result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningFramesInDeviceFormat, &framesProcessedInDeviceFormat, pFramesInClientFormat, &framesProcessedInClientFormat); if (result != MA_SUCCESS) { break; } result = ma_pcm_rb_commit_write(pRB, (ma_uint32)framesProcessedInClientFormat); /* Safe cast. */ if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "Failed to commit capture PCM frames to ring buffer."); break; } pRunningFramesInDeviceFormat = ma_offset_ptr(pRunningFramesInDeviceFormat, framesProcessedInDeviceFormat * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); totalDeviceFramesProcessed += (ma_uint32)framesProcessedInDeviceFormat; /* Safe cast. */ /* We're done when we're unable to process any client nor device frames. */ if (framesProcessedInClientFormat == 0 && framesProcessedInDeviceFormat == 0) { break; /* Done. */ } } return MA_SUCCESS; } static ma_result ma_device__handle_duplex_callback_playback(ma_device* pDevice, ma_uint32 frameCount, void* pFramesInInternalFormat, ma_pcm_rb* pRB) { ma_result result; ma_uint8 silentInputFrames[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; ma_uint32 totalFramesReadOut = 0; MA_ASSERT(pDevice != NULL); MA_ASSERT(frameCount > 0); MA_ASSERT(pFramesInInternalFormat != NULL); MA_ASSERT(pRB != NULL); MA_ASSERT(pDevice->playback.pInputCache != NULL); /* Sitting in the ring buffer should be captured data from the capture callback in external format. If there's not enough data in there for the whole frameCount frames we just use silence instead for the input data. */ MA_ZERO_MEMORY(silentInputFrames, sizeof(silentInputFrames)); while (totalFramesReadOut < frameCount && ma_device_is_started(pDevice)) { /* We should have a buffer allocated on the heap. Any playback frames still sitting in there need to be sent to the internal device before we process any more data from the client. */ if (pDevice->playback.inputCacheRemaining > 0) { ma_uint64 framesConvertedIn = pDevice->playback.inputCacheRemaining; ma_uint64 framesConvertedOut = (frameCount - totalFramesReadOut); ma_data_converter_process_pcm_frames(&pDevice->playback.converter, ma_offset_pcm_frames_ptr(pDevice->playback.pInputCache, pDevice->playback.inputCacheConsumed, pDevice->playback.format, pDevice->playback.channels), &framesConvertedIn, pFramesInInternalFormat, &framesConvertedOut); pDevice->playback.inputCacheConsumed += framesConvertedIn; pDevice->playback.inputCacheRemaining -= framesConvertedIn; totalFramesReadOut += (ma_uint32)framesConvertedOut; /* Safe cast. */ pFramesInInternalFormat = ma_offset_ptr(pFramesInInternalFormat, framesConvertedOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); } /* If there's no more data in the cache we'll need to fill it with some. */ if (totalFramesReadOut < frameCount && pDevice->playback.inputCacheRemaining == 0) { ma_uint32 inputFrameCount; void* pInputFrames; inputFrameCount = (ma_uint32)pDevice->playback.inputCacheCap; result = ma_pcm_rb_acquire_read(pRB, &inputFrameCount, &pInputFrames); if (result == MA_SUCCESS) { if (inputFrameCount > 0) { ma_device__handle_data_callback(pDevice, pDevice->playback.pInputCache, pInputFrames, inputFrameCount); } else { if (ma_pcm_rb_pointer_distance(pRB) == 0) { break; /* Underrun. */ } } } else { /* No capture data available. Feed in silence. */ inputFrameCount = (ma_uint32)ma_min(pDevice->playback.inputCacheCap, sizeof(silentInputFrames) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)); ma_device__handle_data_callback(pDevice, pDevice->playback.pInputCache, silentInputFrames, inputFrameCount); } pDevice->playback.inputCacheConsumed = 0; pDevice->playback.inputCacheRemaining = inputFrameCount; result = ma_pcm_rb_commit_read(pRB, inputFrameCount); if (result != MA_SUCCESS) { return result; /* Should never happen. */ } } } return MA_SUCCESS; } /* A helper for changing the state of the device. */ static MA_INLINE void ma_device__set_state(ma_device* pDevice, ma_device_state newState) { ma_atomic_device_state_set(&pDevice->state, newState); } MA_API ma_uint32 ma_get_format_priority_index(ma_format format) /* Lower = better. */ { ma_uint32 i; for (i = 0; i < ma_countof(g_maFormatPriorities); ++i) { if (g_maFormatPriorities[i] == format) { return i; } } /* Getting here means the format could not be found or is equal to ma_format_unknown. */ return (ma_uint32)-1; } static ma_result ma_device__post_init_setup(ma_device* pDevice, ma_device_type deviceType); static ma_bool32 ma_device_descriptor_is_valid(const ma_device_descriptor* pDeviceDescriptor) { if (pDeviceDescriptor == NULL) { return MA_FALSE; } if (pDeviceDescriptor->format == ma_format_unknown) { return MA_FALSE; } if (pDeviceDescriptor->channels == 0 || pDeviceDescriptor->channels > MA_MAX_CHANNELS) { return MA_FALSE; } if (pDeviceDescriptor->sampleRate == 0) { return MA_FALSE; } return MA_TRUE; } static ma_result ma_device_audio_thread__default_read_write(ma_device* pDevice) { ma_result result = MA_SUCCESS; ma_bool32 exitLoop = MA_FALSE; ma_uint8 capturedDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; ma_uint8 playbackDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; ma_uint32 capturedDeviceDataCapInFrames = 0; ma_uint32 playbackDeviceDataCapInFrames = 0; MA_ASSERT(pDevice != NULL); /* Just some quick validation on the device type and the available callbacks. */ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) { if (pDevice->pContext->callbacks.onDeviceRead == NULL) { return MA_NOT_IMPLEMENTED; } capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { if (pDevice->pContext->callbacks.onDeviceWrite == NULL) { return MA_NOT_IMPLEMENTED; } playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); } /* NOTE: The device was started outside of this function, in the worker thread. */ while (ma_device_get_state(pDevice) == ma_device_state_started && !exitLoop) { switch (pDevice->type) { case ma_device_type_duplex: { /* The process is: onDeviceRead() -> convert -> callback -> convert -> onDeviceWrite() */ ma_uint32 totalCapturedDeviceFramesProcessed = 0; ma_uint32 capturedDevicePeriodSizeInFrames = ma_min(pDevice->capture.internalPeriodSizeInFrames, pDevice->playback.internalPeriodSizeInFrames); while (totalCapturedDeviceFramesProcessed < capturedDevicePeriodSizeInFrames) { ma_uint32 capturedDeviceFramesRemaining; ma_uint32 capturedDeviceFramesProcessed; ma_uint32 capturedDeviceFramesToProcess; ma_uint32 capturedDeviceFramesToTryProcessing = capturedDevicePeriodSizeInFrames - totalCapturedDeviceFramesProcessed; if (capturedDeviceFramesToTryProcessing > capturedDeviceDataCapInFrames) { capturedDeviceFramesToTryProcessing = capturedDeviceDataCapInFrames; } result = pDevice->pContext->callbacks.onDeviceRead(pDevice, capturedDeviceData, capturedDeviceFramesToTryProcessing, &capturedDeviceFramesToProcess); if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; } capturedDeviceFramesRemaining = capturedDeviceFramesToProcess; capturedDeviceFramesProcessed = 0; /* At this point we have our captured data in device format and we now need to convert it to client format. */ for (;;) { ma_uint8 capturedClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; ma_uint8 playbackClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; ma_uint32 capturedClientDataCapInFrames = sizeof(capturedClientData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); ma_uint32 playbackClientDataCapInFrames = sizeof(playbackClientData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); ma_uint64 capturedClientFramesToProcessThisIteration = ma_min(capturedClientDataCapInFrames, playbackClientDataCapInFrames); ma_uint64 capturedDeviceFramesToProcessThisIteration = capturedDeviceFramesRemaining; ma_uint8* pRunningCapturedDeviceFrames = ma_offset_ptr(capturedDeviceData, capturedDeviceFramesProcessed * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); /* Convert capture data from device format to client format. */ result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningCapturedDeviceFrames, &capturedDeviceFramesToProcessThisIteration, capturedClientData, &capturedClientFramesToProcessThisIteration); if (result != MA_SUCCESS) { break; } /* If we weren't able to generate any output frames it must mean we've exhausted all of our input. The only time this would not be the case is if capturedClientData was too small which should never be the case when it's of the size MA_DATA_CONVERTER_STACK_BUFFER_SIZE. */ if (capturedClientFramesToProcessThisIteration == 0) { break; } ma_device__handle_data_callback(pDevice, playbackClientData, capturedClientData, (ma_uint32)capturedClientFramesToProcessThisIteration); /* Safe cast .*/ capturedDeviceFramesProcessed += (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ capturedDeviceFramesRemaining -= (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ /* At this point the playbackClientData buffer should be holding data that needs to be written to the device. */ for (;;) { ma_uint64 convertedClientFrameCount = capturedClientFramesToProcessThisIteration; ma_uint64 convertedDeviceFrameCount = playbackDeviceDataCapInFrames; result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackClientData, &convertedClientFrameCount, playbackDeviceData, &convertedDeviceFrameCount); if (result != MA_SUCCESS) { break; } result = pDevice->pContext->callbacks.onDeviceWrite(pDevice, playbackDeviceData, (ma_uint32)convertedDeviceFrameCount, NULL); /* Safe cast. */ if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; } capturedClientFramesToProcessThisIteration -= (ma_uint32)convertedClientFrameCount; /* Safe cast. */ if (capturedClientFramesToProcessThisIteration == 0) { break; } } /* In case an error happened from ma_device_write__null()... */ if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; } } /* Make sure we don't get stuck in the inner loop. */ if (capturedDeviceFramesProcessed == 0) { break; } totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed; } } break; case ma_device_type_capture: case ma_device_type_loopback: { ma_uint32 periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames; ma_uint32 framesReadThisPeriod = 0; while (framesReadThisPeriod < periodSizeInFrames) { ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; ma_uint32 framesProcessed; ma_uint32 framesToReadThisIteration = framesRemainingInPeriod; if (framesToReadThisIteration > capturedDeviceDataCapInFrames) { framesToReadThisIteration = capturedDeviceDataCapInFrames; } result = pDevice->pContext->callbacks.onDeviceRead(pDevice, capturedDeviceData, framesToReadThisIteration, &framesProcessed); if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; } /* Make sure we don't get stuck in the inner loop. */ if (framesProcessed == 0) { break; } ma_device__send_frames_to_client(pDevice, framesProcessed, capturedDeviceData); framesReadThisPeriod += framesProcessed; } } break; case ma_device_type_playback: { /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ ma_uint32 periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames; ma_uint32 framesWrittenThisPeriod = 0; while (framesWrittenThisPeriod < periodSizeInFrames) { ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; ma_uint32 framesProcessed; ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; if (framesToWriteThisIteration > playbackDeviceDataCapInFrames) { framesToWriteThisIteration = playbackDeviceDataCapInFrames; } ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, playbackDeviceData); result = pDevice->pContext->callbacks.onDeviceWrite(pDevice, playbackDeviceData, framesToWriteThisIteration, &framesProcessed); if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; } /* Make sure we don't get stuck in the inner loop. */ if (framesProcessed == 0) { break; } framesWrittenThisPeriod += framesProcessed; } } break; /* Should never get here. */ default: break; } } return result; } /******************************************************************************* Null Backend *******************************************************************************/ #ifdef MA_HAS_NULL #define MA_DEVICE_OP_NONE__NULL 0 #define MA_DEVICE_OP_START__NULL 1 #define MA_DEVICE_OP_SUSPEND__NULL 2 #define MA_DEVICE_OP_KILL__NULL 3 static ma_thread_result MA_THREADCALL ma_device_thread__null(void* pData) { ma_device* pDevice = (ma_device*)pData; MA_ASSERT(pDevice != NULL); for (;;) { /* Keep the thread alive until the device is uninitialized. */ ma_uint32 operation; /* Wait for an operation to be requested. */ ma_event_wait(&pDevice->null_device.operationEvent); /* At this point an event should have been triggered. */ operation = pDevice->null_device.operation; /* Starting the device needs to put the thread into a loop. */ if (operation == MA_DEVICE_OP_START__NULL) { /* Reset the timer just in case. */ ma_timer_init(&pDevice->null_device.timer); /* Getting here means a suspend or kill operation has been requested. */ pDevice->null_device.operationResult = MA_SUCCESS; ma_event_signal(&pDevice->null_device.operationCompletionEvent); ma_semaphore_release(&pDevice->null_device.operationSemaphore); continue; } /* Suspending the device means we need to stop the timer and just continue the loop. */ if (operation == MA_DEVICE_OP_SUSPEND__NULL) { /* We need to add the current run time to the prior run time, then reset the timer. */ pDevice->null_device.priorRunTime += ma_timer_get_time_in_seconds(&pDevice->null_device.timer); ma_timer_init(&pDevice->null_device.timer); /* We're done. */ pDevice->null_device.operationResult = MA_SUCCESS; ma_event_signal(&pDevice->null_device.operationCompletionEvent); ma_semaphore_release(&pDevice->null_device.operationSemaphore); continue; } /* Killing the device means we need to get out of this loop so that this thread can terminate. */ if (operation == MA_DEVICE_OP_KILL__NULL) { pDevice->null_device.operationResult = MA_SUCCESS; ma_event_signal(&pDevice->null_device.operationCompletionEvent); ma_semaphore_release(&pDevice->null_device.operationSemaphore); break; } /* Getting a signal on a "none" operation probably means an error. Return invalid operation. */ if (operation == MA_DEVICE_OP_NONE__NULL) { MA_ASSERT(MA_FALSE); /* <-- Trigger this in debug mode to ensure developers are aware they're doing something wrong (or there's a bug in a miniaudio). */ pDevice->null_device.operationResult = MA_INVALID_OPERATION; ma_event_signal(&pDevice->null_device.operationCompletionEvent); ma_semaphore_release(&pDevice->null_device.operationSemaphore); continue; /* Continue the loop. Don't terminate. */ } } return (ma_thread_result)0; } static ma_result ma_device_do_operation__null(ma_device* pDevice, ma_uint32 operation) { ma_result result; /* TODO: Need to review this and consider just using mutual exclusion. I think the original motivation for this was to just post the event to a queue and return immediately, but that has since changed and now this function is synchronous. I think this can be simplified to just use a mutex. */ /* The first thing to do is wait for an operation slot to become available. We only have a single slot for this, but we could extend this later to support queuing of operations. */ result = ma_semaphore_wait(&pDevice->null_device.operationSemaphore); if (result != MA_SUCCESS) { return result; /* Failed to wait for the event. */ } /* When we get here it means the background thread is not referencing the operation code and it can be changed. After changing this we need to signal an event to the worker thread to let it know that it can start work. */ pDevice->null_device.operation = operation; /* Once the operation code has been set, the worker thread can start work. */ if (ma_event_signal(&pDevice->null_device.operationEvent) != MA_SUCCESS) { return MA_ERROR; } /* We want everything to be synchronous so we're going to wait for the worker thread to complete it's operation. */ if (ma_event_wait(&pDevice->null_device.operationCompletionEvent) != MA_SUCCESS) { return MA_ERROR; } return pDevice->null_device.operationResult; } static ma_uint64 ma_device_get_total_run_time_in_frames__null(ma_device* pDevice) { ma_uint32 internalSampleRate; if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { internalSampleRate = pDevice->capture.internalSampleRate; } else { internalSampleRate = pDevice->playback.internalSampleRate; } return (ma_uint64)((pDevice->null_device.priorRunTime + ma_timer_get_time_in_seconds(&pDevice->null_device.timer)) * internalSampleRate); } static ma_result ma_context_enumerate_devices__null(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { ma_bool32 cbResult = MA_TRUE; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); /* Playback. */ if (cbResult) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Playback Device", (size_t)-1); deviceInfo.isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */ cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } /* Capture. */ if (cbResult) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Capture Device", (size_t)-1); deviceInfo.isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */ cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); } (void)cbResult; /* Silence a static analysis warning. */ return MA_SUCCESS; } static ma_result ma_context_get_device_info__null(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { MA_ASSERT(pContext != NULL); if (pDeviceID != NULL && pDeviceID->nullbackend != 0) { return MA_NO_DEVICE; /* Don't know the device. */ } /* Name / Description */ if (deviceType == ma_device_type_playback) { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Playback Device", (size_t)-1); } else { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Capture Device", (size_t)-1); } pDeviceInfo->isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */ /* Support everything on the null backend. */ pDeviceInfo->nativeDataFormats[0].format = ma_format_unknown; pDeviceInfo->nativeDataFormats[0].channels = 0; pDeviceInfo->nativeDataFormats[0].sampleRate = 0; pDeviceInfo->nativeDataFormats[0].flags = 0; pDeviceInfo->nativeDataFormatCount = 1; (void)pContext; return MA_SUCCESS; } static ma_result ma_device_uninit__null(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); /* Keep it clean and wait for the device thread to finish before returning. */ ma_device_do_operation__null(pDevice, MA_DEVICE_OP_KILL__NULL); /* Wait for the thread to finish before continuing. */ ma_thread_wait(&pDevice->null_device.deviceThread); /* At this point the loop in the device thread is as good as terminated so we can uninitialize our events. */ ma_semaphore_uninit(&pDevice->null_device.operationSemaphore); ma_event_uninit(&pDevice->null_device.operationCompletionEvent); ma_event_uninit(&pDevice->null_device.operationEvent); return MA_SUCCESS; } static ma_result ma_device_init__null(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { ma_result result; MA_ASSERT(pDevice != NULL); MA_ZERO_OBJECT(&pDevice->null_device); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } /* The null backend supports everything exactly as we specify it. */ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { pDescriptorCapture->format = (pDescriptorCapture->format != ma_format_unknown) ? pDescriptorCapture->format : MA_DEFAULT_FORMAT; pDescriptorCapture->channels = (pDescriptorCapture->channels != 0) ? pDescriptorCapture->channels : MA_DEFAULT_CHANNELS; pDescriptorCapture->sampleRate = (pDescriptorCapture->sampleRate != 0) ? pDescriptorCapture->sampleRate : MA_DEFAULT_SAMPLE_RATE; if (pDescriptorCapture->channelMap[0] == MA_CHANNEL_NONE) { ma_channel_map_init_standard(ma_standard_channel_map_default, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorCapture->channels); } pDescriptorCapture->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile); } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { pDescriptorPlayback->format = (pDescriptorPlayback->format != ma_format_unknown) ? pDescriptorPlayback->format : MA_DEFAULT_FORMAT; pDescriptorPlayback->channels = (pDescriptorPlayback->channels != 0) ? pDescriptorPlayback->channels : MA_DEFAULT_CHANNELS; pDescriptorPlayback->sampleRate = (pDescriptorPlayback->sampleRate != 0) ? pDescriptorPlayback->sampleRate : MA_DEFAULT_SAMPLE_RATE; if (pDescriptorPlayback->channelMap[0] == MA_CHANNEL_NONE) { ma_channel_map_init_standard(ma_standard_channel_map_default, pDescriptorPlayback->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorPlayback->channels); } pDescriptorPlayback->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile); } /* In order to get timing right, we need to create a thread that does nothing but keeps track of the timer. This timer is started when the first period is "written" to it, and then stopped in ma_device_stop__null(). */ result = ma_event_init(&pDevice->null_device.operationEvent); if (result != MA_SUCCESS) { return result; } result = ma_event_init(&pDevice->null_device.operationCompletionEvent); if (result != MA_SUCCESS) { return result; } result = ma_semaphore_init(1, &pDevice->null_device.operationSemaphore); /* <-- It's important that the initial value is set to 1. */ if (result != MA_SUCCESS) { return result; } result = ma_thread_create(&pDevice->null_device.deviceThread, pDevice->pContext->threadPriority, 0, ma_device_thread__null, pDevice, &pDevice->pContext->allocationCallbacks); if (result != MA_SUCCESS) { return result; } return MA_SUCCESS; } static ma_result ma_device_start__null(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); ma_device_do_operation__null(pDevice, MA_DEVICE_OP_START__NULL); ma_atomic_bool32_set(&pDevice->null_device.isStarted, MA_TRUE); return MA_SUCCESS; } static ma_result ma_device_stop__null(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); ma_device_do_operation__null(pDevice, MA_DEVICE_OP_SUSPEND__NULL); ma_atomic_bool32_set(&pDevice->null_device.isStarted, MA_FALSE); return MA_SUCCESS; } static ma_bool32 ma_device_is_started__null(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); return ma_atomic_bool32_get(&pDevice->null_device.isStarted); } static ma_result ma_device_write__null(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { ma_result result = MA_SUCCESS; ma_uint32 totalPCMFramesProcessed; ma_bool32 wasStartedOnEntry; if (pFramesWritten != NULL) { *pFramesWritten = 0; } wasStartedOnEntry = ma_device_is_started__null(pDevice); /* Keep going until everything has been read. */ totalPCMFramesProcessed = 0; while (totalPCMFramesProcessed < frameCount) { ma_uint64 targetFrame; /* If there are any frames remaining in the current period, consume those first. */ if (pDevice->null_device.currentPeriodFramesRemainingPlayback > 0) { ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed); ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingPlayback; if (framesToProcess > framesRemaining) { framesToProcess = framesRemaining; } /* We don't actually do anything with pPCMFrames, so just mark it as unused to prevent a warning. */ (void)pPCMFrames; pDevice->null_device.currentPeriodFramesRemainingPlayback -= framesToProcess; totalPCMFramesProcessed += framesToProcess; } /* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */ if (pDevice->null_device.currentPeriodFramesRemainingPlayback == 0) { pDevice->null_device.currentPeriodFramesRemainingPlayback = 0; if (!ma_device_is_started__null(pDevice) && !wasStartedOnEntry) { result = ma_device_start__null(pDevice); if (result != MA_SUCCESS) { break; } } } /* If we've consumed the whole buffer we can return now. */ MA_ASSERT(totalPCMFramesProcessed <= frameCount); if (totalPCMFramesProcessed == frameCount) { break; } /* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */ targetFrame = pDevice->null_device.lastProcessedFramePlayback; for (;;) { ma_uint64 currentFrame; /* Stop waiting if the device has been stopped. */ if (!ma_device_is_started__null(pDevice)) { break; } currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice); if (currentFrame >= targetFrame) { break; } /* Getting here means we haven't yet reached the target sample, so continue waiting. */ ma_sleep(10); } pDevice->null_device.lastProcessedFramePlayback += pDevice->playback.internalPeriodSizeInFrames; pDevice->null_device.currentPeriodFramesRemainingPlayback = pDevice->playback.internalPeriodSizeInFrames; } if (pFramesWritten != NULL) { *pFramesWritten = totalPCMFramesProcessed; } return result; } static ma_result ma_device_read__null(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) { ma_result result = MA_SUCCESS; ma_uint32 totalPCMFramesProcessed; if (pFramesRead != NULL) { *pFramesRead = 0; } /* Keep going until everything has been read. */ totalPCMFramesProcessed = 0; while (totalPCMFramesProcessed < frameCount) { ma_uint64 targetFrame; /* If there are any frames remaining in the current period, consume those first. */ if (pDevice->null_device.currentPeriodFramesRemainingCapture > 0) { ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed); ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingCapture; if (framesToProcess > framesRemaining) { framesToProcess = framesRemaining; } /* We need to ensure the output buffer is zeroed. */ MA_ZERO_MEMORY(ma_offset_ptr(pPCMFrames, totalPCMFramesProcessed*bpf), framesToProcess*bpf); pDevice->null_device.currentPeriodFramesRemainingCapture -= framesToProcess; totalPCMFramesProcessed += framesToProcess; } /* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */ if (pDevice->null_device.currentPeriodFramesRemainingCapture == 0) { pDevice->null_device.currentPeriodFramesRemainingCapture = 0; } /* If we've consumed the whole buffer we can return now. */ MA_ASSERT(totalPCMFramesProcessed <= frameCount); if (totalPCMFramesProcessed == frameCount) { break; } /* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */ targetFrame = pDevice->null_device.lastProcessedFrameCapture + pDevice->capture.internalPeriodSizeInFrames; for (;;) { ma_uint64 currentFrame; /* Stop waiting if the device has been stopped. */ if (!ma_device_is_started__null(pDevice)) { break; } currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice); if (currentFrame >= targetFrame) { break; } /* Getting here means we haven't yet reached the target sample, so continue waiting. */ ma_sleep(10); } pDevice->null_device.lastProcessedFrameCapture += pDevice->capture.internalPeriodSizeInFrames; pDevice->null_device.currentPeriodFramesRemainingCapture = pDevice->capture.internalPeriodSizeInFrames; } if (pFramesRead != NULL) { *pFramesRead = totalPCMFramesProcessed; } return result; } static ma_result ma_context_uninit__null(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_null); (void)pContext; return MA_SUCCESS; } static ma_result ma_context_init__null(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { MA_ASSERT(pContext != NULL); (void)pConfig; (void)pContext; pCallbacks->onContextInit = ma_context_init__null; pCallbacks->onContextUninit = ma_context_uninit__null; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__null; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__null; pCallbacks->onDeviceInit = ma_device_init__null; pCallbacks->onDeviceUninit = ma_device_uninit__null; pCallbacks->onDeviceStart = ma_device_start__null; pCallbacks->onDeviceStop = ma_device_stop__null; pCallbacks->onDeviceRead = ma_device_read__null; pCallbacks->onDeviceWrite = ma_device_write__null; pCallbacks->onDeviceDataLoop = NULL; /* Our backend is asynchronous with a blocking read-write API which means we can get miniaudio to deal with the audio thread. */ /* The null backend always works. */ return MA_SUCCESS; } #endif /******************************************************************************* WIN32 COMMON *******************************************************************************/ #if defined(MA_WIN32) && !defined(MA_XBOX) #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) #define ma_CoInitializeEx(pContext, pvReserved, dwCoInit) ((pContext->win32.CoInitializeEx) ? ((MA_PFN_CoInitializeEx)pContext->win32.CoInitializeEx)(pvReserved, dwCoInit) : ((MA_PFN_CoInitialize)pContext->win32.CoInitialize)(pvReserved)) #define ma_CoUninitialize(pContext) ((MA_PFN_CoUninitialize)pContext->win32.CoUninitialize)() #define ma_CoCreateInstance(pContext, rclsid, pUnkOuter, dwClsContext, riid, ppv) ((MA_PFN_CoCreateInstance)pContext->win32.CoCreateInstance)(rclsid, pUnkOuter, dwClsContext, riid, ppv) #define ma_CoTaskMemFree(pContext, pv) ((MA_PFN_CoTaskMemFree)pContext->win32.CoTaskMemFree)(pv) #define ma_PropVariantClear(pContext, pvar) ((MA_PFN_PropVariantClear)pContext->win32.PropVariantClear)(pvar) #else #define ma_CoInitializeEx(pContext, pvReserved, dwCoInit) CoInitializeEx(pvReserved, dwCoInit) #define ma_CoUninitialize(pContext) CoUninitialize() #define ma_CoCreateInstance(pContext, rclsid, pUnkOuter, dwClsContext, riid, ppv) CoCreateInstance(rclsid, pUnkOuter, dwClsContext, riid, ppv) #define ma_CoTaskMemFree(pContext, pv) CoTaskMemFree(pv) #define ma_PropVariantClear(pContext, pvar) PropVariantClear(pvar) #endif #if !defined(MAXULONG_PTR) && !defined(__WATCOMC__) && !defined(MA_XBOX_NXDK) typedef size_t DWORD_PTR; #endif #if !defined(WAVE_FORMAT_1M08) #define WAVE_FORMAT_1M08 0x00000001 #define WAVE_FORMAT_1S08 0x00000002 #define WAVE_FORMAT_1M16 0x00000004 #define WAVE_FORMAT_1S16 0x00000008 #define WAVE_FORMAT_2M08 0x00000010 #define WAVE_FORMAT_2S08 0x00000020 #define WAVE_FORMAT_2M16 0x00000040 #define WAVE_FORMAT_2S16 0x00000080 #define WAVE_FORMAT_4M08 0x00000100 #define WAVE_FORMAT_4S08 0x00000200 #define WAVE_FORMAT_4M16 0x00000400 #define WAVE_FORMAT_4S16 0x00000800 #endif #if !defined(WAVE_FORMAT_44M08) #define WAVE_FORMAT_44M08 0x00000100 #define WAVE_FORMAT_44S08 0x00000200 #define WAVE_FORMAT_44M16 0x00000400 #define WAVE_FORMAT_44S16 0x00000800 #define WAVE_FORMAT_48M08 0x00001000 #define WAVE_FORMAT_48S08 0x00002000 #define WAVE_FORMAT_48M16 0x00004000 #define WAVE_FORMAT_48S16 0x00008000 #define WAVE_FORMAT_96M08 0x00010000 #define WAVE_FORMAT_96S08 0x00020000 #define WAVE_FORMAT_96M16 0x00040000 #define WAVE_FORMAT_96S16 0x00080000 #endif #ifndef SPEAKER_FRONT_LEFT #define SPEAKER_FRONT_LEFT 0x1 #define SPEAKER_FRONT_RIGHT 0x2 #define SPEAKER_FRONT_CENTER 0x4 #define SPEAKER_LOW_FREQUENCY 0x8 #define SPEAKER_BACK_LEFT 0x10 #define SPEAKER_BACK_RIGHT 0x20 #define SPEAKER_FRONT_LEFT_OF_CENTER 0x40 #define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80 #define SPEAKER_BACK_CENTER 0x100 #define SPEAKER_SIDE_LEFT 0x200 #define SPEAKER_SIDE_RIGHT 0x400 #define SPEAKER_TOP_CENTER 0x800 #define SPEAKER_TOP_FRONT_LEFT 0x1000 #define SPEAKER_TOP_FRONT_CENTER 0x2000 #define SPEAKER_TOP_FRONT_RIGHT 0x4000 #define SPEAKER_TOP_BACK_LEFT 0x8000 #define SPEAKER_TOP_BACK_CENTER 0x10000 #define SPEAKER_TOP_BACK_RIGHT 0x20000 #endif /* Implement our own version of MA_WAVEFORMATEXTENSIBLE so we can avoid a header. Be careful with this because MA_WAVEFORMATEX has an extra two bytes over standard WAVEFORMATEX due to padding. The standard version uses tight packing, but for compiler compatibility we're not doing that with ours. */ typedef struct { WORD wFormatTag; WORD nChannels; DWORD nSamplesPerSec; DWORD nAvgBytesPerSec; WORD nBlockAlign; WORD wBitsPerSample; WORD cbSize; } MA_WAVEFORMATEX; typedef struct { WORD wFormatTag; WORD nChannels; DWORD nSamplesPerSec; DWORD nAvgBytesPerSec; WORD nBlockAlign; WORD wBitsPerSample; WORD cbSize; union { WORD wValidBitsPerSample; WORD wSamplesPerBlock; WORD wReserved; } Samples; DWORD dwChannelMask; GUID SubFormat; } MA_WAVEFORMATEXTENSIBLE; #ifndef WAVE_FORMAT_EXTENSIBLE #define WAVE_FORMAT_EXTENSIBLE 0xFFFE #endif #ifndef WAVE_FORMAT_PCM #define WAVE_FORMAT_PCM 1 #endif #ifndef WAVE_FORMAT_IEEE_FLOAT #define WAVE_FORMAT_IEEE_FLOAT 0x0003 #endif /* Converts an individual Win32-style channel identifier (SPEAKER_FRONT_LEFT, etc.) to miniaudio. */ static ma_uint8 ma_channel_id_to_ma__win32(DWORD id) { switch (id) { case SPEAKER_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT; case SPEAKER_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT; case SPEAKER_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER; case SPEAKER_LOW_FREQUENCY: return MA_CHANNEL_LFE; case SPEAKER_BACK_LEFT: return MA_CHANNEL_BACK_LEFT; case SPEAKER_BACK_RIGHT: return MA_CHANNEL_BACK_RIGHT; case SPEAKER_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER; case SPEAKER_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER; case SPEAKER_BACK_CENTER: return MA_CHANNEL_BACK_CENTER; case SPEAKER_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT; case SPEAKER_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT; case SPEAKER_TOP_CENTER: return MA_CHANNEL_TOP_CENTER; case SPEAKER_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT; case SPEAKER_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER; case SPEAKER_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT; case SPEAKER_TOP_BACK_LEFT: return MA_CHANNEL_TOP_BACK_LEFT; case SPEAKER_TOP_BACK_CENTER: return MA_CHANNEL_TOP_BACK_CENTER; case SPEAKER_TOP_BACK_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT; default: return 0; } } /* Converts an individual miniaudio channel identifier (MA_CHANNEL_FRONT_LEFT, etc.) to Win32-style. */ static DWORD ma_channel_id_to_win32(DWORD id) { switch (id) { case MA_CHANNEL_MONO: return SPEAKER_FRONT_CENTER; case MA_CHANNEL_FRONT_LEFT: return SPEAKER_FRONT_LEFT; case MA_CHANNEL_FRONT_RIGHT: return SPEAKER_FRONT_RIGHT; case MA_CHANNEL_FRONT_CENTER: return SPEAKER_FRONT_CENTER; case MA_CHANNEL_LFE: return SPEAKER_LOW_FREQUENCY; case MA_CHANNEL_BACK_LEFT: return SPEAKER_BACK_LEFT; case MA_CHANNEL_BACK_RIGHT: return SPEAKER_BACK_RIGHT; case MA_CHANNEL_FRONT_LEFT_CENTER: return SPEAKER_FRONT_LEFT_OF_CENTER; case MA_CHANNEL_FRONT_RIGHT_CENTER: return SPEAKER_FRONT_RIGHT_OF_CENTER; case MA_CHANNEL_BACK_CENTER: return SPEAKER_BACK_CENTER; case MA_CHANNEL_SIDE_LEFT: return SPEAKER_SIDE_LEFT; case MA_CHANNEL_SIDE_RIGHT: return SPEAKER_SIDE_RIGHT; case MA_CHANNEL_TOP_CENTER: return SPEAKER_TOP_CENTER; case MA_CHANNEL_TOP_FRONT_LEFT: return SPEAKER_TOP_FRONT_LEFT; case MA_CHANNEL_TOP_FRONT_CENTER: return SPEAKER_TOP_FRONT_CENTER; case MA_CHANNEL_TOP_FRONT_RIGHT: return SPEAKER_TOP_FRONT_RIGHT; case MA_CHANNEL_TOP_BACK_LEFT: return SPEAKER_TOP_BACK_LEFT; case MA_CHANNEL_TOP_BACK_CENTER: return SPEAKER_TOP_BACK_CENTER; case MA_CHANNEL_TOP_BACK_RIGHT: return SPEAKER_TOP_BACK_RIGHT; default: return 0; } } /* Converts a channel mapping to a Win32-style channel mask. */ static DWORD ma_channel_map_to_channel_mask__win32(const ma_channel* pChannelMap, ma_uint32 channels) { DWORD dwChannelMask = 0; ma_uint32 iChannel; for (iChannel = 0; iChannel < channels; ++iChannel) { dwChannelMask |= ma_channel_id_to_win32(pChannelMap[iChannel]); } return dwChannelMask; } /* Converts a Win32-style channel mask to a miniaudio channel map. */ static void ma_channel_mask_to_channel_map__win32(DWORD dwChannelMask, ma_uint32 channels, ma_channel* pChannelMap) { /* If the channel mask is set to 0, just assume a default Win32 channel map. */ if (dwChannelMask == 0) { ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pChannelMap, channels, channels); } else { if (channels == 1 && (dwChannelMask & SPEAKER_FRONT_CENTER) != 0) { pChannelMap[0] = MA_CHANNEL_MONO; } else { /* Just iterate over each bit. */ ma_uint32 iChannel = 0; ma_uint32 iBit; for (iBit = 0; iBit < 32 && iChannel < channels; ++iBit) { DWORD bitValue = (dwChannelMask & (1UL << iBit)); if (bitValue != 0) { /* The bit is set. */ pChannelMap[iChannel] = ma_channel_id_to_ma__win32(bitValue); iChannel += 1; } } } } } #ifdef __cplusplus static ma_bool32 ma_is_guid_equal(const void* a, const void* b) { return IsEqualGUID(*(const GUID*)a, *(const GUID*)b); } #else #define ma_is_guid_equal(a, b) IsEqualGUID((const GUID*)a, (const GUID*)b) #endif static MA_INLINE ma_bool32 ma_is_guid_null(const void* guid) { static GUID nullguid = {0x00000000, 0x0000, 0x0000, {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00}}; return ma_is_guid_equal(guid, &nullguid); } static ma_format ma_format_from_WAVEFORMATEX(const MA_WAVEFORMATEX* pWF) { MA_ASSERT(pWF != NULL); if (pWF->wFormatTag == WAVE_FORMAT_EXTENSIBLE) { const MA_WAVEFORMATEXTENSIBLE* pWFEX = (const MA_WAVEFORMATEXTENSIBLE*)pWF; if (ma_is_guid_equal(&pWFEX->SubFormat, &MA_GUID_KSDATAFORMAT_SUBTYPE_PCM)) { if (pWFEX->Samples.wValidBitsPerSample == 32) { return ma_format_s32; } if (pWFEX->Samples.wValidBitsPerSample == 24) { if (pWFEX->wBitsPerSample == 32) { return ma_format_s32; } if (pWFEX->wBitsPerSample == 24) { return ma_format_s24; } } if (pWFEX->Samples.wValidBitsPerSample == 16) { return ma_format_s16; } if (pWFEX->Samples.wValidBitsPerSample == 8) { return ma_format_u8; } } if (ma_is_guid_equal(&pWFEX->SubFormat, &MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)) { if (pWFEX->Samples.wValidBitsPerSample == 32) { return ma_format_f32; } /* if (pWFEX->Samples.wValidBitsPerSample == 64) { return ma_format_f64; } */ } } else { if (pWF->wFormatTag == WAVE_FORMAT_PCM) { if (pWF->wBitsPerSample == 32) { return ma_format_s32; } if (pWF->wBitsPerSample == 24) { return ma_format_s24; } if (pWF->wBitsPerSample == 16) { return ma_format_s16; } if (pWF->wBitsPerSample == 8) { return ma_format_u8; } } if (pWF->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) { if (pWF->wBitsPerSample == 32) { return ma_format_f32; } if (pWF->wBitsPerSample == 64) { /*return ma_format_f64;*/ } } } return ma_format_unknown; } #endif /******************************************************************************* WASAPI Backend *******************************************************************************/ #ifdef MA_HAS_WASAPI #if 0 #if defined(_MSC_VER) #pragma warning(push) #pragma warning(disable:4091) /* 'typedef ': ignored on left of '' when no variable is declared */ #endif #include <audioclient.h> #include <mmdeviceapi.h> #if defined(_MSC_VER) #pragma warning(pop) #endif #endif /* 0 */ static ma_result ma_device_reroute__wasapi(ma_device* pDevice, ma_device_type deviceType); /* Some compilers don't define VerifyVersionInfoW. Need to write this ourselves. */ #define MA_WIN32_WINNT_VISTA 0x0600 #define MA_VER_MINORVERSION 0x01 #define MA_VER_MAJORVERSION 0x02 #define MA_VER_SERVICEPACKMAJOR 0x20 #define MA_VER_GREATER_EQUAL 0x03 typedef struct { DWORD dwOSVersionInfoSize; DWORD dwMajorVersion; DWORD dwMinorVersion; DWORD dwBuildNumber; DWORD dwPlatformId; WCHAR szCSDVersion[128]; WORD wServicePackMajor; WORD wServicePackMinor; WORD wSuiteMask; BYTE wProductType; BYTE wReserved; } ma_OSVERSIONINFOEXW; typedef BOOL (WINAPI * ma_PFNVerifyVersionInfoW) (ma_OSVERSIONINFOEXW* lpVersionInfo, DWORD dwTypeMask, DWORDLONG dwlConditionMask); typedef ULONGLONG (WINAPI * ma_PFNVerSetConditionMask)(ULONGLONG dwlConditionMask, DWORD dwTypeBitMask, BYTE dwConditionMask); #ifndef PROPERTYKEY_DEFINED #define PROPERTYKEY_DEFINED #ifndef __WATCOMC__ typedef struct { GUID fmtid; DWORD pid; } PROPERTYKEY; #endif #endif /* Some compilers don't define PropVariantInit(). We just do this ourselves since it's just a memset(). */ static MA_INLINE void ma_PropVariantInit(MA_PROPVARIANT* pProp) { MA_ZERO_OBJECT(pProp); } static const PROPERTYKEY MA_PKEY_Device_FriendlyName = {{0xA45C254E, 0xDF1C, 0x4EFD, {0x80, 0x20, 0x67, 0xD1, 0x46, 0xA8, 0x50, 0xE0}}, 14}; static const PROPERTYKEY MA_PKEY_AudioEngine_DeviceFormat = {{0xF19F064D, 0x82C, 0x4E27, {0xBC, 0x73, 0x68, 0x82, 0xA1, 0xBB, 0x8E, 0x4C}}, 0}; static const IID MA_IID_IUnknown = {0x00000000, 0x0000, 0x0000, {0xC0, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x46}}; /* 00000000-0000-0000-C000-000000000046 */ #if !defined(MA_WIN32_DESKTOP) && !defined(MA_WIN32_GDK) static const IID MA_IID_IAgileObject = {0x94EA2B94, 0xE9CC, 0x49E0, {0xC0, 0xFF, 0xEE, 0x64, 0xCA, 0x8F, 0x5B, 0x90}}; /* 94EA2B94-E9CC-49E0-C0FF-EE64CA8F5B90 */ #endif static const IID MA_IID_IAudioClient = {0x1CB9AD4C, 0xDBFA, 0x4C32, {0xB1, 0x78, 0xC2, 0xF5, 0x68, 0xA7, 0x03, 0xB2}}; /* 1CB9AD4C-DBFA-4C32-B178-C2F568A703B2 = __uuidof(IAudioClient) */ static const IID MA_IID_IAudioClient2 = {0x726778CD, 0xF60A, 0x4EDA, {0x82, 0xDE, 0xE4, 0x76, 0x10, 0xCD, 0x78, 0xAA}}; /* 726778CD-F60A-4EDA-82DE-E47610CD78AA = __uuidof(IAudioClient2) */ static const IID MA_IID_IAudioClient3 = {0x7ED4EE07, 0x8E67, 0x4CD4, {0x8C, 0x1A, 0x2B, 0x7A, 0x59, 0x87, 0xAD, 0x42}}; /* 7ED4EE07-8E67-4CD4-8C1A-2B7A5987AD42 = __uuidof(IAudioClient3) */ static const IID MA_IID_IAudioRenderClient = {0xF294ACFC, 0x3146, 0x4483, {0xA7, 0xBF, 0xAD, 0xDC, 0xA7, 0xC2, 0x60, 0xE2}}; /* F294ACFC-3146-4483-A7BF-ADDCA7C260E2 = __uuidof(IAudioRenderClient) */ static const IID MA_IID_IAudioCaptureClient = {0xC8ADBD64, 0xE71E, 0x48A0, {0xA4, 0xDE, 0x18, 0x5C, 0x39, 0x5C, 0xD3, 0x17}}; /* C8ADBD64-E71E-48A0-A4DE-185C395CD317 = __uuidof(IAudioCaptureClient) */ static const IID MA_IID_IMMNotificationClient = {0x7991EEC9, 0x7E89, 0x4D85, {0x83, 0x90, 0x6C, 0x70, 0x3C, 0xEC, 0x60, 0xC0}}; /* 7991EEC9-7E89-4D85-8390-6C703CEC60C0 = __uuidof(IMMNotificationClient) */ #if !defined(MA_WIN32_DESKTOP) && !defined(MA_WIN32_GDK) static const IID MA_IID_DEVINTERFACE_AUDIO_RENDER = {0xE6327CAD, 0xDCEC, 0x4949, {0xAE, 0x8A, 0x99, 0x1E, 0x97, 0x6A, 0x79, 0xD2}}; /* E6327CAD-DCEC-4949-AE8A-991E976A79D2 */ static const IID MA_IID_DEVINTERFACE_AUDIO_CAPTURE = {0x2EEF81BE, 0x33FA, 0x4800, {0x96, 0x70, 0x1C, 0xD4, 0x74, 0x97, 0x2C, 0x3F}}; /* 2EEF81BE-33FA-4800-9670-1CD474972C3F */ static const IID MA_IID_IActivateAudioInterfaceCompletionHandler = {0x41D949AB, 0x9862, 0x444A, {0x80, 0xF6, 0xC2, 0x61, 0x33, 0x4D, 0xA5, 0xEB}}; /* 41D949AB-9862-444A-80F6-C261334DA5EB */ #endif static const IID MA_CLSID_MMDeviceEnumerator = {0xBCDE0395, 0xE52F, 0x467C, {0x8E, 0x3D, 0xC4, 0x57, 0x92, 0x91, 0x69, 0x2E}}; /* BCDE0395-E52F-467C-8E3D-C4579291692E = __uuidof(MMDeviceEnumerator) */ static const IID MA_IID_IMMDeviceEnumerator = {0xA95664D2, 0x9614, 0x4F35, {0xA7, 0x46, 0xDE, 0x8D, 0xB6, 0x36, 0x17, 0xE6}}; /* A95664D2-9614-4F35-A746-DE8DB63617E6 = __uuidof(IMMDeviceEnumerator) */ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) #define MA_MM_DEVICE_STATE_ACTIVE 1 #define MA_MM_DEVICE_STATE_DISABLED 2 #define MA_MM_DEVICE_STATE_NOTPRESENT 4 #define MA_MM_DEVICE_STATE_UNPLUGGED 8 typedef struct ma_IMMDeviceEnumerator ma_IMMDeviceEnumerator; typedef struct ma_IMMDeviceCollection ma_IMMDeviceCollection; typedef struct ma_IMMDevice ma_IMMDevice; #else typedef struct ma_IActivateAudioInterfaceCompletionHandler ma_IActivateAudioInterfaceCompletionHandler; typedef struct ma_IActivateAudioInterfaceAsyncOperation ma_IActivateAudioInterfaceAsyncOperation; #endif typedef struct ma_IPropertyStore ma_IPropertyStore; typedef struct ma_IAudioClient ma_IAudioClient; typedef struct ma_IAudioClient2 ma_IAudioClient2; typedef struct ma_IAudioClient3 ma_IAudioClient3; typedef struct ma_IAudioRenderClient ma_IAudioRenderClient; typedef struct ma_IAudioCaptureClient ma_IAudioCaptureClient; typedef ma_int64 MA_REFERENCE_TIME; #define MA_AUDCLNT_STREAMFLAGS_CROSSPROCESS 0x00010000 #define MA_AUDCLNT_STREAMFLAGS_LOOPBACK 0x00020000 #define MA_AUDCLNT_STREAMFLAGS_EVENTCALLBACK 0x00040000 #define MA_AUDCLNT_STREAMFLAGS_NOPERSIST 0x00080000 #define MA_AUDCLNT_STREAMFLAGS_RATEADJUST 0x00100000 #define MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY 0x08000000 #define MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM 0x80000000 #define MA_AUDCLNT_SESSIONFLAGS_EXPIREWHENUNOWNED 0x10000000 #define MA_AUDCLNT_SESSIONFLAGS_DISPLAY_HIDE 0x20000000 #define MA_AUDCLNT_SESSIONFLAGS_DISPLAY_HIDEWHENEXPIRED 0x40000000 /* Buffer flags. */ #define MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY 1 #define MA_AUDCLNT_BUFFERFLAGS_SILENT 2 #define MA_AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR 4 typedef enum { ma_eRender = 0, ma_eCapture = 1, ma_eAll = 2 } ma_EDataFlow; typedef enum { ma_eConsole = 0, ma_eMultimedia = 1, ma_eCommunications = 2 } ma_ERole; typedef enum { MA_AUDCLNT_SHAREMODE_SHARED, MA_AUDCLNT_SHAREMODE_EXCLUSIVE } MA_AUDCLNT_SHAREMODE; typedef enum { MA_AudioCategory_Other = 0 /* <-- miniaudio is only caring about Other. */ } MA_AUDIO_STREAM_CATEGORY; typedef enum { MA_AUDCLNT_STREAMOPTIONS_NONE, MA_AUDCLNT_STREAMOPTIONS_RAW, MA_AUDCLNT_STREAMOPTIONS_MATCH_FORMAT, MA_AUDCLNT_STREAMOPTIONS_AMBISONICS, MA_AUDCLNT_STREAMOPTIONS_POST_VOLUME_LOOPBACK } MA_AUDCLNT_STREAMOPTIONS; typedef struct { ma_uint32 cbSize; BOOL bIsOffload; MA_AUDIO_STREAM_CATEGORY eCategory; MA_AUDCLNT_STREAMOPTIONS Options; } ma_AudioClientProperties; /* IUnknown */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IUnknown* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IUnknown* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IUnknown* pThis); } ma_IUnknownVtbl; struct ma_IUnknown { ma_IUnknownVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IUnknown_QueryInterface(ma_IUnknown* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IUnknown_AddRef(ma_IUnknown* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IUnknown_Release(ma_IUnknown* pThis) { return pThis->lpVtbl->Release(pThis); } #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) /* IMMNotificationClient */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMNotificationClient* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMNotificationClient* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IMMNotificationClient* pThis); /* IMMNotificationClient */ HRESULT (STDMETHODCALLTYPE * OnDeviceStateChanged) (ma_IMMNotificationClient* pThis, const WCHAR* pDeviceID, DWORD dwNewState); HRESULT (STDMETHODCALLTYPE * OnDeviceAdded) (ma_IMMNotificationClient* pThis, const WCHAR* pDeviceID); HRESULT (STDMETHODCALLTYPE * OnDeviceRemoved) (ma_IMMNotificationClient* pThis, const WCHAR* pDeviceID); HRESULT (STDMETHODCALLTYPE * OnDefaultDeviceChanged)(ma_IMMNotificationClient* pThis, ma_EDataFlow dataFlow, ma_ERole role, const WCHAR* pDefaultDeviceID); HRESULT (STDMETHODCALLTYPE * OnPropertyValueChanged)(ma_IMMNotificationClient* pThis, const WCHAR* pDeviceID, const PROPERTYKEY key); } ma_IMMNotificationClientVtbl; /* IMMDeviceEnumerator */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDeviceEnumerator* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDeviceEnumerator* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDeviceEnumerator* pThis); /* IMMDeviceEnumerator */ HRESULT (STDMETHODCALLTYPE * EnumAudioEndpoints) (ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, DWORD dwStateMask, ma_IMMDeviceCollection** ppDevices); HRESULT (STDMETHODCALLTYPE * GetDefaultAudioEndpoint) (ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, ma_ERole role, ma_IMMDevice** ppEndpoint); HRESULT (STDMETHODCALLTYPE * GetDevice) (ma_IMMDeviceEnumerator* pThis, const WCHAR* pID, ma_IMMDevice** ppDevice); HRESULT (STDMETHODCALLTYPE * RegisterEndpointNotificationCallback) (ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient); HRESULT (STDMETHODCALLTYPE * UnregisterEndpointNotificationCallback)(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient); } ma_IMMDeviceEnumeratorVtbl; struct ma_IMMDeviceEnumerator { ma_IMMDeviceEnumeratorVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IMMDeviceEnumerator_QueryInterface(ma_IMMDeviceEnumerator* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IMMDeviceEnumerator_AddRef(ma_IMMDeviceEnumerator* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IMMDeviceEnumerator_Release(ma_IMMDeviceEnumerator* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IMMDeviceEnumerator_EnumAudioEndpoints(ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, DWORD dwStateMask, ma_IMMDeviceCollection** ppDevices) { return pThis->lpVtbl->EnumAudioEndpoints(pThis, dataFlow, dwStateMask, ppDevices); } static MA_INLINE HRESULT ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, ma_ERole role, ma_IMMDevice** ppEndpoint) { return pThis->lpVtbl->GetDefaultAudioEndpoint(pThis, dataFlow, role, ppEndpoint); } static MA_INLINE HRESULT ma_IMMDeviceEnumerator_GetDevice(ma_IMMDeviceEnumerator* pThis, const WCHAR* pID, ma_IMMDevice** ppDevice) { return pThis->lpVtbl->GetDevice(pThis, pID, ppDevice); } static MA_INLINE HRESULT ma_IMMDeviceEnumerator_RegisterEndpointNotificationCallback(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient) { return pThis->lpVtbl->RegisterEndpointNotificationCallback(pThis, pClient); } static MA_INLINE HRESULT ma_IMMDeviceEnumerator_UnregisterEndpointNotificationCallback(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient) { return pThis->lpVtbl->UnregisterEndpointNotificationCallback(pThis, pClient); } /* IMMDeviceCollection */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDeviceCollection* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDeviceCollection* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDeviceCollection* pThis); /* IMMDeviceCollection */ HRESULT (STDMETHODCALLTYPE * GetCount)(ma_IMMDeviceCollection* pThis, UINT* pDevices); HRESULT (STDMETHODCALLTYPE * Item) (ma_IMMDeviceCollection* pThis, UINT nDevice, ma_IMMDevice** ppDevice); } ma_IMMDeviceCollectionVtbl; struct ma_IMMDeviceCollection { ma_IMMDeviceCollectionVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IMMDeviceCollection_QueryInterface(ma_IMMDeviceCollection* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IMMDeviceCollection_AddRef(ma_IMMDeviceCollection* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IMMDeviceCollection_Release(ma_IMMDeviceCollection* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IMMDeviceCollection_GetCount(ma_IMMDeviceCollection* pThis, UINT* pDevices) { return pThis->lpVtbl->GetCount(pThis, pDevices); } static MA_INLINE HRESULT ma_IMMDeviceCollection_Item(ma_IMMDeviceCollection* pThis, UINT nDevice, ma_IMMDevice** ppDevice) { return pThis->lpVtbl->Item(pThis, nDevice, ppDevice); } /* IMMDevice */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDevice* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDevice* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDevice* pThis); /* IMMDevice */ HRESULT (STDMETHODCALLTYPE * Activate) (ma_IMMDevice* pThis, const IID* const iid, DWORD dwClsCtx, MA_PROPVARIANT* pActivationParams, void** ppInterface); HRESULT (STDMETHODCALLTYPE * OpenPropertyStore)(ma_IMMDevice* pThis, DWORD stgmAccess, ma_IPropertyStore** ppProperties); HRESULT (STDMETHODCALLTYPE * GetId) (ma_IMMDevice* pThis, WCHAR** pID); HRESULT (STDMETHODCALLTYPE * GetState) (ma_IMMDevice* pThis, DWORD *pState); } ma_IMMDeviceVtbl; struct ma_IMMDevice { ma_IMMDeviceVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IMMDevice_QueryInterface(ma_IMMDevice* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IMMDevice_AddRef(ma_IMMDevice* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IMMDevice_Release(ma_IMMDevice* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IMMDevice_Activate(ma_IMMDevice* pThis, const IID* const iid, DWORD dwClsCtx, MA_PROPVARIANT* pActivationParams, void** ppInterface) { return pThis->lpVtbl->Activate(pThis, iid, dwClsCtx, pActivationParams, ppInterface); } static MA_INLINE HRESULT ma_IMMDevice_OpenPropertyStore(ma_IMMDevice* pThis, DWORD stgmAccess, ma_IPropertyStore** ppProperties) { return pThis->lpVtbl->OpenPropertyStore(pThis, stgmAccess, ppProperties); } static MA_INLINE HRESULT ma_IMMDevice_GetId(ma_IMMDevice* pThis, WCHAR** pID) { return pThis->lpVtbl->GetId(pThis, pID); } static MA_INLINE HRESULT ma_IMMDevice_GetState(ma_IMMDevice* pThis, DWORD *pState) { return pThis->lpVtbl->GetState(pThis, pState); } #else /* IActivateAudioInterfaceAsyncOperation */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IActivateAudioInterfaceAsyncOperation* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IActivateAudioInterfaceAsyncOperation* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IActivateAudioInterfaceAsyncOperation* pThis); /* IActivateAudioInterfaceAsyncOperation */ HRESULT (STDMETHODCALLTYPE * GetActivateResult)(ma_IActivateAudioInterfaceAsyncOperation* pThis, HRESULT *pActivateResult, ma_IUnknown** ppActivatedInterface); } ma_IActivateAudioInterfaceAsyncOperationVtbl; struct ma_IActivateAudioInterfaceAsyncOperation { ma_IActivateAudioInterfaceAsyncOperationVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IActivateAudioInterfaceAsyncOperation_QueryInterface(ma_IActivateAudioInterfaceAsyncOperation* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IActivateAudioInterfaceAsyncOperation_AddRef(ma_IActivateAudioInterfaceAsyncOperation* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IActivateAudioInterfaceAsyncOperation_Release(ma_IActivateAudioInterfaceAsyncOperation* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IActivateAudioInterfaceAsyncOperation_GetActivateResult(ma_IActivateAudioInterfaceAsyncOperation* pThis, HRESULT *pActivateResult, ma_IUnknown** ppActivatedInterface) { return pThis->lpVtbl->GetActivateResult(pThis, pActivateResult, ppActivatedInterface); } #endif /* IPropertyStore */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IPropertyStore* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IPropertyStore* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IPropertyStore* pThis); /* IPropertyStore */ HRESULT (STDMETHODCALLTYPE * GetCount)(ma_IPropertyStore* pThis, DWORD* pPropCount); HRESULT (STDMETHODCALLTYPE * GetAt) (ma_IPropertyStore* pThis, DWORD propIndex, PROPERTYKEY* pPropKey); HRESULT (STDMETHODCALLTYPE * GetValue)(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, MA_PROPVARIANT* pPropVar); HRESULT (STDMETHODCALLTYPE * SetValue)(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, const MA_PROPVARIANT* const pPropVar); HRESULT (STDMETHODCALLTYPE * Commit) (ma_IPropertyStore* pThis); } ma_IPropertyStoreVtbl; struct ma_IPropertyStore { ma_IPropertyStoreVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IPropertyStore_QueryInterface(ma_IPropertyStore* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IPropertyStore_AddRef(ma_IPropertyStore* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IPropertyStore_Release(ma_IPropertyStore* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IPropertyStore_GetCount(ma_IPropertyStore* pThis, DWORD* pPropCount) { return pThis->lpVtbl->GetCount(pThis, pPropCount); } static MA_INLINE HRESULT ma_IPropertyStore_GetAt(ma_IPropertyStore* pThis, DWORD propIndex, PROPERTYKEY* pPropKey) { return pThis->lpVtbl->GetAt(pThis, propIndex, pPropKey); } static MA_INLINE HRESULT ma_IPropertyStore_GetValue(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, MA_PROPVARIANT* pPropVar) { return pThis->lpVtbl->GetValue(pThis, pKey, pPropVar); } static MA_INLINE HRESULT ma_IPropertyStore_SetValue(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, const MA_PROPVARIANT* const pPropVar) { return pThis->lpVtbl->SetValue(pThis, pKey, pPropVar); } static MA_INLINE HRESULT ma_IPropertyStore_Commit(ma_IPropertyStore* pThis) { return pThis->lpVtbl->Commit(pThis); } /* IAudioClient */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient* pThis); /* IAudioClient */ HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const MA_WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient* pThis, ma_uint32* pNumBufferFrames); HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient* pThis, MA_REFERENCE_TIME* pLatency); HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient* pThis, ma_uint32* pNumPaddingFrames); HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, const MA_WAVEFORMATEX* pFormat, MA_WAVEFORMATEX** ppClosestMatch); HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient* pThis, MA_WAVEFORMATEX** ppDeviceFormat); HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod); HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient* pThis); HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient* pThis); HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient* pThis); HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient* pThis, HANDLE eventHandle); HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient* pThis, const IID* const riid, void** pp); } ma_IAudioClientVtbl; struct ma_IAudioClient { ma_IAudioClientVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IAudioClient_QueryInterface(ma_IAudioClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IAudioClient_AddRef(ma_IAudioClient* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IAudioClient_Release(ma_IAudioClient* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IAudioClient_Initialize(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const MA_WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); } static MA_INLINE HRESULT ma_IAudioClient_GetBufferSize(ma_IAudioClient* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); } static MA_INLINE HRESULT ma_IAudioClient_GetStreamLatency(ma_IAudioClient* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); } static MA_INLINE HRESULT ma_IAudioClient_GetCurrentPadding(ma_IAudioClient* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); } static MA_INLINE HRESULT ma_IAudioClient_IsFormatSupported(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, const MA_WAVEFORMATEX* pFormat, MA_WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); } static MA_INLINE HRESULT ma_IAudioClient_GetMixFormat(ma_IAudioClient* pThis, MA_WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); } static MA_INLINE HRESULT ma_IAudioClient_GetDevicePeriod(ma_IAudioClient* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); } static MA_INLINE HRESULT ma_IAudioClient_Start(ma_IAudioClient* pThis) { return pThis->lpVtbl->Start(pThis); } static MA_INLINE HRESULT ma_IAudioClient_Stop(ma_IAudioClient* pThis) { return pThis->lpVtbl->Stop(pThis); } static MA_INLINE HRESULT ma_IAudioClient_Reset(ma_IAudioClient* pThis) { return pThis->lpVtbl->Reset(pThis); } static MA_INLINE HRESULT ma_IAudioClient_SetEventHandle(ma_IAudioClient* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); } static MA_INLINE HRESULT ma_IAudioClient_GetService(ma_IAudioClient* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); } /* IAudioClient2 */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient2* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient2* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient2* pThis); /* IAudioClient */ HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const MA_WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient2* pThis, ma_uint32* pNumBufferFrames); HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pLatency); HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient2* pThis, ma_uint32* pNumPaddingFrames); HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, const MA_WAVEFORMATEX* pFormat, MA_WAVEFORMATEX** ppClosestMatch); HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient2* pThis, MA_WAVEFORMATEX** ppDeviceFormat); HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod); HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient2* pThis); HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient2* pThis); HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient2* pThis); HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient2* pThis, HANDLE eventHandle); HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient2* pThis, const IID* const riid, void** pp); /* IAudioClient2 */ HRESULT (STDMETHODCALLTYPE * IsOffloadCapable) (ma_IAudioClient2* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable); HRESULT (STDMETHODCALLTYPE * SetClientProperties)(ma_IAudioClient2* pThis, const ma_AudioClientProperties* pProperties); HRESULT (STDMETHODCALLTYPE * GetBufferSizeLimits)(ma_IAudioClient2* pThis, const MA_WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration); } ma_IAudioClient2Vtbl; struct ma_IAudioClient2 { ma_IAudioClient2Vtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IAudioClient2_QueryInterface(ma_IAudioClient2* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IAudioClient2_AddRef(ma_IAudioClient2* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IAudioClient2_Release(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IAudioClient2_Initialize(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const MA_WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); } static MA_INLINE HRESULT ma_IAudioClient2_GetBufferSize(ma_IAudioClient2* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); } static MA_INLINE HRESULT ma_IAudioClient2_GetStreamLatency(ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); } static MA_INLINE HRESULT ma_IAudioClient2_GetCurrentPadding(ma_IAudioClient2* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); } static MA_INLINE HRESULT ma_IAudioClient2_IsFormatSupported(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, const MA_WAVEFORMATEX* pFormat, MA_WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); } static MA_INLINE HRESULT ma_IAudioClient2_GetMixFormat(ma_IAudioClient2* pThis, MA_WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); } static MA_INLINE HRESULT ma_IAudioClient2_GetDevicePeriod(ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); } static MA_INLINE HRESULT ma_IAudioClient2_Start(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Start(pThis); } static MA_INLINE HRESULT ma_IAudioClient2_Stop(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Stop(pThis); } static MA_INLINE HRESULT ma_IAudioClient2_Reset(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Reset(pThis); } static MA_INLINE HRESULT ma_IAudioClient2_SetEventHandle(ma_IAudioClient2* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); } static MA_INLINE HRESULT ma_IAudioClient2_GetService(ma_IAudioClient2* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); } static MA_INLINE HRESULT ma_IAudioClient2_IsOffloadCapable(ma_IAudioClient2* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable) { return pThis->lpVtbl->IsOffloadCapable(pThis, category, pOffloadCapable); } static MA_INLINE HRESULT ma_IAudioClient2_SetClientProperties(ma_IAudioClient2* pThis, const ma_AudioClientProperties* pProperties) { return pThis->lpVtbl->SetClientProperties(pThis, pProperties); } static MA_INLINE HRESULT ma_IAudioClient2_GetBufferSizeLimits(ma_IAudioClient2* pThis, const MA_WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration) { return pThis->lpVtbl->GetBufferSizeLimits(pThis, pFormat, eventDriven, pMinBufferDuration, pMaxBufferDuration); } /* IAudioClient3 */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient3* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient3* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient3* pThis); /* IAudioClient */ HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const MA_WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient3* pThis, ma_uint32* pNumBufferFrames); HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pLatency); HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient3* pThis, ma_uint32* pNumPaddingFrames); HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, const MA_WAVEFORMATEX* pFormat, MA_WAVEFORMATEX** ppClosestMatch); HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient3* pThis, MA_WAVEFORMATEX** ppDeviceFormat); HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod); HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient3* pThis); HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient3* pThis); HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient3* pThis); HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient3* pThis, HANDLE eventHandle); HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient3* pThis, const IID* const riid, void** pp); /* IAudioClient2 */ HRESULT (STDMETHODCALLTYPE * IsOffloadCapable) (ma_IAudioClient3* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable); HRESULT (STDMETHODCALLTYPE * SetClientProperties)(ma_IAudioClient3* pThis, const ma_AudioClientProperties* pProperties); HRESULT (STDMETHODCALLTYPE * GetBufferSizeLimits)(ma_IAudioClient3* pThis, const MA_WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration); /* IAudioClient3 */ HRESULT (STDMETHODCALLTYPE * GetSharedModeEnginePeriod) (ma_IAudioClient3* pThis, const MA_WAVEFORMATEX* pFormat, ma_uint32* pDefaultPeriodInFrames, ma_uint32* pFundamentalPeriodInFrames, ma_uint32* pMinPeriodInFrames, ma_uint32* pMaxPeriodInFrames); HRESULT (STDMETHODCALLTYPE * GetCurrentSharedModeEnginePeriod)(ma_IAudioClient3* pThis, MA_WAVEFORMATEX** ppFormat, ma_uint32* pCurrentPeriodInFrames); HRESULT (STDMETHODCALLTYPE * InitializeSharedAudioStream) (ma_IAudioClient3* pThis, DWORD streamFlags, ma_uint32 periodInFrames, const MA_WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); } ma_IAudioClient3Vtbl; struct ma_IAudioClient3 { ma_IAudioClient3Vtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IAudioClient3_QueryInterface(ma_IAudioClient3* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IAudioClient3_AddRef(ma_IAudioClient3* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IAudioClient3_Release(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IAudioClient3_Initialize(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const MA_WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); } static MA_INLINE HRESULT ma_IAudioClient3_GetBufferSize(ma_IAudioClient3* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); } static MA_INLINE HRESULT ma_IAudioClient3_GetStreamLatency(ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); } static MA_INLINE HRESULT ma_IAudioClient3_GetCurrentPadding(ma_IAudioClient3* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); } static MA_INLINE HRESULT ma_IAudioClient3_IsFormatSupported(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, const MA_WAVEFORMATEX* pFormat, MA_WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); } static MA_INLINE HRESULT ma_IAudioClient3_GetMixFormat(ma_IAudioClient3* pThis, MA_WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); } static MA_INLINE HRESULT ma_IAudioClient3_GetDevicePeriod(ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); } static MA_INLINE HRESULT ma_IAudioClient3_Start(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Start(pThis); } static MA_INLINE HRESULT ma_IAudioClient3_Stop(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Stop(pThis); } static MA_INLINE HRESULT ma_IAudioClient3_Reset(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Reset(pThis); } static MA_INLINE HRESULT ma_IAudioClient3_SetEventHandle(ma_IAudioClient3* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); } static MA_INLINE HRESULT ma_IAudioClient3_GetService(ma_IAudioClient3* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); } static MA_INLINE HRESULT ma_IAudioClient3_IsOffloadCapable(ma_IAudioClient3* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable) { return pThis->lpVtbl->IsOffloadCapable(pThis, category, pOffloadCapable); } static MA_INLINE HRESULT ma_IAudioClient3_SetClientProperties(ma_IAudioClient3* pThis, const ma_AudioClientProperties* pProperties) { return pThis->lpVtbl->SetClientProperties(pThis, pProperties); } static MA_INLINE HRESULT ma_IAudioClient3_GetBufferSizeLimits(ma_IAudioClient3* pThis, const MA_WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration) { return pThis->lpVtbl->GetBufferSizeLimits(pThis, pFormat, eventDriven, pMinBufferDuration, pMaxBufferDuration); } static MA_INLINE HRESULT ma_IAudioClient3_GetSharedModeEnginePeriod(ma_IAudioClient3* pThis, const MA_WAVEFORMATEX* pFormat, ma_uint32* pDefaultPeriodInFrames, ma_uint32* pFundamentalPeriodInFrames, ma_uint32* pMinPeriodInFrames, ma_uint32* pMaxPeriodInFrames) { return pThis->lpVtbl->GetSharedModeEnginePeriod(pThis, pFormat, pDefaultPeriodInFrames, pFundamentalPeriodInFrames, pMinPeriodInFrames, pMaxPeriodInFrames); } static MA_INLINE HRESULT ma_IAudioClient3_GetCurrentSharedModeEnginePeriod(ma_IAudioClient3* pThis, MA_WAVEFORMATEX** ppFormat, ma_uint32* pCurrentPeriodInFrames) { return pThis->lpVtbl->GetCurrentSharedModeEnginePeriod(pThis, ppFormat, pCurrentPeriodInFrames); } static MA_INLINE HRESULT ma_IAudioClient3_InitializeSharedAudioStream(ma_IAudioClient3* pThis, DWORD streamFlags, ma_uint32 periodInFrames, const MA_WAVEFORMATEX* pFormat, const GUID* pAudioSessionGUID) { return pThis->lpVtbl->InitializeSharedAudioStream(pThis, streamFlags, periodInFrames, pFormat, pAudioSessionGUID); } /* IAudioRenderClient */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioRenderClient* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioRenderClient* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioRenderClient* pThis); /* IAudioRenderClient */ HRESULT (STDMETHODCALLTYPE * GetBuffer) (ma_IAudioRenderClient* pThis, ma_uint32 numFramesRequested, BYTE** ppData); HRESULT (STDMETHODCALLTYPE * ReleaseBuffer)(ma_IAudioRenderClient* pThis, ma_uint32 numFramesWritten, DWORD dwFlags); } ma_IAudioRenderClientVtbl; struct ma_IAudioRenderClient { ma_IAudioRenderClientVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IAudioRenderClient_QueryInterface(ma_IAudioRenderClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IAudioRenderClient_AddRef(ma_IAudioRenderClient* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IAudioRenderClient_Release(ma_IAudioRenderClient* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IAudioRenderClient_GetBuffer(ma_IAudioRenderClient* pThis, ma_uint32 numFramesRequested, BYTE** ppData) { return pThis->lpVtbl->GetBuffer(pThis, numFramesRequested, ppData); } static MA_INLINE HRESULT ma_IAudioRenderClient_ReleaseBuffer(ma_IAudioRenderClient* pThis, ma_uint32 numFramesWritten, DWORD dwFlags) { return pThis->lpVtbl->ReleaseBuffer(pThis, numFramesWritten, dwFlags); } /* IAudioCaptureClient */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioCaptureClient* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioCaptureClient* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioCaptureClient* pThis); /* IAudioRenderClient */ HRESULT (STDMETHODCALLTYPE * GetBuffer) (ma_IAudioCaptureClient* pThis, BYTE** ppData, ma_uint32* pNumFramesToRead, DWORD* pFlags, ma_uint64* pDevicePosition, ma_uint64* pQPCPosition); HRESULT (STDMETHODCALLTYPE * ReleaseBuffer) (ma_IAudioCaptureClient* pThis, ma_uint32 numFramesRead); HRESULT (STDMETHODCALLTYPE * GetNextPacketSize)(ma_IAudioCaptureClient* pThis, ma_uint32* pNumFramesInNextPacket); } ma_IAudioCaptureClientVtbl; struct ma_IAudioCaptureClient { ma_IAudioCaptureClientVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IAudioCaptureClient_QueryInterface(ma_IAudioCaptureClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IAudioCaptureClient_AddRef(ma_IAudioCaptureClient* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IAudioCaptureClient_Release(ma_IAudioCaptureClient* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IAudioCaptureClient_GetBuffer(ma_IAudioCaptureClient* pThis, BYTE** ppData, ma_uint32* pNumFramesToRead, DWORD* pFlags, ma_uint64* pDevicePosition, ma_uint64* pQPCPosition) { return pThis->lpVtbl->GetBuffer(pThis, ppData, pNumFramesToRead, pFlags, pDevicePosition, pQPCPosition); } static MA_INLINE HRESULT ma_IAudioCaptureClient_ReleaseBuffer(ma_IAudioCaptureClient* pThis, ma_uint32 numFramesRead) { return pThis->lpVtbl->ReleaseBuffer(pThis, numFramesRead); } static MA_INLINE HRESULT ma_IAudioCaptureClient_GetNextPacketSize(ma_IAudioCaptureClient* pThis, ma_uint32* pNumFramesInNextPacket) { return pThis->lpVtbl->GetNextPacketSize(pThis, pNumFramesInNextPacket); } #if defined(MA_WIN32_UWP) /* mmdevapi Functions */ typedef HRESULT (WINAPI * MA_PFN_ActivateAudioInterfaceAsync)(const wchar_t* deviceInterfacePath, const IID* riid, MA_PROPVARIANT* activationParams, ma_IActivateAudioInterfaceCompletionHandler* completionHandler, ma_IActivateAudioInterfaceAsyncOperation** activationOperation); #endif /* Avrt Functions */ typedef HANDLE (WINAPI * MA_PFN_AvSetMmThreadCharacteristicsA)(const char* TaskName, DWORD* TaskIndex); typedef BOOL (WINAPI * MA_PFN_AvRevertMmThreadCharacteristics)(HANDLE AvrtHandle); #if !defined(MA_WIN32_DESKTOP) && !defined(MA_WIN32_GDK) typedef struct ma_completion_handler_uwp ma_completion_handler_uwp; typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_completion_handler_uwp* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_completion_handler_uwp* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_completion_handler_uwp* pThis); /* IActivateAudioInterfaceCompletionHandler */ HRESULT (STDMETHODCALLTYPE * ActivateCompleted)(ma_completion_handler_uwp* pThis, ma_IActivateAudioInterfaceAsyncOperation* pActivateOperation); } ma_completion_handler_uwp_vtbl; struct ma_completion_handler_uwp { ma_completion_handler_uwp_vtbl* lpVtbl; MA_ATOMIC(4, ma_uint32) counter; HANDLE hEvent; }; static HRESULT STDMETHODCALLTYPE ma_completion_handler_uwp_QueryInterface(ma_completion_handler_uwp* pThis, const IID* const riid, void** ppObject) { /* We need to "implement" IAgileObject which is just an indicator that's used internally by WASAPI for some multithreading management. To "implement" this, we just make sure we return pThis when the IAgileObject is requested. */ if (!ma_is_guid_equal(riid, &MA_IID_IUnknown) && !ma_is_guid_equal(riid, &MA_IID_IActivateAudioInterfaceCompletionHandler) && !ma_is_guid_equal(riid, &MA_IID_IAgileObject)) { *ppObject = NULL; return E_NOINTERFACE; } /* Getting here means the IID is IUnknown or IMMNotificationClient. */ *ppObject = (void*)pThis; ((ma_completion_handler_uwp_vtbl*)pThis->lpVtbl)->AddRef(pThis); return S_OK; } static ULONG STDMETHODCALLTYPE ma_completion_handler_uwp_AddRef(ma_completion_handler_uwp* pThis) { return (ULONG)ma_atomic_fetch_add_32(&pThis->counter, 1) + 1; } static ULONG STDMETHODCALLTYPE ma_completion_handler_uwp_Release(ma_completion_handler_uwp* pThis) { ma_uint32 newRefCount = ma_atomic_fetch_sub_32(&pThis->counter, 1) - 1; if (newRefCount == 0) { return 0; /* We don't free anything here because we never allocate the object on the heap. */ } return (ULONG)newRefCount; } static HRESULT STDMETHODCALLTYPE ma_completion_handler_uwp_ActivateCompleted(ma_completion_handler_uwp* pThis, ma_IActivateAudioInterfaceAsyncOperation* pActivateOperation) { (void)pActivateOperation; SetEvent(pThis->hEvent); return S_OK; } static ma_completion_handler_uwp_vtbl g_maCompletionHandlerVtblInstance = { ma_completion_handler_uwp_QueryInterface, ma_completion_handler_uwp_AddRef, ma_completion_handler_uwp_Release, ma_completion_handler_uwp_ActivateCompleted }; static ma_result ma_completion_handler_uwp_init(ma_completion_handler_uwp* pHandler) { MA_ASSERT(pHandler != NULL); MA_ZERO_OBJECT(pHandler); pHandler->lpVtbl = &g_maCompletionHandlerVtblInstance; pHandler->counter = 1; pHandler->hEvent = CreateEventA(NULL, FALSE, FALSE, NULL); if (pHandler->hEvent == NULL) { return ma_result_from_GetLastError(GetLastError()); } return MA_SUCCESS; } static void ma_completion_handler_uwp_uninit(ma_completion_handler_uwp* pHandler) { if (pHandler->hEvent != NULL) { CloseHandle(pHandler->hEvent); } } static void ma_completion_handler_uwp_wait(ma_completion_handler_uwp* pHandler) { WaitForSingleObject((HANDLE)pHandler->hEvent, INFINITE); } #endif /* !MA_WIN32_DESKTOP */ /* We need a virtual table for our notification client object that's used for detecting changes to the default device. */ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_QueryInterface(ma_IMMNotificationClient* pThis, const IID* const riid, void** ppObject) { /* We care about two interfaces - IUnknown and IMMNotificationClient. If the requested IID is something else we just return E_NOINTERFACE. Otherwise we need to increment the reference counter and return S_OK. */ if (!ma_is_guid_equal(riid, &MA_IID_IUnknown) && !ma_is_guid_equal(riid, &MA_IID_IMMNotificationClient)) { *ppObject = NULL; return E_NOINTERFACE; } /* Getting here means the IID is IUnknown or IMMNotificationClient. */ *ppObject = (void*)pThis; ((ma_IMMNotificationClientVtbl*)pThis->lpVtbl)->AddRef(pThis); return S_OK; } static ULONG STDMETHODCALLTYPE ma_IMMNotificationClient_AddRef(ma_IMMNotificationClient* pThis) { return (ULONG)ma_atomic_fetch_add_32(&pThis->counter, 1) + 1; } static ULONG STDMETHODCALLTYPE ma_IMMNotificationClient_Release(ma_IMMNotificationClient* pThis) { ma_uint32 newRefCount = ma_atomic_fetch_sub_32(&pThis->counter, 1) - 1; if (newRefCount == 0) { return 0; /* We don't free anything here because we never allocate the object on the heap. */ } return (ULONG)newRefCount; } static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceStateChanged(ma_IMMNotificationClient* pThis, const WCHAR* pDeviceID, DWORD dwNewState) { ma_bool32 isThisDevice = MA_FALSE; ma_bool32 isCapture = MA_FALSE; ma_bool32 isPlayback = MA_FALSE; #ifdef MA_DEBUG_OUTPUT /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDeviceStateChanged(pDeviceID=%S, dwNewState=%u)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)", (unsigned int)dwNewState);*/ #endif /* There have been reports of a hang when a playback device is disconnected. The idea with this code is to explicitly stop the device if we detect that the device is disabled or has been unplugged. */ if (pThis->pDevice->wasapi.allowCaptureAutoStreamRouting && (pThis->pDevice->type == ma_device_type_capture || pThis->pDevice->type == ma_device_type_duplex || pThis->pDevice->type == ma_device_type_loopback)) { isCapture = MA_TRUE; if (ma_strcmp_WCHAR(pThis->pDevice->capture.id.wasapi, pDeviceID) == 0) { isThisDevice = MA_TRUE; } } if (pThis->pDevice->wasapi.allowPlaybackAutoStreamRouting && (pThis->pDevice->type == ma_device_type_playback || pThis->pDevice->type == ma_device_type_duplex)) { isPlayback = MA_TRUE; if (ma_strcmp_WCHAR(pThis->pDevice->playback.id.wasapi, pDeviceID) == 0) { isThisDevice = MA_TRUE; } } /* If the device ID matches our device we need to mark our device as detached and stop it. When a device is added in OnDeviceAdded(), we'll restart it. We only mark it as detached if the device was started at the time of being removed. */ if (isThisDevice) { if ((dwNewState & MA_MM_DEVICE_STATE_ACTIVE) == 0) { /* Unplugged or otherwise unavailable. Mark as detached if we were in a playing state. We'll use this to determine whether or not we need to automatically start the device when it's plugged back in again. */ if (ma_device_get_state(pThis->pDevice) == ma_device_state_started) { if (isPlayback) { pThis->pDevice->wasapi.isDetachedPlayback = MA_TRUE; } if (isCapture) { pThis->pDevice->wasapi.isDetachedCapture = MA_TRUE; } ma_device_stop(pThis->pDevice); } } if ((dwNewState & MA_MM_DEVICE_STATE_ACTIVE) != 0) { /* The device was activated. If we were detached, we need to start it again. */ ma_bool8 tryRestartingDevice = MA_FALSE; if (isPlayback) { if (pThis->pDevice->wasapi.isDetachedPlayback) { pThis->pDevice->wasapi.isDetachedPlayback = MA_FALSE; ma_device_reroute__wasapi(pThis->pDevice, ma_device_type_playback); tryRestartingDevice = MA_TRUE; } } if (isCapture) { if (pThis->pDevice->wasapi.isDetachedCapture) { pThis->pDevice->wasapi.isDetachedCapture = MA_FALSE; ma_device_reroute__wasapi(pThis->pDevice, (pThis->pDevice->type == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture); tryRestartingDevice = MA_TRUE; } } if (tryRestartingDevice) { if (pThis->pDevice->wasapi.isDetachedPlayback == MA_FALSE && pThis->pDevice->wasapi.isDetachedCapture == MA_FALSE) { ma_device_start(pThis->pDevice); } } } } return S_OK; } static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceAdded(ma_IMMNotificationClient* pThis, const WCHAR* pDeviceID) { #ifdef MA_DEBUG_OUTPUT /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDeviceAdded(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)");*/ #endif /* We don't need to worry about this event for our purposes. */ (void)pThis; (void)pDeviceID; return S_OK; } static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceRemoved(ma_IMMNotificationClient* pThis, const WCHAR* pDeviceID) { #ifdef MA_DEBUG_OUTPUT /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDeviceRemoved(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)");*/ #endif /* We don't need to worry about this event for our purposes. */ (void)pThis; (void)pDeviceID; return S_OK; } static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDefaultDeviceChanged(ma_IMMNotificationClient* pThis, ma_EDataFlow dataFlow, ma_ERole role, const WCHAR* pDefaultDeviceID) { #ifdef MA_DEBUG_OUTPUT /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDefaultDeviceChanged(dataFlow=%d, role=%d, pDefaultDeviceID=%S)\n", dataFlow, role, (pDefaultDeviceID != NULL) ? pDefaultDeviceID : L"(NULL)");*/ #endif (void)role; /* We only care about devices with the same data flow as the current device. */ if ((pThis->pDevice->type == ma_device_type_playback && dataFlow != ma_eRender) || (pThis->pDevice->type == ma_device_type_capture && dataFlow != ma_eCapture) || (pThis->pDevice->type == ma_device_type_loopback && dataFlow != ma_eRender)) { ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting abandoned because dataFlow does match device type.\n"); return S_OK; } /* We need to consider dataFlow as ma_eCapture if device is ma_device_type_loopback */ if (pThis->pDevice->type == ma_device_type_loopback) { dataFlow = ma_eCapture; } /* Don't do automatic stream routing if we're not allowed. */ if ((dataFlow == ma_eRender && pThis->pDevice->wasapi.allowPlaybackAutoStreamRouting == MA_FALSE) || (dataFlow == ma_eCapture && pThis->pDevice->wasapi.allowCaptureAutoStreamRouting == MA_FALSE)) { ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting abandoned because automatic stream routing has been disabled by the device config.\n"); return S_OK; } /* Not currently supporting automatic stream routing in exclusive mode. This is not working correctly on my machine due to AUDCLNT_E_DEVICE_IN_USE errors when reinitializing the device. If this is a bug in miniaudio, we can try re-enabling this once it's fixed. */ if ((dataFlow == ma_eRender && pThis->pDevice->playback.shareMode == ma_share_mode_exclusive) || (dataFlow == ma_eCapture && pThis->pDevice->capture.shareMode == ma_share_mode_exclusive)) { ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting abandoned because the device shared mode is exclusive.\n"); return S_OK; } /* Second attempt at device rerouting. We're going to retrieve the device's state at the time of the route change. We're then going to stop the device, reinitialize the device, and then start it again if the state before stopping was ma_device_state_started. */ { ma_uint32 previousState = ma_device_get_state(pThis->pDevice); ma_bool8 restartDevice = MA_FALSE; if (previousState == ma_device_state_uninitialized || previousState == ma_device_state_starting) { ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting abandoned because the device is in the process of starting.\n"); return S_OK; } if (previousState == ma_device_state_started) { ma_device_stop(pThis->pDevice); restartDevice = MA_TRUE; } if (pDefaultDeviceID != NULL) { /* <-- The input device ID will be null if there's no other device available. */ ma_mutex_lock(&pThis->pDevice->wasapi.rerouteLock); { if (dataFlow == ma_eRender) { ma_device_reroute__wasapi(pThis->pDevice, ma_device_type_playback); if (pThis->pDevice->wasapi.isDetachedPlayback) { pThis->pDevice->wasapi.isDetachedPlayback = MA_FALSE; if (pThis->pDevice->type == ma_device_type_duplex && pThis->pDevice->wasapi.isDetachedCapture) { restartDevice = MA_FALSE; /* It's a duplex device and the capture side is detached. We cannot be restarting the device just yet. */ } else { restartDevice = MA_TRUE; /* It's not a duplex device, or the capture side is also attached so we can go ahead and restart the device. */ } } } else { ma_device_reroute__wasapi(pThis->pDevice, (pThis->pDevice->type == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture); if (pThis->pDevice->wasapi.isDetachedCapture) { pThis->pDevice->wasapi.isDetachedCapture = MA_FALSE; if (pThis->pDevice->type == ma_device_type_duplex && pThis->pDevice->wasapi.isDetachedPlayback) { restartDevice = MA_FALSE; /* It's a duplex device and the playback side is detached. We cannot be restarting the device just yet. */ } else { restartDevice = MA_TRUE; /* It's not a duplex device, or the playback side is also attached so we can go ahead and restart the device. */ } } } } ma_mutex_unlock(&pThis->pDevice->wasapi.rerouteLock); if (restartDevice) { ma_device_start(pThis->pDevice); } } } return S_OK; } static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnPropertyValueChanged(ma_IMMNotificationClient* pThis, const WCHAR* pDeviceID, const PROPERTYKEY key) { #ifdef MA_DEBUG_OUTPUT /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnPropertyValueChanged(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)");*/ #endif (void)pThis; (void)pDeviceID; (void)key; return S_OK; } static ma_IMMNotificationClientVtbl g_maNotificationCientVtbl = { ma_IMMNotificationClient_QueryInterface, ma_IMMNotificationClient_AddRef, ma_IMMNotificationClient_Release, ma_IMMNotificationClient_OnDeviceStateChanged, ma_IMMNotificationClient_OnDeviceAdded, ma_IMMNotificationClient_OnDeviceRemoved, ma_IMMNotificationClient_OnDefaultDeviceChanged, ma_IMMNotificationClient_OnPropertyValueChanged }; #endif /* MA_WIN32_DESKTOP */ static const char* ma_to_usage_string__wasapi(ma_wasapi_usage usage) { switch (usage) { case ma_wasapi_usage_default: return NULL; case ma_wasapi_usage_games: return "Games"; case ma_wasapi_usage_pro_audio: return "Pro Audio"; default: break; } return NULL; } #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) typedef ma_IMMDevice ma_WASAPIDeviceInterface; #else typedef ma_IUnknown ma_WASAPIDeviceInterface; #endif #define MA_CONTEXT_COMMAND_QUIT__WASAPI 1 #define MA_CONTEXT_COMMAND_CREATE_IAUDIOCLIENT__WASAPI 2 #define MA_CONTEXT_COMMAND_RELEASE_IAUDIOCLIENT__WASAPI 3 static ma_context_command__wasapi ma_context_init_command__wasapi(int code) { ma_context_command__wasapi cmd; MA_ZERO_OBJECT(&cmd); cmd.code = code; return cmd; } static ma_result ma_context_post_command__wasapi(ma_context* pContext, const ma_context_command__wasapi* pCmd) { /* For now we are doing everything synchronously, but I might relax this later if the need arises. */ ma_result result; ma_bool32 isUsingLocalEvent = MA_FALSE; ma_event localEvent; MA_ASSERT(pContext != NULL); MA_ASSERT(pCmd != NULL); if (pCmd->pEvent == NULL) { isUsingLocalEvent = MA_TRUE; result = ma_event_init(&localEvent); if (result != MA_SUCCESS) { return result; /* Failed to create the event for this command. */ } } /* Here is where we add the command to the list. If there's not enough room we'll spin until there is. */ ma_mutex_lock(&pContext->wasapi.commandLock); { ma_uint32 index; /* Spin until we've got some space available. */ while (pContext->wasapi.commandCount == ma_countof(pContext->wasapi.commands)) { ma_yield(); } /* Space is now available. Can safely add to the list. */ index = (pContext->wasapi.commandIndex + pContext->wasapi.commandCount) % ma_countof(pContext->wasapi.commands); pContext->wasapi.commands[index] = *pCmd; pContext->wasapi.commands[index].pEvent = &localEvent; pContext->wasapi.commandCount += 1; /* Now that the command has been added, release the semaphore so ma_context_next_command__wasapi() can return. */ ma_semaphore_release(&pContext->wasapi.commandSem); } ma_mutex_unlock(&pContext->wasapi.commandLock); if (isUsingLocalEvent) { ma_event_wait(&localEvent); ma_event_uninit(&localEvent); } return MA_SUCCESS; } static ma_result ma_context_next_command__wasapi(ma_context* pContext, ma_context_command__wasapi* pCmd) { ma_result result = MA_SUCCESS; MA_ASSERT(pContext != NULL); MA_ASSERT(pCmd != NULL); result = ma_semaphore_wait(&pContext->wasapi.commandSem); if (result == MA_SUCCESS) { ma_mutex_lock(&pContext->wasapi.commandLock); { *pCmd = pContext->wasapi.commands[pContext->wasapi.commandIndex]; pContext->wasapi.commandIndex = (pContext->wasapi.commandIndex + 1) % ma_countof(pContext->wasapi.commands); pContext->wasapi.commandCount -= 1; } ma_mutex_unlock(&pContext->wasapi.commandLock); } return result; } static ma_thread_result MA_THREADCALL ma_context_command_thread__wasapi(void* pUserData) { ma_result result; ma_context* pContext = (ma_context*)pUserData; MA_ASSERT(pContext != NULL); for (;;) { ma_context_command__wasapi cmd; result = ma_context_next_command__wasapi(pContext, &cmd); if (result != MA_SUCCESS) { break; } switch (cmd.code) { case MA_CONTEXT_COMMAND_QUIT__WASAPI: { /* Do nothing. Handled after the switch. */ } break; case MA_CONTEXT_COMMAND_CREATE_IAUDIOCLIENT__WASAPI: { if (cmd.data.createAudioClient.deviceType == ma_device_type_playback) { *cmd.data.createAudioClient.pResult = ma_result_from_HRESULT(ma_IAudioClient_GetService((ma_IAudioClient*)cmd.data.createAudioClient.pAudioClient, &MA_IID_IAudioRenderClient, cmd.data.createAudioClient.ppAudioClientService)); } else { *cmd.data.createAudioClient.pResult = ma_result_from_HRESULT(ma_IAudioClient_GetService((ma_IAudioClient*)cmd.data.createAudioClient.pAudioClient, &MA_IID_IAudioCaptureClient, cmd.data.createAudioClient.ppAudioClientService)); } } break; case MA_CONTEXT_COMMAND_RELEASE_IAUDIOCLIENT__WASAPI: { if (cmd.data.releaseAudioClient.deviceType == ma_device_type_playback) { if (cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback != NULL) { ma_IAudioClient_Release((ma_IAudioClient*)cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback); cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback = NULL; } } if (cmd.data.releaseAudioClient.deviceType == ma_device_type_capture) { if (cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientCapture != NULL) { ma_IAudioClient_Release((ma_IAudioClient*)cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientCapture); cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientCapture = NULL; } } } break; default: { /* Unknown command. Ignore it, but trigger an assert in debug mode so we're aware of it. */ MA_ASSERT(MA_FALSE); } break; } if (cmd.pEvent != NULL) { ma_event_signal(cmd.pEvent); } if (cmd.code == MA_CONTEXT_COMMAND_QUIT__WASAPI) { break; /* Received a quit message. Get out of here. */ } } return (ma_thread_result)0; } static ma_result ma_device_create_IAudioClient_service__wasapi(ma_context* pContext, ma_device_type deviceType, ma_IAudioClient* pAudioClient, void** ppAudioClientService) { ma_result result; ma_result cmdResult; ma_context_command__wasapi cmd = ma_context_init_command__wasapi(MA_CONTEXT_COMMAND_CREATE_IAUDIOCLIENT__WASAPI); cmd.data.createAudioClient.deviceType = deviceType; cmd.data.createAudioClient.pAudioClient = (void*)pAudioClient; cmd.data.createAudioClient.ppAudioClientService = ppAudioClientService; cmd.data.createAudioClient.pResult = &cmdResult; /* Declared locally, but won't be dereferenced after this function returns since execution of the command will wait here. */ result = ma_context_post_command__wasapi(pContext, &cmd); /* This will not return until the command has actually been run. */ if (result != MA_SUCCESS) { return result; } return *cmd.data.createAudioClient.pResult; } #if 0 /* Not used at the moment, but leaving here for future use. */ static ma_result ma_device_release_IAudioClient_service__wasapi(ma_device* pDevice, ma_device_type deviceType) { ma_result result; ma_context_command__wasapi cmd = ma_context_init_command__wasapi(MA_CONTEXT_COMMAND_RELEASE_IAUDIOCLIENT__WASAPI); cmd.data.releaseAudioClient.pDevice = pDevice; cmd.data.releaseAudioClient.deviceType = deviceType; result = ma_context_post_command__wasapi(pDevice->pContext, &cmd); /* This will not return until the command has actually been run. */ if (result != MA_SUCCESS) { return result; } return MA_SUCCESS; } #endif static void ma_add_native_data_format_to_device_info_from_WAVEFORMATEX(const MA_WAVEFORMATEX* pWF, ma_share_mode shareMode, ma_device_info* pInfo) { MA_ASSERT(pWF != NULL); MA_ASSERT(pInfo != NULL); if (pInfo->nativeDataFormatCount >= ma_countof(pInfo->nativeDataFormats)) { return; /* Too many data formats. Need to ignore this one. Don't think this should ever happen with WASAPI. */ } pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].format = ma_format_from_WAVEFORMATEX(pWF); pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].channels = pWF->nChannels; pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].sampleRate = pWF->nSamplesPerSec; pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].flags = (shareMode == ma_share_mode_exclusive) ? MA_DATA_FORMAT_FLAG_EXCLUSIVE_MODE : 0; pInfo->nativeDataFormatCount += 1; } static ma_result ma_context_get_device_info_from_IAudioClient__wasapi(ma_context* pContext, /*ma_IMMDevice**/void* pMMDevice, ma_IAudioClient* pAudioClient, ma_device_info* pInfo) { HRESULT hr; MA_WAVEFORMATEX* pWF = NULL; MA_ASSERT(pAudioClient != NULL); MA_ASSERT(pInfo != NULL); /* Shared Mode. We use GetMixFormat() here. */ hr = ma_IAudioClient_GetMixFormat((ma_IAudioClient*)pAudioClient, (MA_WAVEFORMATEX**)&pWF); if (SUCCEEDED(hr)) { ma_add_native_data_format_to_device_info_from_WAVEFORMATEX(pWF, ma_share_mode_shared, pInfo); } else { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve mix format for device info retrieval."); return ma_result_from_HRESULT(hr); } /* Exclusive Mode. We repeatedly call IsFormatSupported() here. This is not currently supported on UWP. Failure to retrieve the exclusive mode format is not considered an error, so from here on out, MA_SUCCESS is guaranteed to be returned. */ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) { ma_IPropertyStore *pProperties; /* The first thing to do is get the format from PKEY_AudioEngine_DeviceFormat. This should give us a channel count we assume is correct which will simplify our searching. */ hr = ma_IMMDevice_OpenPropertyStore((ma_IMMDevice*)pMMDevice, STGM_READ, &pProperties); if (SUCCEEDED(hr)) { MA_PROPVARIANT var; ma_PropVariantInit(&var); hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_AudioEngine_DeviceFormat, &var); if (SUCCEEDED(hr)) { pWF = (MA_WAVEFORMATEX*)var.blob.pBlobData; /* In my testing, the format returned by PKEY_AudioEngine_DeviceFormat is suitable for exclusive mode so we check this format first. If this fails, fall back to a search. */ hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, pWF, NULL); if (SUCCEEDED(hr)) { /* The format returned by PKEY_AudioEngine_DeviceFormat is supported. */ ma_add_native_data_format_to_device_info_from_WAVEFORMATEX(pWF, ma_share_mode_exclusive, pInfo); } else { /* The format returned by PKEY_AudioEngine_DeviceFormat is not supported, so fall back to a search. We assume the channel count returned by MA_PKEY_AudioEngine_DeviceFormat is valid and correct. For simplicity we're only returning one format. */ ma_uint32 channels = pWF->nChannels; ma_channel defaultChannelMap[MA_MAX_CHANNELS]; MA_WAVEFORMATEXTENSIBLE wf; ma_bool32 found; ma_uint32 iFormat; /* Make sure we don't overflow the channel map. */ if (channels > MA_MAX_CHANNELS) { channels = MA_MAX_CHANNELS; } ma_channel_map_init_standard(ma_standard_channel_map_microsoft, defaultChannelMap, ma_countof(defaultChannelMap), channels); MA_ZERO_OBJECT(&wf); wf.cbSize = sizeof(wf); wf.wFormatTag = WAVE_FORMAT_EXTENSIBLE; wf.nChannels = (WORD)channels; wf.dwChannelMask = ma_channel_map_to_channel_mask__win32(defaultChannelMap, channels); found = MA_FALSE; for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); ++iFormat) { ma_format format = g_maFormatPriorities[iFormat]; ma_uint32 iSampleRate; wf.wBitsPerSample = (WORD)(ma_get_bytes_per_sample(format)*8); wf.nBlockAlign = (WORD)(wf.nChannels * wf.wBitsPerSample / 8); wf.nAvgBytesPerSec = wf.nBlockAlign * wf.nSamplesPerSec; wf.Samples.wValidBitsPerSample = /*(format == ma_format_s24_32) ? 24 :*/ wf.wBitsPerSample; if (format == ma_format_f32) { wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; } else { wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM; } for (iSampleRate = 0; iSampleRate < ma_countof(g_maStandardSampleRatePriorities); ++iSampleRate) { wf.nSamplesPerSec = g_maStandardSampleRatePriorities[iSampleRate]; hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, (MA_WAVEFORMATEX*)&wf, NULL); if (SUCCEEDED(hr)) { ma_add_native_data_format_to_device_info_from_WAVEFORMATEX((MA_WAVEFORMATEX*)&wf, ma_share_mode_exclusive, pInfo); found = MA_TRUE; break; } } if (found) { break; } } ma_PropVariantClear(pContext, &var); if (!found) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "[WASAPI] Failed to find suitable device format for device info retrieval."); } } } else { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "[WASAPI] Failed to retrieve device format for device info retrieval."); } ma_IPropertyStore_Release(pProperties); } else { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "[WASAPI] Failed to open property store for device info retrieval."); } } #else { (void)pMMDevice; /* Unused. */ } #endif return MA_SUCCESS; } #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) static ma_EDataFlow ma_device_type_to_EDataFlow(ma_device_type deviceType) { if (deviceType == ma_device_type_playback) { return ma_eRender; } else if (deviceType == ma_device_type_capture) { return ma_eCapture; } else { MA_ASSERT(MA_FALSE); return ma_eRender; /* Should never hit this. */ } } static ma_result ma_context_create_IMMDeviceEnumerator__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator** ppDeviceEnumerator) { HRESULT hr; ma_IMMDeviceEnumerator* pDeviceEnumerator; MA_ASSERT(pContext != NULL); MA_ASSERT(ppDeviceEnumerator != NULL); *ppDeviceEnumerator = NULL; /* Safety. */ hr = ma_CoCreateInstance(pContext, &MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, &MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator); if (FAILED(hr)) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator."); return ma_result_from_HRESULT(hr); } *ppDeviceEnumerator = pDeviceEnumerator; return MA_SUCCESS; } static WCHAR* ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator* pDeviceEnumerator, ma_device_type deviceType) { HRESULT hr; ma_IMMDevice* pMMDefaultDevice = NULL; WCHAR* pDefaultDeviceID = NULL; ma_EDataFlow dataFlow; ma_ERole role; MA_ASSERT(pContext != NULL); MA_ASSERT(pDeviceEnumerator != NULL); (void)pContext; /* Grab the EDataFlow type from the device type. */ dataFlow = ma_device_type_to_EDataFlow(deviceType); /* The role is always eConsole, but we may make this configurable later. */ role = ma_eConsole; hr = ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(pDeviceEnumerator, dataFlow, role, &pMMDefaultDevice); if (FAILED(hr)) { return NULL; } hr = ma_IMMDevice_GetId(pMMDefaultDevice, &pDefaultDeviceID); ma_IMMDevice_Release(pMMDefaultDevice); pMMDefaultDevice = NULL; if (FAILED(hr)) { return NULL; } return pDefaultDeviceID; } static WCHAR* ma_context_get_default_device_id__wasapi(ma_context* pContext, ma_device_type deviceType) /* Free the returned pointer with ma_CoTaskMemFree() */ { ma_result result; ma_IMMDeviceEnumerator* pDeviceEnumerator; WCHAR* pDefaultDeviceID = NULL; MA_ASSERT(pContext != NULL); result = ma_context_create_IMMDeviceEnumerator__wasapi(pContext, &pDeviceEnumerator); if (result != MA_SUCCESS) { return NULL; } pDefaultDeviceID = ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(pContext, pDeviceEnumerator, deviceType); ma_IMMDeviceEnumerator_Release(pDeviceEnumerator); return pDefaultDeviceID; } static ma_result ma_context_get_MMDevice__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IMMDevice** ppMMDevice) { ma_IMMDeviceEnumerator* pDeviceEnumerator; HRESULT hr; HRESULT CoInitializeResult; MA_ASSERT(pContext != NULL); MA_ASSERT(ppMMDevice != NULL); /* This weird COM init/uninit here is a hack to work around a crash when changing devices. What is happening is WASAPI fires a callback from another thread when the device is changed. It's from that thread where this function is getting called. What I'm suspecting is that the other thread is not initializing COM which in turn results in CoCreateInstance() failing. The community has reported that this seems to fix the crash. There are future plans to move all WASAPI operation over to a single thread to make everything safer, but in the meantime while we wait for that to come online I'm happy enough to use this hack instead. CoUninitialize should only be called if we successfully initialized. S_OK and S_FALSE both mean that we need to call CoUninitialize since the internal ref count was increased. RPC_E_CHANGED_MODE means that CoInitializeEx was called with a different COINIT value, and we don't call CoUninitialize in that case. Other errors are possible, so we check for S_OK and S_FALSE specifically. */ CoInitializeResult = ma_CoInitializeEx(pContext, NULL, MA_COINIT_VALUE); { hr = ma_CoCreateInstance(pContext, &MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, &MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator); } if (CoInitializeResult == S_OK || CoInitializeResult == S_FALSE) { ma_CoUninitialize(pContext); } if (FAILED(hr)) { /* <-- This is checking the call above to ma_CoCreateInstance(). */ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create IMMDeviceEnumerator.\n"); return ma_result_from_HRESULT(hr); } if (pDeviceID == NULL) { hr = ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(pDeviceEnumerator, (deviceType == ma_device_type_capture) ? ma_eCapture : ma_eRender, ma_eConsole, ppMMDevice); } else { hr = ma_IMMDeviceEnumerator_GetDevice(pDeviceEnumerator, pDeviceID->wasapi, ppMMDevice); } ma_IMMDeviceEnumerator_Release(pDeviceEnumerator); if (FAILED(hr)) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve IMMDevice.\n"); return ma_result_from_HRESULT(hr); } return MA_SUCCESS; } static ma_result ma_context_get_device_id_from_MMDevice__wasapi(ma_context* pContext, ma_IMMDevice* pMMDevice, ma_device_id* pDeviceID) { WCHAR* pDeviceIDString; HRESULT hr; MA_ASSERT(pDeviceID != NULL); hr = ma_IMMDevice_GetId(pMMDevice, &pDeviceIDString); if (SUCCEEDED(hr)) { size_t idlen = ma_strlen_WCHAR(pDeviceIDString); if (idlen+1 > ma_countof(pDeviceID->wasapi)) { ma_CoTaskMemFree(pContext, pDeviceIDString); MA_ASSERT(MA_FALSE); /* NOTE: If this is triggered, please report it. It means the format of the ID must have changed and is too long to fit in our fixed sized buffer. */ return MA_ERROR; } MA_COPY_MEMORY(pDeviceID->wasapi, pDeviceIDString, idlen * sizeof(wchar_t)); pDeviceID->wasapi[idlen] = '\0'; ma_CoTaskMemFree(pContext, pDeviceIDString); return MA_SUCCESS; } return MA_ERROR; } static ma_result ma_context_get_device_info_from_MMDevice__wasapi(ma_context* pContext, ma_IMMDevice* pMMDevice, WCHAR* pDefaultDeviceID, ma_bool32 onlySimpleInfo, ma_device_info* pInfo) { ma_result result; HRESULT hr; MA_ASSERT(pContext != NULL); MA_ASSERT(pMMDevice != NULL); MA_ASSERT(pInfo != NULL); /* ID. */ result = ma_context_get_device_id_from_MMDevice__wasapi(pContext, pMMDevice, &pInfo->id); if (result == MA_SUCCESS) { if (pDefaultDeviceID != NULL) { if (ma_strcmp_WCHAR(pInfo->id.wasapi, pDefaultDeviceID) == 0) { pInfo->isDefault = MA_TRUE; } } } /* Description / Friendly Name */ { ma_IPropertyStore *pProperties; hr = ma_IMMDevice_OpenPropertyStore(pMMDevice, STGM_READ, &pProperties); if (SUCCEEDED(hr)) { MA_PROPVARIANT var; ma_PropVariantInit(&var); hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_Device_FriendlyName, &var); if (SUCCEEDED(hr)) { WideCharToMultiByte(CP_UTF8, 0, var.pwszVal, -1, pInfo->name, sizeof(pInfo->name), 0, FALSE); ma_PropVariantClear(pContext, &var); } ma_IPropertyStore_Release(pProperties); } } /* Format */ if (!onlySimpleInfo) { ma_IAudioClient* pAudioClient; hr = ma_IMMDevice_Activate(pMMDevice, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pAudioClient); if (SUCCEEDED(hr)) { result = ma_context_get_device_info_from_IAudioClient__wasapi(pContext, pMMDevice, pAudioClient, pInfo); ma_IAudioClient_Release(pAudioClient); return result; } else { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to activate audio client for device info retrieval."); return ma_result_from_HRESULT(hr); } } return MA_SUCCESS; } static ma_result ma_context_enumerate_devices_by_type__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator* pDeviceEnumerator, ma_device_type deviceType, ma_enum_devices_callback_proc callback, void* pUserData) { ma_result result = MA_SUCCESS; UINT deviceCount; HRESULT hr; ma_uint32 iDevice; WCHAR* pDefaultDeviceID = NULL; ma_IMMDeviceCollection* pDeviceCollection = NULL; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); /* Grab the default device. We use this to know whether or not flag the returned device info as being the default. */ pDefaultDeviceID = ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(pContext, pDeviceEnumerator, deviceType); /* We need to enumerate the devices which returns a device collection. */ hr = ma_IMMDeviceEnumerator_EnumAudioEndpoints(pDeviceEnumerator, ma_device_type_to_EDataFlow(deviceType), MA_MM_DEVICE_STATE_ACTIVE, &pDeviceCollection); if (SUCCEEDED(hr)) { hr = ma_IMMDeviceCollection_GetCount(pDeviceCollection, &deviceCount); if (FAILED(hr)) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to get device count.\n"); result = ma_result_from_HRESULT(hr); goto done; } for (iDevice = 0; iDevice < deviceCount; ++iDevice) { ma_device_info deviceInfo; ma_IMMDevice* pMMDevice; MA_ZERO_OBJECT(&deviceInfo); hr = ma_IMMDeviceCollection_Item(pDeviceCollection, iDevice, &pMMDevice); if (SUCCEEDED(hr)) { result = ma_context_get_device_info_from_MMDevice__wasapi(pContext, pMMDevice, pDefaultDeviceID, MA_TRUE, &deviceInfo); /* MA_TRUE = onlySimpleInfo. */ ma_IMMDevice_Release(pMMDevice); if (result == MA_SUCCESS) { ma_bool32 cbResult = callback(pContext, deviceType, &deviceInfo, pUserData); if (cbResult == MA_FALSE) { break; } } } } } done: if (pDefaultDeviceID != NULL) { ma_CoTaskMemFree(pContext, pDefaultDeviceID); pDefaultDeviceID = NULL; } if (pDeviceCollection != NULL) { ma_IMMDeviceCollection_Release(pDeviceCollection); pDeviceCollection = NULL; } return result; } static ma_result ma_context_get_IAudioClient_Desktop__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, MA_PROPVARIANT* pActivationParams, ma_IAudioClient** ppAudioClient, ma_IMMDevice** ppMMDevice) { ma_result result; HRESULT hr; MA_ASSERT(pContext != NULL); MA_ASSERT(ppAudioClient != NULL); MA_ASSERT(ppMMDevice != NULL); result = ma_context_get_MMDevice__wasapi(pContext, deviceType, pDeviceID, ppMMDevice); if (result != MA_SUCCESS) { return result; } hr = ma_IMMDevice_Activate(*ppMMDevice, &MA_IID_IAudioClient, CLSCTX_ALL, pActivationParams, (void**)ppAudioClient); if (FAILED(hr)) { return ma_result_from_HRESULT(hr); } return MA_SUCCESS; } #else static ma_result ma_context_get_IAudioClient_UWP__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, MA_PROPVARIANT* pActivationParams, ma_IAudioClient** ppAudioClient, ma_IUnknown** ppActivatedInterface) { ma_IActivateAudioInterfaceAsyncOperation *pAsyncOp = NULL; ma_completion_handler_uwp completionHandler; IID iid; WCHAR* iidStr; HRESULT hr; ma_result result; HRESULT activateResult; ma_IUnknown* pActivatedInterface; MA_ASSERT(pContext != NULL); MA_ASSERT(ppAudioClient != NULL); if (pDeviceID != NULL) { iidStr = (WCHAR*)pDeviceID->wasapi; } else { if (deviceType == ma_device_type_capture) { iid = MA_IID_DEVINTERFACE_AUDIO_CAPTURE; } else { iid = MA_IID_DEVINTERFACE_AUDIO_RENDER; } #if defined(__cplusplus) hr = StringFromIID(iid, &iidStr); #else hr = StringFromIID(&iid, &iidStr); #endif if (FAILED(hr)) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to convert device IID to string for ActivateAudioInterfaceAsync(). Out of memory.\n"); return ma_result_from_HRESULT(hr); } } result = ma_completion_handler_uwp_init(&completionHandler); if (result != MA_SUCCESS) { ma_CoTaskMemFree(pContext, iidStr); ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for waiting for ActivateAudioInterfaceAsync().\n"); return result; } hr = ((MA_PFN_ActivateAudioInterfaceAsync)pContext->wasapi.ActivateAudioInterfaceAsync)(iidStr, &MA_IID_IAudioClient, pActivationParams, (ma_IActivateAudioInterfaceCompletionHandler*)&completionHandler, (ma_IActivateAudioInterfaceAsyncOperation**)&pAsyncOp); if (FAILED(hr)) { ma_completion_handler_uwp_uninit(&completionHandler); ma_CoTaskMemFree(pContext, iidStr); ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] ActivateAudioInterfaceAsync() failed.\n"); return ma_result_from_HRESULT(hr); } if (pDeviceID == NULL) { ma_CoTaskMemFree(pContext, iidStr); } /* Wait for the async operation for finish. */ ma_completion_handler_uwp_wait(&completionHandler); ma_completion_handler_uwp_uninit(&completionHandler); hr = ma_IActivateAudioInterfaceAsyncOperation_GetActivateResult(pAsyncOp, &activateResult, &pActivatedInterface); ma_IActivateAudioInterfaceAsyncOperation_Release(pAsyncOp); if (FAILED(hr) || FAILED(activateResult)) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to activate device.\n"); return FAILED(hr) ? ma_result_from_HRESULT(hr) : ma_result_from_HRESULT(activateResult); } /* Here is where we grab the IAudioClient interface. */ hr = ma_IUnknown_QueryInterface(pActivatedInterface, &MA_IID_IAudioClient, (void**)ppAudioClient); if (FAILED(hr)) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to query IAudioClient interface.\n"); return ma_result_from_HRESULT(hr); } if (ppActivatedInterface) { *ppActivatedInterface = pActivatedInterface; } else { ma_IUnknown_Release(pActivatedInterface); } return MA_SUCCESS; } #endif /* https://docs.microsoft.com/en-us/windows/win32/api/audioclientactivationparams/ne-audioclientactivationparams-audioclient_activation_type */ typedef enum { MA_AUDIOCLIENT_ACTIVATION_TYPE_DEFAULT, MA_AUDIOCLIENT_ACTIVATION_TYPE_PROCESS_LOOPBACK } MA_AUDIOCLIENT_ACTIVATION_TYPE; /* https://docs.microsoft.com/en-us/windows/win32/api/audioclientactivationparams/ne-audioclientactivationparams-process_loopback_mode */ typedef enum { MA_PROCESS_LOOPBACK_MODE_INCLUDE_TARGET_PROCESS_TREE, MA_PROCESS_LOOPBACK_MODE_EXCLUDE_TARGET_PROCESS_TREE } MA_PROCESS_LOOPBACK_MODE; /* https://docs.microsoft.com/en-us/windows/win32/api/audioclientactivationparams/ns-audioclientactivationparams-audioclient_process_loopback_params */ typedef struct { DWORD TargetProcessId; MA_PROCESS_LOOPBACK_MODE ProcessLoopbackMode; } MA_AUDIOCLIENT_PROCESS_LOOPBACK_PARAMS; #if defined(_MSC_VER) && !defined(__clang__) #pragma warning(push) #pragma warning(disable:4201) /* nonstandard extension used: nameless struct/union */ #elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))) #pragma GCC diagnostic push #pragma GCC diagnostic ignored "-Wpedantic" /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */ #if defined(__clang__) #pragma GCC diagnostic ignored "-Wc11-extensions" /* anonymous unions are a C11 extension */ #endif #endif /* https://docs.microsoft.com/en-us/windows/win32/api/audioclientactivationparams/ns-audioclientactivationparams-audioclient_activation_params */ typedef struct { MA_AUDIOCLIENT_ACTIVATION_TYPE ActivationType; union { MA_AUDIOCLIENT_PROCESS_LOOPBACK_PARAMS ProcessLoopbackParams; }; } MA_AUDIOCLIENT_ACTIVATION_PARAMS; #if defined(_MSC_VER) && !defined(__clang__) #pragma warning(pop) #elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))) #pragma GCC diagnostic pop #endif #define MA_VIRTUAL_AUDIO_DEVICE_PROCESS_LOOPBACK L"VAD\\Process_Loopback" static ma_result ma_context_get_IAudioClient__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_uint32 loopbackProcessID, ma_bool32 loopbackProcessExclude, ma_IAudioClient** ppAudioClient, ma_WASAPIDeviceInterface** ppDeviceInterface) { ma_result result; ma_bool32 usingProcessLoopback = MA_FALSE; MA_AUDIOCLIENT_ACTIVATION_PARAMS audioclientActivationParams; MA_PROPVARIANT activationParams; MA_PROPVARIANT* pActivationParams = NULL; ma_device_id virtualDeviceID; /* Activation parameters specific to loopback mode. Note that process-specific loopback will only work when a default device ID is specified. */ if (deviceType == ma_device_type_loopback && loopbackProcessID != 0 && pDeviceID == NULL) { usingProcessLoopback = MA_TRUE; } if (usingProcessLoopback) { MA_ZERO_OBJECT(&audioclientActivationParams); audioclientActivationParams.ActivationType = MA_AUDIOCLIENT_ACTIVATION_TYPE_PROCESS_LOOPBACK; audioclientActivationParams.ProcessLoopbackParams.ProcessLoopbackMode = (loopbackProcessExclude) ? MA_PROCESS_LOOPBACK_MODE_EXCLUDE_TARGET_PROCESS_TREE : MA_PROCESS_LOOPBACK_MODE_INCLUDE_TARGET_PROCESS_TREE; audioclientActivationParams.ProcessLoopbackParams.TargetProcessId = (DWORD)loopbackProcessID; ma_PropVariantInit(&activationParams); activationParams.vt = MA_VT_BLOB; activationParams.blob.cbSize = sizeof(audioclientActivationParams); activationParams.blob.pBlobData = (BYTE*)&audioclientActivationParams; pActivationParams = &activationParams; /* When requesting a specific device ID we need to use a special device ID. */ MA_COPY_MEMORY(virtualDeviceID.wasapi, MA_VIRTUAL_AUDIO_DEVICE_PROCESS_LOOPBACK, (ma_wcslen(MA_VIRTUAL_AUDIO_DEVICE_PROCESS_LOOPBACK) + 1) * sizeof(wchar_t)); /* +1 for the null terminator. */ pDeviceID = &virtualDeviceID; } else { pActivationParams = NULL; /* No activation parameters required. */ } #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) result = ma_context_get_IAudioClient_Desktop__wasapi(pContext, deviceType, pDeviceID, pActivationParams, ppAudioClient, ppDeviceInterface); #else result = ma_context_get_IAudioClient_UWP__wasapi(pContext, deviceType, pDeviceID, pActivationParams, ppAudioClient, ppDeviceInterface); #endif /* If loopback mode was requested with a process ID and initialization failed, it could be because it's trying to run on an older version of Windows where it's not supported. We need to let the caller know about this with a log message. */ if (result != MA_SUCCESS) { if (usingProcessLoopback) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Loopback mode requested to %s process ID %u, but initialization failed. Support for this feature begins with Windows 10 Build 20348. Confirm your version of Windows or consider not using process-specific loopback.\n", (loopbackProcessExclude) ? "exclude" : "include", loopbackProcessID); } } return result; } static ma_result ma_context_enumerate_devices__wasapi(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { /* Different enumeration for desktop and UWP. */ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) /* Desktop */ HRESULT hr; ma_IMMDeviceEnumerator* pDeviceEnumerator; hr = ma_CoCreateInstance(pContext, &MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, &MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator); if (FAILED(hr)) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator."); return ma_result_from_HRESULT(hr); } ma_context_enumerate_devices_by_type__wasapi(pContext, pDeviceEnumerator, ma_device_type_playback, callback, pUserData); ma_context_enumerate_devices_by_type__wasapi(pContext, pDeviceEnumerator, ma_device_type_capture, callback, pUserData); ma_IMMDeviceEnumerator_Release(pDeviceEnumerator); #else /* UWP The MMDevice API is only supported on desktop applications. For now, while I'm still figuring out how to properly enumerate over devices without using MMDevice, I'm restricting devices to defaults. Hint: DeviceInformation::FindAllAsync() with DeviceClass.AudioCapture/AudioRender. https://blogs.windows.com/buildingapps/2014/05/15/real-time-audio-in-windows-store-and-windows-phone-apps/ */ if (callback) { ma_bool32 cbResult = MA_TRUE; /* Playback. */ if (cbResult) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); deviceInfo.isDefault = MA_TRUE; cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } /* Capture. */ if (cbResult) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); deviceInfo.isDefault = MA_TRUE; cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); } } #endif return MA_SUCCESS; } static ma_result ma_context_get_device_info__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) ma_result result; ma_IMMDevice* pMMDevice = NULL; WCHAR* pDefaultDeviceID = NULL; result = ma_context_get_MMDevice__wasapi(pContext, deviceType, pDeviceID, &pMMDevice); if (result != MA_SUCCESS) { return result; } /* We need the default device ID so we can set the isDefault flag in the device info. */ pDefaultDeviceID = ma_context_get_default_device_id__wasapi(pContext, deviceType); result = ma_context_get_device_info_from_MMDevice__wasapi(pContext, pMMDevice, pDefaultDeviceID, MA_FALSE, pDeviceInfo); /* MA_FALSE = !onlySimpleInfo. */ if (pDefaultDeviceID != NULL) { ma_CoTaskMemFree(pContext, pDefaultDeviceID); pDefaultDeviceID = NULL; } ma_IMMDevice_Release(pMMDevice); return result; #else ma_IAudioClient* pAudioClient; ma_result result; /* UWP currently only uses default devices. */ if (deviceType == ma_device_type_playback) { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); } else { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); } result = ma_context_get_IAudioClient_UWP__wasapi(pContext, deviceType, pDeviceID, NULL, &pAudioClient, NULL); if (result != MA_SUCCESS) { return result; } result = ma_context_get_device_info_from_IAudioClient__wasapi(pContext, NULL, pAudioClient, pDeviceInfo); pDeviceInfo->isDefault = MA_TRUE; /* UWP only supports default devices. */ ma_IAudioClient_Release(pAudioClient); return result; #endif } static ma_result ma_device_uninit__wasapi(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) { if (pDevice->wasapi.pDeviceEnumerator) { ((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator)->lpVtbl->UnregisterEndpointNotificationCallback((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator, &pDevice->wasapi.notificationClient); ma_IMMDeviceEnumerator_Release((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator); } ma_mutex_uninit(&pDevice->wasapi.rerouteLock); } #endif if (pDevice->wasapi.pRenderClient) { if (pDevice->wasapi.pMappedBufferPlayback != NULL) { ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, pDevice->wasapi.mappedBufferPlaybackCap, 0); pDevice->wasapi.pMappedBufferPlayback = NULL; pDevice->wasapi.mappedBufferPlaybackCap = 0; pDevice->wasapi.mappedBufferPlaybackLen = 0; } ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient); } if (pDevice->wasapi.pCaptureClient) { if (pDevice->wasapi.pMappedBufferCapture != NULL) { ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap); pDevice->wasapi.pMappedBufferCapture = NULL; pDevice->wasapi.mappedBufferCaptureCap = 0; pDevice->wasapi.mappedBufferCaptureLen = 0; } ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); } if (pDevice->wasapi.pAudioClientPlayback) { ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); } if (pDevice->wasapi.pAudioClientCapture) { ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); } if (pDevice->wasapi.hEventPlayback) { CloseHandle((HANDLE)pDevice->wasapi.hEventPlayback); } if (pDevice->wasapi.hEventCapture) { CloseHandle((HANDLE)pDevice->wasapi.hEventCapture); } return MA_SUCCESS; } typedef struct { /* Input. */ ma_format formatIn; ma_uint32 channelsIn; ma_uint32 sampleRateIn; ma_channel channelMapIn[MA_MAX_CHANNELS]; ma_uint32 periodSizeInFramesIn; ma_uint32 periodSizeInMillisecondsIn; ma_uint32 periodsIn; ma_share_mode shareMode; ma_performance_profile performanceProfile; ma_bool32 noAutoConvertSRC; ma_bool32 noDefaultQualitySRC; ma_bool32 noHardwareOffloading; ma_uint32 loopbackProcessID; ma_bool32 loopbackProcessExclude; /* Output. */ ma_IAudioClient* pAudioClient; ma_IAudioRenderClient* pRenderClient; ma_IAudioCaptureClient* pCaptureClient; ma_format formatOut; ma_uint32 channelsOut; ma_uint32 sampleRateOut; ma_channel channelMapOut[MA_MAX_CHANNELS]; ma_uint32 periodSizeInFramesOut; ma_uint32 periodsOut; ma_bool32 usingAudioClient3; char deviceName[256]; ma_device_id id; } ma_device_init_internal_data__wasapi; static ma_result ma_device_init_internal__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_init_internal_data__wasapi* pData) { HRESULT hr; ma_result result = MA_SUCCESS; const char* errorMsg = ""; MA_AUDCLNT_SHAREMODE shareMode = MA_AUDCLNT_SHAREMODE_SHARED; DWORD streamFlags = 0; MA_REFERENCE_TIME periodDurationInMicroseconds; ma_bool32 wasInitializedUsingIAudioClient3 = MA_FALSE; MA_WAVEFORMATEXTENSIBLE wf; ma_WASAPIDeviceInterface* pDeviceInterface = NULL; ma_IAudioClient2* pAudioClient2; ma_uint32 nativeSampleRate; ma_bool32 usingProcessLoopback = MA_FALSE; MA_ASSERT(pContext != NULL); MA_ASSERT(pData != NULL); /* This function is only used to initialize one device type: either playback, capture or loopback. Never full-duplex. */ if (deviceType == ma_device_type_duplex) { return MA_INVALID_ARGS; } usingProcessLoopback = deviceType == ma_device_type_loopback && pData->loopbackProcessID != 0 && pDeviceID == NULL; pData->pAudioClient = NULL; pData->pRenderClient = NULL; pData->pCaptureClient = NULL; streamFlags = MA_AUDCLNT_STREAMFLAGS_EVENTCALLBACK; if (!pData->noAutoConvertSRC && pData->sampleRateIn != 0 && pData->shareMode != ma_share_mode_exclusive) { /* <-- Exclusive streams must use the native sample rate. */ streamFlags |= MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM; } if (!pData->noDefaultQualitySRC && pData->sampleRateIn != 0 && (streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) != 0) { streamFlags |= MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY; } if (deviceType == ma_device_type_loopback) { streamFlags |= MA_AUDCLNT_STREAMFLAGS_LOOPBACK; } result = ma_context_get_IAudioClient__wasapi(pContext, deviceType, pDeviceID, pData->loopbackProcessID, pData->loopbackProcessExclude, &pData->pAudioClient, &pDeviceInterface); if (result != MA_SUCCESS) { goto done; } MA_ZERO_OBJECT(&wf); /* Try enabling hardware offloading. */ if (!pData->noHardwareOffloading) { hr = ma_IAudioClient_QueryInterface(pData->pAudioClient, &MA_IID_IAudioClient2, (void**)&pAudioClient2); if (SUCCEEDED(hr)) { BOOL isHardwareOffloadingSupported = 0; hr = ma_IAudioClient2_IsOffloadCapable(pAudioClient2, MA_AudioCategory_Other, &isHardwareOffloadingSupported); if (SUCCEEDED(hr) && isHardwareOffloadingSupported) { ma_AudioClientProperties clientProperties; MA_ZERO_OBJECT(&clientProperties); clientProperties.cbSize = sizeof(clientProperties); clientProperties.bIsOffload = 1; clientProperties.eCategory = MA_AudioCategory_Other; ma_IAudioClient2_SetClientProperties(pAudioClient2, &clientProperties); } pAudioClient2->lpVtbl->Release(pAudioClient2); } } /* Here is where we try to determine the best format to use with the device. If the client if wanting exclusive mode, first try finding the best format for that. If this fails, fall back to shared mode. */ result = MA_FORMAT_NOT_SUPPORTED; if (pData->shareMode == ma_share_mode_exclusive) { #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) /* In exclusive mode on desktop we always use the backend's native format. */ ma_IPropertyStore* pStore = NULL; hr = ma_IMMDevice_OpenPropertyStore(pDeviceInterface, STGM_READ, &pStore); if (SUCCEEDED(hr)) { MA_PROPVARIANT prop; ma_PropVariantInit(&prop); hr = ma_IPropertyStore_GetValue(pStore, &MA_PKEY_AudioEngine_DeviceFormat, &prop); if (SUCCEEDED(hr)) { MA_WAVEFORMATEX* pActualFormat = (MA_WAVEFORMATEX*)prop.blob.pBlobData; hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pData->pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, pActualFormat, NULL); if (SUCCEEDED(hr)) { MA_COPY_MEMORY(&wf, pActualFormat, sizeof(MA_WAVEFORMATEXTENSIBLE)); } ma_PropVariantClear(pContext, &prop); } ma_IPropertyStore_Release(pStore); } #else /* I do not know how to query the device's native format on UWP so for now I'm just disabling support for exclusive mode. The alternative is to enumerate over different formats and check IsFormatSupported() until you find one that works. TODO: Add support for exclusive mode to UWP. */ hr = S_FALSE; #endif if (hr == S_OK) { shareMode = MA_AUDCLNT_SHAREMODE_EXCLUSIVE; result = MA_SUCCESS; } else { result = MA_SHARE_MODE_NOT_SUPPORTED; } } else { /* In shared mode we are always using the format reported by the operating system. */ MA_WAVEFORMATEXTENSIBLE* pNativeFormat = NULL; hr = ma_IAudioClient_GetMixFormat((ma_IAudioClient*)pData->pAudioClient, (MA_WAVEFORMATEX**)&pNativeFormat); if (hr != S_OK) { /* When using process-specific loopback, GetMixFormat() seems to always fail. */ if (usingProcessLoopback) { wf.wFormatTag = WAVE_FORMAT_IEEE_FLOAT; wf.nChannels = 2; wf.nSamplesPerSec = 44100; wf.wBitsPerSample = 32; wf.nBlockAlign = wf.nChannels * wf.wBitsPerSample / 8; wf.nAvgBytesPerSec = wf.nSamplesPerSec * wf.nBlockAlign; wf.cbSize = sizeof(MA_WAVEFORMATEX); result = MA_SUCCESS; } else { result = MA_FORMAT_NOT_SUPPORTED; } } else { /* I've seen cases where cbSize will be set to sizeof(WAVEFORMATEX) even though the structure itself is given the format tag of WAVE_FORMAT_EXTENSIBLE. If the format tag is WAVE_FORMAT_EXTENSIBLE want to make sure we copy the whole WAVEFORMATEXTENSIBLE structure. Otherwise we'll have to be safe and only copy the WAVEFORMATEX part. */ if (pNativeFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE) { MA_COPY_MEMORY(&wf, pNativeFormat, sizeof(MA_WAVEFORMATEXTENSIBLE)); } else { /* I've seen a case where cbSize was set to 0. Assume sizeof(WAVEFORMATEX) in this case. */ size_t cbSize = pNativeFormat->cbSize; if (cbSize == 0) { cbSize = sizeof(MA_WAVEFORMATEX); } /* Make sure we don't copy more than the capacity of `wf`. */ if (cbSize > sizeof(wf)) { cbSize = sizeof(wf); } MA_COPY_MEMORY(&wf, pNativeFormat, cbSize); } result = MA_SUCCESS; } ma_CoTaskMemFree(pContext, pNativeFormat); shareMode = MA_AUDCLNT_SHAREMODE_SHARED; } /* Return an error if we still haven't found a format. */ if (result != MA_SUCCESS) { errorMsg = "[WASAPI] Failed to find best device mix format."; goto done; } /* Override the native sample rate with the one requested by the caller, but only if we're not using the default sample rate. We'll use WASAPI to perform the sample rate conversion. */ nativeSampleRate = wf.nSamplesPerSec; if (streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) { wf.nSamplesPerSec = (pData->sampleRateIn != 0) ? pData->sampleRateIn : MA_DEFAULT_SAMPLE_RATE; wf.nAvgBytesPerSec = wf.nSamplesPerSec * wf.nBlockAlign; } pData->formatOut = ma_format_from_WAVEFORMATEX((MA_WAVEFORMATEX*)&wf); if (pData->formatOut == ma_format_unknown) { /* The format isn't supported. This is almost certainly because the exclusive mode format isn't supported by miniaudio. We need to return MA_SHARE_MODE_NOT_SUPPORTED in this case so that the caller can detect it and fall back to shared mode if desired. We should never get here if shared mode was requested, but just for completeness we'll check for it and return MA_FORMAT_NOT_SUPPORTED. */ if (shareMode == MA_AUDCLNT_SHAREMODE_EXCLUSIVE) { result = MA_SHARE_MODE_NOT_SUPPORTED; } else { result = MA_FORMAT_NOT_SUPPORTED; } errorMsg = "[WASAPI] Native format not supported."; goto done; } pData->channelsOut = wf.nChannels; pData->sampleRateOut = wf.nSamplesPerSec; /* Get the internal channel map based on the channel mask. There is a possibility that GetMixFormat() returns a WAVEFORMATEX instead of a WAVEFORMATEXTENSIBLE, in which case the channel mask will be undefined. In this case we'll just use the default channel map. */ if (wf.wFormatTag == WAVE_FORMAT_EXTENSIBLE || wf.cbSize >= sizeof(MA_WAVEFORMATEXTENSIBLE)) { ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pData->channelsOut, pData->channelMapOut); } else { ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pData->channelMapOut, ma_countof(pData->channelMapOut), pData->channelsOut); } /* Period size. */ pData->periodsOut = (pData->periodsIn != 0) ? pData->periodsIn : MA_DEFAULT_PERIODS; pData->periodSizeInFramesOut = pData->periodSizeInFramesIn; if (pData->periodSizeInFramesOut == 0) { if (pData->periodSizeInMillisecondsIn == 0) { if (pData->performanceProfile == ma_performance_profile_low_latency) { pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, wf.nSamplesPerSec); } else { pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, wf.nSamplesPerSec); } } else { pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->periodSizeInMillisecondsIn, wf.nSamplesPerSec); } } periodDurationInMicroseconds = ((ma_uint64)pData->periodSizeInFramesOut * 1000 * 1000) / wf.nSamplesPerSec; /* Slightly different initialization for shared and exclusive modes. We try exclusive mode first, and if it fails, fall back to shared mode. */ if (shareMode == MA_AUDCLNT_SHAREMODE_EXCLUSIVE) { MA_REFERENCE_TIME bufferDuration = periodDurationInMicroseconds * pData->periodsOut * 10; /* If the periodicity is too small, Initialize() will fail with AUDCLNT_E_INVALID_DEVICE_PERIOD. In this case we should just keep increasing it and trying it again. */ hr = E_FAIL; for (;;) { hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (MA_WAVEFORMATEX*)&wf, NULL); if (hr == MA_AUDCLNT_E_INVALID_DEVICE_PERIOD) { if (bufferDuration > 500*10000) { break; } else { if (bufferDuration == 0) { /* <-- Just a sanity check to prevent an infinite loop. Should never happen, but it makes me feel better. */ break; } bufferDuration = bufferDuration * 2; continue; } } else { break; } } if (hr == MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) { ma_uint32 bufferSizeInFrames; hr = ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pData->pAudioClient, &bufferSizeInFrames); if (SUCCEEDED(hr)) { bufferDuration = (MA_REFERENCE_TIME)((10000.0 * 1000 / wf.nSamplesPerSec * bufferSizeInFrames) + 0.5); /* Unfortunately we need to release and re-acquire the audio client according to MSDN. Seems silly - why not just call IAudioClient_Initialize() again?! */ ma_IAudioClient_Release((ma_IAudioClient*)pData->pAudioClient); #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) hr = ma_IMMDevice_Activate(pDeviceInterface, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pData->pAudioClient); #else hr = ma_IUnknown_QueryInterface(pDeviceInterface, &MA_IID_IAudioClient, (void**)&pData->pAudioClient); #endif if (SUCCEEDED(hr)) { hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (MA_WAVEFORMATEX*)&wf, NULL); } } } if (FAILED(hr)) { /* Failed to initialize in exclusive mode. Don't fall back to shared mode - instead tell the client about it. They can reinitialize in shared mode if they want. */ if (hr == E_ACCESSDENIED) { errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Access denied.", result = MA_ACCESS_DENIED; } else if (hr == MA_AUDCLNT_E_DEVICE_IN_USE) { errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Device in use.", result = MA_BUSY; } else { errorMsg = "[WASAPI] Failed to initialize device in exclusive mode."; result = ma_result_from_HRESULT(hr); } goto done; } } if (shareMode == MA_AUDCLNT_SHAREMODE_SHARED) { /* Low latency shared mode via IAudioClient3. NOTE ==== Contrary to the documentation on MSDN (https://docs.microsoft.com/en-us/windows/win32/api/audioclient/nf-audioclient-iaudioclient3-initializesharedaudiostream), the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM and AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY with IAudioClient3_InitializeSharedAudioStream() absolutely does not work. Using any of these flags will result in HRESULT code 0x88890021. The other problem is that calling IAudioClient3_GetSharedModeEnginePeriod() with a sample rate different to that returned by IAudioClient_GetMixFormat() also results in an error. I'm therefore disabling low-latency shared mode with AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */ #ifndef MA_WASAPI_NO_LOW_LATENCY_SHARED_MODE { if ((streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) == 0 || nativeSampleRate == wf.nSamplesPerSec) { ma_IAudioClient3* pAudioClient3 = NULL; hr = ma_IAudioClient_QueryInterface(pData->pAudioClient, &MA_IID_IAudioClient3, (void**)&pAudioClient3); if (SUCCEEDED(hr)) { ma_uint32 defaultPeriodInFrames; ma_uint32 fundamentalPeriodInFrames; ma_uint32 minPeriodInFrames; ma_uint32 maxPeriodInFrames; hr = ma_IAudioClient3_GetSharedModeEnginePeriod(pAudioClient3, (MA_WAVEFORMATEX*)&wf, &defaultPeriodInFrames, &fundamentalPeriodInFrames, &minPeriodInFrames, &maxPeriodInFrames); if (SUCCEEDED(hr)) { ma_uint32 desiredPeriodInFrames = pData->periodSizeInFramesOut; ma_uint32 actualPeriodInFrames = desiredPeriodInFrames; /* Make sure the period size is a multiple of fundamentalPeriodInFrames. */ actualPeriodInFrames = actualPeriodInFrames / fundamentalPeriodInFrames; actualPeriodInFrames = actualPeriodInFrames * fundamentalPeriodInFrames; /* The period needs to be clamped between minPeriodInFrames and maxPeriodInFrames. */ actualPeriodInFrames = ma_clamp(actualPeriodInFrames, minPeriodInFrames, maxPeriodInFrames); ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Trying IAudioClient3_InitializeSharedAudioStream(actualPeriodInFrames=%d)\n", actualPeriodInFrames); ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " defaultPeriodInFrames=%d\n", defaultPeriodInFrames); ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " fundamentalPeriodInFrames=%d\n", fundamentalPeriodInFrames); ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " minPeriodInFrames=%d\n", minPeriodInFrames); ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " maxPeriodInFrames=%d\n", maxPeriodInFrames); /* If the client requested a largish buffer than we don't actually want to use low latency shared mode because it forces small buffers. */ if (actualPeriodInFrames >= desiredPeriodInFrames) { /* MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY must not be in the stream flags. If either of these are specified, IAudioClient3_InitializeSharedAudioStream() will fail. */ hr = ma_IAudioClient3_InitializeSharedAudioStream(pAudioClient3, streamFlags & ~(MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY), actualPeriodInFrames, (MA_WAVEFORMATEX*)&wf, NULL); if (SUCCEEDED(hr)) { wasInitializedUsingIAudioClient3 = MA_TRUE; pData->periodSizeInFramesOut = actualPeriodInFrames; ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Using IAudioClient3\n"); ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " periodSizeInFramesOut=%d\n", pData->periodSizeInFramesOut); } else { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] IAudioClient3_InitializeSharedAudioStream failed. Falling back to IAudioClient.\n"); } } else { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Not using IAudioClient3 because the desired period size is larger than the maximum supported by IAudioClient3.\n"); } } else { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] IAudioClient3_GetSharedModeEnginePeriod failed. Falling back to IAudioClient.\n"); } ma_IAudioClient3_Release(pAudioClient3); pAudioClient3 = NULL; } } } #else { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Not using IAudioClient3 because MA_WASAPI_NO_LOW_LATENCY_SHARED_MODE is enabled.\n"); } #endif /* If we don't have an IAudioClient3 then we need to use the normal initialization routine. */ if (!wasInitializedUsingIAudioClient3) { MA_REFERENCE_TIME bufferDuration = periodDurationInMicroseconds * pData->periodsOut * 10; /* <-- Multiply by 10 for microseconds to 100-nanoseconds. */ hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, 0, (const MA_WAVEFORMATEX*)&wf, NULL); if (FAILED(hr)) { if (hr == E_ACCESSDENIED) { errorMsg = "[WASAPI] Failed to initialize device. Access denied.", result = MA_ACCESS_DENIED; } else if (hr == MA_AUDCLNT_E_DEVICE_IN_USE) { errorMsg = "[WASAPI] Failed to initialize device. Device in use.", result = MA_BUSY; } else { errorMsg = "[WASAPI] Failed to initialize device.", result = ma_result_from_HRESULT(hr); } goto done; } } } if (!wasInitializedUsingIAudioClient3) { ma_uint32 bufferSizeInFrames = 0; hr = ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pData->pAudioClient, &bufferSizeInFrames); if (FAILED(hr)) { errorMsg = "[WASAPI] Failed to get audio client's actual buffer size.", result = ma_result_from_HRESULT(hr); goto done; } /* When using process loopback mode, retrieval of the buffer size seems to result in totally incorrect values. In this case we'll just assume it's the same size as what we requested when we initialized the client. */ if (usingProcessLoopback) { bufferSizeInFrames = (ma_uint32)((periodDurationInMicroseconds * pData->periodsOut) * pData->sampleRateOut / 1000000); } pData->periodSizeInFramesOut = bufferSizeInFrames / pData->periodsOut; } pData->usingAudioClient3 = wasInitializedUsingIAudioClient3; if (deviceType == ma_device_type_playback) { result = ma_device_create_IAudioClient_service__wasapi(pContext, deviceType, (ma_IAudioClient*)pData->pAudioClient, (void**)&pData->pRenderClient); } else { result = ma_device_create_IAudioClient_service__wasapi(pContext, deviceType, (ma_IAudioClient*)pData->pAudioClient, (void**)&pData->pCaptureClient); } /*if (FAILED(hr)) {*/ if (result != MA_SUCCESS) { errorMsg = "[WASAPI] Failed to get audio client service."; goto done; } /* Grab the name of the device. */ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) { ma_IPropertyStore *pProperties; hr = ma_IMMDevice_OpenPropertyStore(pDeviceInterface, STGM_READ, &pProperties); if (SUCCEEDED(hr)) { MA_PROPVARIANT varName; ma_PropVariantInit(&varName); hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_Device_FriendlyName, &varName); if (SUCCEEDED(hr)) { WideCharToMultiByte(CP_UTF8, 0, varName.pwszVal, -1, pData->deviceName, sizeof(pData->deviceName), 0, FALSE); ma_PropVariantClear(pContext, &varName); } ma_IPropertyStore_Release(pProperties); } } #endif /* For the WASAPI backend we need to know the actual IDs of the device in order to do automatic stream routing so that IDs can be compared and we can determine which device has been detached and whether or not it matches with our ma_device. */ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) { /* Desktop */ ma_context_get_device_id_from_MMDevice__wasapi(pContext, pDeviceInterface, &pData->id); } #else { /* UWP */ /* TODO: Implement me. Need to figure out how to get the ID of the default device. */ } #endif done: /* Clean up. */ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) if (pDeviceInterface != NULL) { ma_IMMDevice_Release(pDeviceInterface); } #else if (pDeviceInterface != NULL) { ma_IUnknown_Release(pDeviceInterface); } #endif if (result != MA_SUCCESS) { if (pData->pRenderClient) { ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pData->pRenderClient); pData->pRenderClient = NULL; } if (pData->pCaptureClient) { ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pData->pCaptureClient); pData->pCaptureClient = NULL; } if (pData->pAudioClient) { ma_IAudioClient_Release((ma_IAudioClient*)pData->pAudioClient); pData->pAudioClient = NULL; } if (errorMsg != NULL && errorMsg[0] != '\0') { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "%s\n", errorMsg); } return result; } else { return MA_SUCCESS; } } static ma_result ma_device_reinit__wasapi(ma_device* pDevice, ma_device_type deviceType) { ma_device_init_internal_data__wasapi data; ma_result result; MA_ASSERT(pDevice != NULL); /* We only re-initialize the playback or capture device. Never a full-duplex device. */ if (deviceType == ma_device_type_duplex) { return MA_INVALID_ARGS; } /* Before reinitializing the device we need to free the previous audio clients. There's a known memory leak here. We will be calling this from the routing change callback that is fired by WASAPI. If we attempt to release the IAudioClient we will deadlock. In my opinion this is a bug. I'm not sure what I need to do to handle this cleanly, but I think we'll probably need some system where we post an event, but delay the execution of it until the callback has returned. I'm not sure how to do this reliably, however. I have set up some infrastructure for a command thread which might be useful for this. */ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) { if (pDevice->wasapi.pCaptureClient) { ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); pDevice->wasapi.pCaptureClient = NULL; } if (pDevice->wasapi.pAudioClientCapture) { /*ma_device_release_IAudioClient_service__wasapi(pDevice, ma_device_type_capture);*/ pDevice->wasapi.pAudioClientCapture = NULL; } } if (deviceType == ma_device_type_playback) { if (pDevice->wasapi.pRenderClient) { ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient); pDevice->wasapi.pRenderClient = NULL; } if (pDevice->wasapi.pAudioClientPlayback) { /*ma_device_release_IAudioClient_service__wasapi(pDevice, ma_device_type_playback);*/ pDevice->wasapi.pAudioClientPlayback = NULL; } } if (deviceType == ma_device_type_playback) { data.formatIn = pDevice->playback.format; data.channelsIn = pDevice->playback.channels; MA_COPY_MEMORY(data.channelMapIn, pDevice->playback.channelMap, sizeof(pDevice->playback.channelMap)); data.shareMode = pDevice->playback.shareMode; } else { data.formatIn = pDevice->capture.format; data.channelsIn = pDevice->capture.channels; MA_COPY_MEMORY(data.channelMapIn, pDevice->capture.channelMap, sizeof(pDevice->capture.channelMap)); data.shareMode = pDevice->capture.shareMode; } data.sampleRateIn = pDevice->sampleRate; data.periodSizeInFramesIn = pDevice->wasapi.originalPeriodSizeInFrames; data.periodSizeInMillisecondsIn = pDevice->wasapi.originalPeriodSizeInMilliseconds; data.periodsIn = pDevice->wasapi.originalPeriods; data.performanceProfile = pDevice->wasapi.originalPerformanceProfile; data.noAutoConvertSRC = pDevice->wasapi.noAutoConvertSRC; data.noDefaultQualitySRC = pDevice->wasapi.noDefaultQualitySRC; data.noHardwareOffloading = pDevice->wasapi.noHardwareOffloading; data.loopbackProcessID = pDevice->wasapi.loopbackProcessID; data.loopbackProcessExclude = pDevice->wasapi.loopbackProcessExclude; result = ma_device_init_internal__wasapi(pDevice->pContext, deviceType, NULL, &data); if (result != MA_SUCCESS) { return result; } /* At this point we have some new objects ready to go. We need to uninitialize the previous ones and then set the new ones. */ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) { pDevice->wasapi.pAudioClientCapture = data.pAudioClient; pDevice->wasapi.pCaptureClient = data.pCaptureClient; pDevice->capture.internalFormat = data.formatOut; pDevice->capture.internalChannels = data.channelsOut; pDevice->capture.internalSampleRate = data.sampleRateOut; MA_COPY_MEMORY(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); pDevice->capture.internalPeriodSizeInFrames = data.periodSizeInFramesOut; pDevice->capture.internalPeriods = data.periodsOut; ma_strcpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), data.deviceName); ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, (HANDLE)pDevice->wasapi.hEventCapture); pDevice->wasapi.periodSizeInFramesCapture = data.periodSizeInFramesOut; ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &pDevice->wasapi.actualBufferSizeInFramesCapture); /* We must always have a valid ID. */ ma_strcpy_s_WCHAR(pDevice->capture.id.wasapi, sizeof(pDevice->capture.id.wasapi), data.id.wasapi); } if (deviceType == ma_device_type_playback) { pDevice->wasapi.pAudioClientPlayback = data.pAudioClient; pDevice->wasapi.pRenderClient = data.pRenderClient; pDevice->playback.internalFormat = data.formatOut; pDevice->playback.internalChannels = data.channelsOut; pDevice->playback.internalSampleRate = data.sampleRateOut; MA_COPY_MEMORY(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); pDevice->playback.internalPeriodSizeInFrames = data.periodSizeInFramesOut; pDevice->playback.internalPeriods = data.periodsOut; ma_strcpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), data.deviceName); ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, (HANDLE)pDevice->wasapi.hEventPlayback); pDevice->wasapi.periodSizeInFramesPlayback = data.periodSizeInFramesOut; ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &pDevice->wasapi.actualBufferSizeInFramesPlayback); /* We must always have a valid ID because rerouting will look at it. */ ma_strcpy_s_WCHAR(pDevice->playback.id.wasapi, sizeof(pDevice->playback.id.wasapi), data.id.wasapi); } return MA_SUCCESS; } static ma_result ma_device_init__wasapi(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { ma_result result = MA_SUCCESS; #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) HRESULT hr; ma_IMMDeviceEnumerator* pDeviceEnumerator; #endif MA_ASSERT(pDevice != NULL); MA_ZERO_OBJECT(&pDevice->wasapi); pDevice->wasapi.usage = pConfig->wasapi.usage; pDevice->wasapi.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC; pDevice->wasapi.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC; pDevice->wasapi.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading; pDevice->wasapi.loopbackProcessID = pConfig->wasapi.loopbackProcessID; pDevice->wasapi.loopbackProcessExclude = pConfig->wasapi.loopbackProcessExclude; /* Exclusive mode is not allowed with loopback. */ if (pConfig->deviceType == ma_device_type_loopback && pConfig->playback.shareMode == ma_share_mode_exclusive) { return MA_INVALID_DEVICE_CONFIG; } if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) { ma_device_init_internal_data__wasapi data; data.formatIn = pDescriptorCapture->format; data.channelsIn = pDescriptorCapture->channels; data.sampleRateIn = pDescriptorCapture->sampleRate; MA_COPY_MEMORY(data.channelMapIn, pDescriptorCapture->channelMap, sizeof(pDescriptorCapture->channelMap)); data.periodSizeInFramesIn = pDescriptorCapture->periodSizeInFrames; data.periodSizeInMillisecondsIn = pDescriptorCapture->periodSizeInMilliseconds; data.periodsIn = pDescriptorCapture->periodCount; data.shareMode = pDescriptorCapture->shareMode; data.performanceProfile = pConfig->performanceProfile; data.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC; data.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC; data.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading; data.loopbackProcessID = pConfig->wasapi.loopbackProcessID; data.loopbackProcessExclude = pConfig->wasapi.loopbackProcessExclude; result = ma_device_init_internal__wasapi(pDevice->pContext, (pConfig->deviceType == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture, pDescriptorCapture->pDeviceID, &data); if (result != MA_SUCCESS) { return result; } pDevice->wasapi.pAudioClientCapture = data.pAudioClient; pDevice->wasapi.pCaptureClient = data.pCaptureClient; pDevice->wasapi.originalPeriodSizeInMilliseconds = pDescriptorCapture->periodSizeInMilliseconds; pDevice->wasapi.originalPeriodSizeInFrames = pDescriptorCapture->periodSizeInFrames; pDevice->wasapi.originalPeriods = pDescriptorCapture->periodCount; pDevice->wasapi.originalPerformanceProfile = pConfig->performanceProfile; /* The event for capture needs to be manual reset for the same reason as playback. We keep the initial state set to unsignaled, however, because we want to block until we actually have something for the first call to ma_device_read(). */ pDevice->wasapi.hEventCapture = (ma_handle)CreateEventA(NULL, FALSE, FALSE, NULL); /* Auto reset, unsignaled by default. */ if (pDevice->wasapi.hEventCapture == NULL) { result = ma_result_from_GetLastError(GetLastError()); if (pDevice->wasapi.pCaptureClient != NULL) { ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); pDevice->wasapi.pCaptureClient = NULL; } if (pDevice->wasapi.pAudioClientCapture != NULL) { ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); pDevice->wasapi.pAudioClientCapture = NULL; } ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for capture."); return result; } ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, (HANDLE)pDevice->wasapi.hEventCapture); pDevice->wasapi.periodSizeInFramesCapture = data.periodSizeInFramesOut; ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &pDevice->wasapi.actualBufferSizeInFramesCapture); /* We must always have a valid ID. */ ma_strcpy_s_WCHAR(pDevice->capture.id.wasapi, sizeof(pDevice->capture.id.wasapi), data.id.wasapi); /* The descriptor needs to be updated with actual values. */ pDescriptorCapture->format = data.formatOut; pDescriptorCapture->channels = data.channelsOut; pDescriptorCapture->sampleRate = data.sampleRateOut; MA_COPY_MEMORY(pDescriptorCapture->channelMap, data.channelMapOut, sizeof(data.channelMapOut)); pDescriptorCapture->periodSizeInFrames = data.periodSizeInFramesOut; pDescriptorCapture->periodCount = data.periodsOut; } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { ma_device_init_internal_data__wasapi data; data.formatIn = pDescriptorPlayback->format; data.channelsIn = pDescriptorPlayback->channels; data.sampleRateIn = pDescriptorPlayback->sampleRate; MA_COPY_MEMORY(data.channelMapIn, pDescriptorPlayback->channelMap, sizeof(pDescriptorPlayback->channelMap)); data.periodSizeInFramesIn = pDescriptorPlayback->periodSizeInFrames; data.periodSizeInMillisecondsIn = pDescriptorPlayback->periodSizeInMilliseconds; data.periodsIn = pDescriptorPlayback->periodCount; data.shareMode = pDescriptorPlayback->shareMode; data.performanceProfile = pConfig->performanceProfile; data.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC; data.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC; data.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading; data.loopbackProcessID = pConfig->wasapi.loopbackProcessID; data.loopbackProcessExclude = pConfig->wasapi.loopbackProcessExclude; result = ma_device_init_internal__wasapi(pDevice->pContext, ma_device_type_playback, pDescriptorPlayback->pDeviceID, &data); if (result != MA_SUCCESS) { if (pConfig->deviceType == ma_device_type_duplex) { if (pDevice->wasapi.pCaptureClient != NULL) { ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); pDevice->wasapi.pCaptureClient = NULL; } if (pDevice->wasapi.pAudioClientCapture != NULL) { ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); pDevice->wasapi.pAudioClientCapture = NULL; } CloseHandle((HANDLE)pDevice->wasapi.hEventCapture); pDevice->wasapi.hEventCapture = NULL; } return result; } pDevice->wasapi.pAudioClientPlayback = data.pAudioClient; pDevice->wasapi.pRenderClient = data.pRenderClient; pDevice->wasapi.originalPeriodSizeInMilliseconds = pDescriptorPlayback->periodSizeInMilliseconds; pDevice->wasapi.originalPeriodSizeInFrames = pDescriptorPlayback->periodSizeInFrames; pDevice->wasapi.originalPeriods = pDescriptorPlayback->periodCount; pDevice->wasapi.originalPerformanceProfile = pConfig->performanceProfile; /* The event for playback is needs to be manual reset because we want to explicitly control the fact that it becomes signalled only after the whole available space has been filled, never before. The playback event also needs to be initially set to a signaled state so that the first call to ma_device_write() is able to get passed WaitForMultipleObjects(). */ pDevice->wasapi.hEventPlayback = (ma_handle)CreateEventA(NULL, FALSE, TRUE, NULL); /* Auto reset, signaled by default. */ if (pDevice->wasapi.hEventPlayback == NULL) { result = ma_result_from_GetLastError(GetLastError()); if (pConfig->deviceType == ma_device_type_duplex) { if (pDevice->wasapi.pCaptureClient != NULL) { ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); pDevice->wasapi.pCaptureClient = NULL; } if (pDevice->wasapi.pAudioClientCapture != NULL) { ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); pDevice->wasapi.pAudioClientCapture = NULL; } CloseHandle((HANDLE)pDevice->wasapi.hEventCapture); pDevice->wasapi.hEventCapture = NULL; } if (pDevice->wasapi.pRenderClient != NULL) { ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient); pDevice->wasapi.pRenderClient = NULL; } if (pDevice->wasapi.pAudioClientPlayback != NULL) { ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); pDevice->wasapi.pAudioClientPlayback = NULL; } ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for playback."); return result; } ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, (HANDLE)pDevice->wasapi.hEventPlayback); pDevice->wasapi.periodSizeInFramesPlayback = data.periodSizeInFramesOut; ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &pDevice->wasapi.actualBufferSizeInFramesPlayback); /* We must always have a valid ID because rerouting will look at it. */ ma_strcpy_s_WCHAR(pDevice->playback.id.wasapi, sizeof(pDevice->playback.id.wasapi), data.id.wasapi); /* The descriptor needs to be updated with actual values. */ pDescriptorPlayback->format = data.formatOut; pDescriptorPlayback->channels = data.channelsOut; pDescriptorPlayback->sampleRate = data.sampleRateOut; MA_COPY_MEMORY(pDescriptorPlayback->channelMap, data.channelMapOut, sizeof(data.channelMapOut)); pDescriptorPlayback->periodSizeInFrames = data.periodSizeInFramesOut; pDescriptorPlayback->periodCount = data.periodsOut; } /* We need to register a notification client to detect when the device has been disabled, unplugged or re-routed (when the default device changes). When we are connecting to the default device we want to do automatic stream routing when the device is disabled or unplugged. Otherwise we want to just stop the device outright and let the application handle it. */ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK) if (pConfig->wasapi.noAutoStreamRouting == MA_FALSE) { if ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) && pConfig->capture.pDeviceID == NULL) { pDevice->wasapi.allowCaptureAutoStreamRouting = MA_TRUE; } if ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.pDeviceID == NULL) { pDevice->wasapi.allowPlaybackAutoStreamRouting = MA_TRUE; } } ma_mutex_init(&pDevice->wasapi.rerouteLock); hr = ma_CoCreateInstance(pDevice->pContext, &MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, &MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator); if (FAILED(hr)) { ma_device_uninit__wasapi(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator."); return ma_result_from_HRESULT(hr); } pDevice->wasapi.notificationClient.lpVtbl = (void*)&g_maNotificationCientVtbl; pDevice->wasapi.notificationClient.counter = 1; pDevice->wasapi.notificationClient.pDevice = pDevice; hr = pDeviceEnumerator->lpVtbl->RegisterEndpointNotificationCallback(pDeviceEnumerator, &pDevice->wasapi.notificationClient); if (SUCCEEDED(hr)) { pDevice->wasapi.pDeviceEnumerator = (ma_ptr)pDeviceEnumerator; } else { /* Not the end of the world if we fail to register the notification callback. We just won't support automatic stream routing. */ ma_IMMDeviceEnumerator_Release(pDeviceEnumerator); } #endif ma_atomic_bool32_set(&pDevice->wasapi.isStartedCapture, MA_FALSE); ma_atomic_bool32_set(&pDevice->wasapi.isStartedPlayback, MA_FALSE); return MA_SUCCESS; } static ma_result ma_device__get_available_frames__wasapi(ma_device* pDevice, ma_IAudioClient* pAudioClient, ma_uint32* pFrameCount) { ma_uint32 paddingFramesCount; HRESULT hr; ma_share_mode shareMode; MA_ASSERT(pDevice != NULL); MA_ASSERT(pFrameCount != NULL); *pFrameCount = 0; if ((ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientPlayback && (ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientCapture) { return MA_INVALID_OPERATION; } /* I've had a report that GetCurrentPadding() is returning a frame count of 0 which is preventing higher level function calls from doing anything because it thinks nothing is available. I have taken a look at the documentation and it looks like this is unnecessary in exclusive mode. From Microsoft's documentation: For an exclusive-mode rendering or capture stream that was initialized with the AUDCLNT_STREAMFLAGS_EVENTCALLBACK flag, the client typically has no use for the padding value reported by GetCurrentPadding. Instead, the client accesses an entire buffer during each processing pass. Considering this, I'm going to skip GetCurrentPadding() for exclusive mode and just report the entire buffer. This depends on the caller making sure they wait on the event handler. */ shareMode = ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) ? pDevice->playback.shareMode : pDevice->capture.shareMode; if (shareMode == ma_share_mode_shared) { /* Shared mode. */ hr = ma_IAudioClient_GetCurrentPadding(pAudioClient, &paddingFramesCount); if (FAILED(hr)) { return ma_result_from_HRESULT(hr); } if ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) { *pFrameCount = pDevice->wasapi.actualBufferSizeInFramesPlayback - paddingFramesCount; } else { *pFrameCount = paddingFramesCount; } } else { /* Exclusive mode. */ if ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) { *pFrameCount = pDevice->wasapi.actualBufferSizeInFramesPlayback; } else { *pFrameCount = pDevice->wasapi.actualBufferSizeInFramesCapture; } } return MA_SUCCESS; } static ma_result ma_device_reroute__wasapi(ma_device* pDevice, ma_device_type deviceType) { ma_result result; if (deviceType == ma_device_type_duplex) { return MA_INVALID_ARGS; } ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "=== CHANGING DEVICE ===\n"); result = ma_device_reinit__wasapi(pDevice, deviceType); if (result != MA_SUCCESS) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[WASAPI] Reinitializing device after route change failed.\n"); return result; } ma_device__post_init_setup(pDevice, deviceType); ma_device__on_notification_rerouted(pDevice); ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "=== DEVICE CHANGED ===\n"); return MA_SUCCESS; } static ma_result ma_device_start__wasapi_nolock(ma_device* pDevice) { HRESULT hr; if (pDevice->pContext->wasapi.hAvrt) { const char* pTaskName = ma_to_usage_string__wasapi(pDevice->wasapi.usage); if (pTaskName) { DWORD idx = 0; pDevice->wasapi.hAvrtHandle = (ma_handle)((MA_PFN_AvSetMmThreadCharacteristicsA)pDevice->pContext->wasapi.AvSetMmThreadCharacteristicsA)(pTaskName, &idx); } } if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) { hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); if (FAILED(hr)) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal capture device. HRESULT = %d.", (int)hr); return ma_result_from_HRESULT(hr); } ma_atomic_bool32_set(&pDevice->wasapi.isStartedCapture, MA_TRUE); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); if (FAILED(hr)) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal playback device. HRESULT = %d.", (int)hr); return ma_result_from_HRESULT(hr); } ma_atomic_bool32_set(&pDevice->wasapi.isStartedPlayback, MA_TRUE); } return MA_SUCCESS; } static ma_result ma_device_start__wasapi(ma_device* pDevice) { ma_result result; MA_ASSERT(pDevice != NULL); /* Wait for any rerouting to finish before attempting to start the device. */ ma_mutex_lock(&pDevice->wasapi.rerouteLock); { result = ma_device_start__wasapi_nolock(pDevice); } ma_mutex_unlock(&pDevice->wasapi.rerouteLock); return result; } static ma_result ma_device_stop__wasapi_nolock(ma_device* pDevice) { ma_result result; HRESULT hr; MA_ASSERT(pDevice != NULL); if (pDevice->wasapi.hAvrtHandle) { ((MA_PFN_AvRevertMmThreadCharacteristics)pDevice->pContext->wasapi.AvRevertMmThreadcharacteristics)((HANDLE)pDevice->wasapi.hAvrtHandle); pDevice->wasapi.hAvrtHandle = NULL; } if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) { /* If we have a mapped buffer we need to release it. */ if (pDevice->wasapi.pMappedBufferCapture != NULL) { ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap); pDevice->wasapi.pMappedBufferCapture = NULL; pDevice->wasapi.mappedBufferCaptureCap = 0; pDevice->wasapi.mappedBufferCaptureLen = 0; } hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal capture device."); return ma_result_from_HRESULT(hr); } /* The audio client needs to be reset otherwise restarting will fail. */ hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal capture device."); return ma_result_from_HRESULT(hr); } ma_atomic_bool32_set(&pDevice->wasapi.isStartedCapture, MA_FALSE); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { if (pDevice->wasapi.pMappedBufferPlayback != NULL) { ma_silence_pcm_frames( ma_offset_pcm_frames_ptr(pDevice->wasapi.pMappedBufferPlayback, pDevice->wasapi.mappedBufferPlaybackLen, pDevice->playback.internalFormat, pDevice->playback.internalChannels), pDevice->wasapi.mappedBufferPlaybackCap - pDevice->wasapi.mappedBufferPlaybackLen, pDevice->playback.internalFormat, pDevice->playback.internalChannels ); ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, pDevice->wasapi.mappedBufferPlaybackCap, 0); pDevice->wasapi.pMappedBufferPlayback = NULL; pDevice->wasapi.mappedBufferPlaybackCap = 0; pDevice->wasapi.mappedBufferPlaybackLen = 0; } /* The buffer needs to be drained before stopping the device. Not doing this will result in the last few frames not getting output to the speakers. This is a problem for very short sounds because it'll result in a significant portion of it not getting played. */ if (ma_atomic_bool32_get(&pDevice->wasapi.isStartedPlayback)) { /* We need to make sure we put a timeout here or else we'll risk getting stuck in a deadlock in some cases. */ DWORD waitTime = (pDevice->wasapi.actualBufferSizeInFramesPlayback * 1000) / pDevice->playback.internalSampleRate; if (pDevice->playback.shareMode == ma_share_mode_exclusive) { WaitForSingleObject((HANDLE)pDevice->wasapi.hEventPlayback, waitTime); } else { ma_uint32 prevFramesAvailablePlayback = (ma_uint32)-1; ma_uint32 framesAvailablePlayback; for (;;) { result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback); if (result != MA_SUCCESS) { break; } if (framesAvailablePlayback >= pDevice->wasapi.actualBufferSizeInFramesPlayback) { break; } /* Just a safety check to avoid an infinite loop. If this iteration results in a situation where the number of available frames has not changed, get out of the loop. I don't think this should ever happen, but I think it's nice to have just in case. */ if (framesAvailablePlayback == prevFramesAvailablePlayback) { break; } prevFramesAvailablePlayback = framesAvailablePlayback; ResetEvent((HANDLE)pDevice->wasapi.hEventPlayback); /* Manual reset. */ WaitForSingleObject((HANDLE)pDevice->wasapi.hEventPlayback, waitTime); } } } hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal playback device."); return ma_result_from_HRESULT(hr); } /* The audio client needs to be reset otherwise restarting will fail. */ { ma_int32 retries = 5; while ((hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback)) == MA_AUDCLNT_E_BUFFER_OPERATION_PENDING && retries > 0) { ma_sleep(10); retries -= 1; } } if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal playback device."); return ma_result_from_HRESULT(hr); } ma_atomic_bool32_set(&pDevice->wasapi.isStartedPlayback, MA_FALSE); } return MA_SUCCESS; } static ma_result ma_device_stop__wasapi(ma_device* pDevice) { ma_result result; MA_ASSERT(pDevice != NULL); /* Wait for any rerouting to finish before attempting to stop the device. */ ma_mutex_lock(&pDevice->wasapi.rerouteLock); { result = ma_device_stop__wasapi_nolock(pDevice); } ma_mutex_unlock(&pDevice->wasapi.rerouteLock); return result; } #ifndef MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS #define MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS 5000 #endif static ma_result ma_device_read__wasapi(ma_device* pDevice, void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) { ma_result result = MA_SUCCESS; ma_uint32 totalFramesProcessed = 0; /* When reading, we need to get a buffer and process all of it before releasing it. Because the frame count (frameCount) can be different to the size of the buffer, we'll need to cache the pointer to the buffer. */ /* Keep running until we've processed the requested number of frames. */ while (ma_device_get_state(pDevice) == ma_device_state_started && totalFramesProcessed < frameCount) { ma_uint32 framesRemaining = frameCount - totalFramesProcessed; /* If we have a mapped data buffer, consume that first. */ if (pDevice->wasapi.pMappedBufferCapture != NULL) { /* We have a cached data pointer so consume that before grabbing another one from WASAPI. */ ma_uint32 framesToProcessNow = framesRemaining; if (framesToProcessNow > pDevice->wasapi.mappedBufferCaptureLen) { framesToProcessNow = pDevice->wasapi.mappedBufferCaptureLen; } /* Now just copy the data over to the output buffer. */ ma_copy_pcm_frames( ma_offset_pcm_frames_ptr(pFrames, totalFramesProcessed, pDevice->capture.internalFormat, pDevice->capture.internalChannels), ma_offset_pcm_frames_const_ptr(pDevice->wasapi.pMappedBufferCapture, pDevice->wasapi.mappedBufferCaptureCap - pDevice->wasapi.mappedBufferCaptureLen, pDevice->capture.internalFormat, pDevice->capture.internalChannels), framesToProcessNow, pDevice->capture.internalFormat, pDevice->capture.internalChannels ); totalFramesProcessed += framesToProcessNow; pDevice->wasapi.mappedBufferCaptureLen -= framesToProcessNow; /* If the data buffer has been fully consumed we need to release it. */ if (pDevice->wasapi.mappedBufferCaptureLen == 0) { ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap); pDevice->wasapi.pMappedBufferCapture = NULL; pDevice->wasapi.mappedBufferCaptureCap = 0; } } else { /* We don't have any cached data pointer, so grab another one. */ HRESULT hr; DWORD flags = 0; /* First just ask WASAPI for a data buffer. If it's not available, we'll wait for more. */ hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pDevice->wasapi.pMappedBufferCapture, &pDevice->wasapi.mappedBufferCaptureCap, &flags, NULL, NULL); if (hr == S_OK) { /* We got a data buffer. Continue to the next loop iteration which will then read from the mapped pointer. */ pDevice->wasapi.mappedBufferCaptureLen = pDevice->wasapi.mappedBufferCaptureCap; /* There have been reports that indicate that at times the AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY is reported for every call to IAudioCaptureClient_GetBuffer() above which results in spamming of the debug messages below. To partially work around this, I'm only outputting these messages when MA_DEBUG_OUTPUT is explicitly defined. The better solution would be to figure out why the flag is always getting reported. */ #if defined(MA_DEBUG_OUTPUT) { if (flags != 0) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Capture Flags: %ld\n", flags); if ((flags & MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) != 0) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Data discontinuity (possible overrun). Attempting recovery. mappedBufferCaptureCap=%d\n", pDevice->wasapi.mappedBufferCaptureCap); } } } #endif /* Overrun detection. */ if ((flags & MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) != 0) { /* Glitched. Probably due to an overrun. */ /* If we got an overrun it probably means we're straddling the end of the buffer. In normal capture mode this is the fault of the client application because they're responsible for ensuring data is processed fast enough. In duplex mode, however, the processing of audio is tied to the playback device, so this can possibly be the result of a timing de-sync. In capture mode we're not going to do any kind of recovery because the real fix is for the client application to process faster. In duplex mode, we'll treat this as a desync and reset the buffers to prevent a never-ending sequence of glitches due to straddling the end of the buffer. */ if (pDevice->type == ma_device_type_duplex) { /* Experiment: If we empty out the *entire* buffer we may end up putting ourselves into an underrun position which isn't really any better than the overrun we're probably in right now. Instead we'll just empty out about half. */ ma_uint32 i; ma_uint32 periodCount = (pDevice->wasapi.actualBufferSizeInFramesCapture / pDevice->wasapi.periodSizeInFramesCapture); ma_uint32 iterationCount = periodCount / 2; if ((periodCount % 2) > 0) { iterationCount += 1; } for (i = 0; i < iterationCount; i += 1) { hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap); if (FAILED(hr)) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Data discontinuity recovery: IAudioCaptureClient_ReleaseBuffer() failed with %ld.\n", hr); break; } flags = 0; hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pDevice->wasapi.pMappedBufferCapture, &pDevice->wasapi.mappedBufferCaptureCap, &flags, NULL, NULL); if (hr == MA_AUDCLNT_S_BUFFER_EMPTY || FAILED(hr)) { /* The buffer has been completely emptied or an error occurred. In this case we'll need to reset the state of the mapped buffer which will trigger the next iteration to get a fresh buffer from WASAPI. */ pDevice->wasapi.pMappedBufferCapture = NULL; pDevice->wasapi.mappedBufferCaptureCap = 0; pDevice->wasapi.mappedBufferCaptureLen = 0; if (hr == MA_AUDCLNT_S_BUFFER_EMPTY) { if ((flags & MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) != 0) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Data discontinuity recovery: Buffer emptied, and data discontinuity still reported.\n"); } else { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Data discontinuity recovery: Buffer emptied.\n"); } } if (FAILED(hr)) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Data discontinuity recovery: IAudioCaptureClient_GetBuffer() failed with %ld.\n", hr); } break; } } /* If at this point we have a valid buffer mapped, make sure the buffer length is set appropriately. */ if (pDevice->wasapi.pMappedBufferCapture != NULL) { pDevice->wasapi.mappedBufferCaptureLen = pDevice->wasapi.mappedBufferCaptureCap; } } } continue; } else { if (hr == MA_AUDCLNT_S_BUFFER_EMPTY || hr == MA_AUDCLNT_E_BUFFER_ERROR) { /* No data is available. We need to wait for more. There's two situations to consider here. The first is normal capture mode. If this times out it probably means the microphone isn't delivering data for whatever reason. In this case we'll just abort the read and return whatever we were able to get. The other situations is loopback mode, in which case a timeout probably just means the nothing is playing through the speakers. */ /* Experiment: Use a shorter timeout for loopback mode. */ DWORD timeoutInMilliseconds = MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS; if (pDevice->type == ma_device_type_loopback) { timeoutInMilliseconds = 10; } if (WaitForSingleObject((HANDLE)pDevice->wasapi.hEventCapture, timeoutInMilliseconds) != WAIT_OBJECT_0) { if (pDevice->type == ma_device_type_loopback) { continue; /* Keep waiting in loopback mode. */ } else { result = MA_ERROR; break; /* Wait failed. */ } } /* At this point we should be able to loop back to the start of the loop and try retrieving a data buffer again. */ } else { /* An error occurred and we need to abort. */ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for reading from the device. HRESULT = %d. Stopping device.\n", (int)hr); result = ma_result_from_HRESULT(hr); break; } } } } /* If we were unable to process the entire requested frame count, but we still have a mapped buffer, there's a good chance either an error occurred or the device was stopped mid-read. In this case we'll need to make sure the buffer is released. */ if (totalFramesProcessed < frameCount && pDevice->wasapi.pMappedBufferCapture != NULL) { ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap); pDevice->wasapi.pMappedBufferCapture = NULL; pDevice->wasapi.mappedBufferCaptureCap = 0; pDevice->wasapi.mappedBufferCaptureLen = 0; } if (pFramesRead != NULL) { *pFramesRead = totalFramesProcessed; } return result; } static ma_result ma_device_write__wasapi(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { ma_result result = MA_SUCCESS; ma_uint32 totalFramesProcessed = 0; /* Keep writing to the device until it's stopped or we've consumed all of our input. */ while (ma_device_get_state(pDevice) == ma_device_state_started && totalFramesProcessed < frameCount) { ma_uint32 framesRemaining = frameCount - totalFramesProcessed; /* We're going to do this in a similar way to capture. We'll first check if the cached data pointer is valid, and if so, read from that. Otherwise We will call IAudioRenderClient_GetBuffer() with a requested buffer size equal to our actual period size. If it returns AUDCLNT_E_BUFFER_TOO_LARGE it means we need to wait for some data to become available. */ if (pDevice->wasapi.pMappedBufferPlayback != NULL) { /* We still have some space available in the mapped data buffer. Write to it. */ ma_uint32 framesToProcessNow = framesRemaining; if (framesToProcessNow > (pDevice->wasapi.mappedBufferPlaybackCap - pDevice->wasapi.mappedBufferPlaybackLen)) { framesToProcessNow = (pDevice->wasapi.mappedBufferPlaybackCap - pDevice->wasapi.mappedBufferPlaybackLen); } /* Now just copy the data over to the output buffer. */ ma_copy_pcm_frames( ma_offset_pcm_frames_ptr(pDevice->wasapi.pMappedBufferPlayback, pDevice->wasapi.mappedBufferPlaybackLen, pDevice->playback.internalFormat, pDevice->playback.internalChannels), ma_offset_pcm_frames_const_ptr(pFrames, totalFramesProcessed, pDevice->playback.internalFormat, pDevice->playback.internalChannels), framesToProcessNow, pDevice->playback.internalFormat, pDevice->playback.internalChannels ); totalFramesProcessed += framesToProcessNow; pDevice->wasapi.mappedBufferPlaybackLen += framesToProcessNow; /* If the data buffer has been fully consumed we need to release it. */ if (pDevice->wasapi.mappedBufferPlaybackLen == pDevice->wasapi.mappedBufferPlaybackCap) { ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, pDevice->wasapi.mappedBufferPlaybackCap, 0); pDevice->wasapi.pMappedBufferPlayback = NULL; pDevice->wasapi.mappedBufferPlaybackCap = 0; pDevice->wasapi.mappedBufferPlaybackLen = 0; /* In exclusive mode we need to wait here. Exclusive mode is weird because GetBuffer() never seems to return AUDCLNT_E_BUFFER_TOO_LARGE, which is what we normally use to determine whether or not we need to wait for more data. */ if (pDevice->playback.shareMode == ma_share_mode_exclusive) { if (WaitForSingleObject((HANDLE)pDevice->wasapi.hEventPlayback, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) { result = MA_ERROR; break; /* Wait failed. Probably timed out. */ } } } } else { /* We don't have a mapped data buffer so we'll need to get one. */ HRESULT hr; ma_uint32 bufferSizeInFrames; /* Special rules for exclusive mode. */ if (pDevice->playback.shareMode == ma_share_mode_exclusive) { bufferSizeInFrames = pDevice->wasapi.actualBufferSizeInFramesPlayback; } else { bufferSizeInFrames = pDevice->wasapi.periodSizeInFramesPlayback; } hr = ma_IAudioRenderClient_GetBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, bufferSizeInFrames, (BYTE**)&pDevice->wasapi.pMappedBufferPlayback); if (hr == S_OK) { /* We have data available. */ pDevice->wasapi.mappedBufferPlaybackCap = bufferSizeInFrames; pDevice->wasapi.mappedBufferPlaybackLen = 0; } else { if (hr == MA_AUDCLNT_E_BUFFER_TOO_LARGE || hr == MA_AUDCLNT_E_BUFFER_ERROR) { /* Not enough data available. We need to wait for more. */ if (WaitForSingleObject((HANDLE)pDevice->wasapi.hEventPlayback, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) { result = MA_ERROR; break; /* Wait failed. Probably timed out. */ } } else { /* Some error occurred. We'll need to abort. */ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from playback device in preparation for writing to the device. HRESULT = %d. Stopping device.\n", (int)hr); result = ma_result_from_HRESULT(hr); break; } } } } if (pFramesWritten != NULL) { *pFramesWritten = totalFramesProcessed; } return result; } static ma_result ma_device_data_loop_wakeup__wasapi(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) { SetEvent((HANDLE)pDevice->wasapi.hEventCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { SetEvent((HANDLE)pDevice->wasapi.hEventPlayback); } return MA_SUCCESS; } static ma_result ma_context_uninit__wasapi(ma_context* pContext) { ma_context_command__wasapi cmd = ma_context_init_command__wasapi(MA_CONTEXT_COMMAND_QUIT__WASAPI); MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_wasapi); ma_context_post_command__wasapi(pContext, &cmd); ma_thread_wait(&pContext->wasapi.commandThread); if (pContext->wasapi.hAvrt) { ma_dlclose(ma_context_get_log(pContext), pContext->wasapi.hAvrt); pContext->wasapi.hAvrt = NULL; } #if defined(MA_WIN32_UWP) { if (pContext->wasapi.hMMDevapi) { ma_dlclose(ma_context_get_log(pContext), pContext->wasapi.hMMDevapi); pContext->wasapi.hMMDevapi = NULL; } } #endif /* Only after the thread has been terminated can we uninitialize the sync objects for the command thread. */ ma_semaphore_uninit(&pContext->wasapi.commandSem); ma_mutex_uninit(&pContext->wasapi.commandLock); return MA_SUCCESS; } static ma_result ma_context_init__wasapi(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { ma_result result = MA_SUCCESS; MA_ASSERT(pContext != NULL); (void)pConfig; #ifdef MA_WIN32_DESKTOP /* WASAPI is only supported in Vista SP1 and newer. The reason for SP1 and not the base version of Vista is that event-driven exclusive mode does not work until SP1. Unfortunately older compilers don't define these functions so we need to dynamically load them in order to avoid a link error. */ { ma_OSVERSIONINFOEXW osvi; ma_handle kernel32DLL; ma_PFNVerifyVersionInfoW _VerifyVersionInfoW; ma_PFNVerSetConditionMask _VerSetConditionMask; kernel32DLL = ma_dlopen(ma_context_get_log(pContext), "kernel32.dll"); if (kernel32DLL == NULL) { return MA_NO_BACKEND; } _VerifyVersionInfoW = (ma_PFNVerifyVersionInfoW )ma_dlsym(ma_context_get_log(pContext), kernel32DLL, "VerifyVersionInfoW"); _VerSetConditionMask = (ma_PFNVerSetConditionMask)ma_dlsym(ma_context_get_log(pContext), kernel32DLL, "VerSetConditionMask"); if (_VerifyVersionInfoW == NULL || _VerSetConditionMask == NULL) { ma_dlclose(ma_context_get_log(pContext), kernel32DLL); return MA_NO_BACKEND; } MA_ZERO_OBJECT(&osvi); osvi.dwOSVersionInfoSize = sizeof(osvi); osvi.dwMajorVersion = ((MA_WIN32_WINNT_VISTA >> 8) & 0xFF); osvi.dwMinorVersion = ((MA_WIN32_WINNT_VISTA >> 0) & 0xFF); osvi.wServicePackMajor = 1; if (_VerifyVersionInfoW(&osvi, MA_VER_MAJORVERSION | MA_VER_MINORVERSION | MA_VER_SERVICEPACKMAJOR, _VerSetConditionMask(_VerSetConditionMask(_VerSetConditionMask(0, MA_VER_MAJORVERSION, MA_VER_GREATER_EQUAL), MA_VER_MINORVERSION, MA_VER_GREATER_EQUAL), MA_VER_SERVICEPACKMAJOR, MA_VER_GREATER_EQUAL))) { result = MA_SUCCESS; } else { result = MA_NO_BACKEND; } ma_dlclose(ma_context_get_log(pContext), kernel32DLL); } #endif if (result != MA_SUCCESS) { return result; } MA_ZERO_OBJECT(&pContext->wasapi); #if defined(MA_WIN32_UWP) { /* Link to mmdevapi so we can get access to ActivateAudioInterfaceAsync(). */ pContext->wasapi.hMMDevapi = ma_dlopen(ma_context_get_log(pContext), "mmdevapi.dll"); if (pContext->wasapi.hMMDevapi) { pContext->wasapi.ActivateAudioInterfaceAsync = ma_dlsym(ma_context_get_log(pContext), pContext->wasapi.hMMDevapi, "ActivateAudioInterfaceAsync"); if (pContext->wasapi.ActivateAudioInterfaceAsync == NULL) { ma_dlclose(ma_context_get_log(pContext), pContext->wasapi.hMMDevapi); return MA_NO_BACKEND; /* ActivateAudioInterfaceAsync() could not be loaded. */ } } else { return MA_NO_BACKEND; /* Failed to load mmdevapi.dll which is required for ActivateAudioInterfaceAsync() */ } } #endif /* Optionally use the Avrt API to specify the audio thread's latency sensitivity requirements */ pContext->wasapi.hAvrt = ma_dlopen(ma_context_get_log(pContext), "avrt.dll"); if (pContext->wasapi.hAvrt) { pContext->wasapi.AvSetMmThreadCharacteristicsA = ma_dlsym(ma_context_get_log(pContext), pContext->wasapi.hAvrt, "AvSetMmThreadCharacteristicsA"); pContext->wasapi.AvRevertMmThreadcharacteristics = ma_dlsym(ma_context_get_log(pContext), pContext->wasapi.hAvrt, "AvRevertMmThreadCharacteristics"); /* If either function could not be found, disable use of avrt entirely. */ if (!pContext->wasapi.AvSetMmThreadCharacteristicsA || !pContext->wasapi.AvRevertMmThreadcharacteristics) { pContext->wasapi.AvSetMmThreadCharacteristicsA = NULL; pContext->wasapi.AvRevertMmThreadcharacteristics = NULL; ma_dlclose(ma_context_get_log(pContext), pContext->wasapi.hAvrt); pContext->wasapi.hAvrt = NULL; } } /* Annoyingly, WASAPI does not allow you to release an IAudioClient object from a different thread than the one that retrieved it with GetService(). This can result in a deadlock in two situations: 1) When calling ma_device_uninit() from a different thread to ma_device_init(); and 2) When uninitializing and reinitializing the internal IAudioClient object in response to automatic stream routing. We could define ma_device_uninit() such that it must be called on the same thread as ma_device_init(). We could also just not release the IAudioClient when performing automatic stream routing to avoid the deadlock. Neither of these are acceptable solutions in my view so we're going to have to work around this with a worker thread. This is not ideal, but I can't think of a better way to do this. More information about this can be found here: https://docs.microsoft.com/en-us/windows/win32/api/audioclient/nn-audioclient-iaudiorenderclient Note this section: When releasing an IAudioRenderClient interface instance, the client must call the interface's Release method from the same thread as the call to IAudioClient::GetService that created the object. */ { result = ma_mutex_init(&pContext->wasapi.commandLock); if (result != MA_SUCCESS) { return result; } result = ma_semaphore_init(0, &pContext->wasapi.commandSem); if (result != MA_SUCCESS) { ma_mutex_uninit(&pContext->wasapi.commandLock); return result; } result = ma_thread_create(&pContext->wasapi.commandThread, ma_thread_priority_normal, 0, ma_context_command_thread__wasapi, pContext, &pContext->allocationCallbacks); if (result != MA_SUCCESS) { ma_semaphore_uninit(&pContext->wasapi.commandSem); ma_mutex_uninit(&pContext->wasapi.commandLock); return result; } } pCallbacks->onContextInit = ma_context_init__wasapi; pCallbacks->onContextUninit = ma_context_uninit__wasapi; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__wasapi; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__wasapi; pCallbacks->onDeviceInit = ma_device_init__wasapi; pCallbacks->onDeviceUninit = ma_device_uninit__wasapi; pCallbacks->onDeviceStart = ma_device_start__wasapi; pCallbacks->onDeviceStop = ma_device_stop__wasapi; pCallbacks->onDeviceRead = ma_device_read__wasapi; pCallbacks->onDeviceWrite = ma_device_write__wasapi; pCallbacks->onDeviceDataLoop = NULL; pCallbacks->onDeviceDataLoopWakeup = ma_device_data_loop_wakeup__wasapi; return MA_SUCCESS; } #endif /****************************************************************************** DirectSound Backend ******************************************************************************/ #ifdef MA_HAS_DSOUND /*#include <dsound.h>*/ /*static const GUID MA_GUID_IID_DirectSoundNotify = {0xb0210783, 0x89cd, 0x11d0, {0xaf, 0x08, 0x00, 0xa0, 0xc9, 0x25, 0xcd, 0x16}};*/ /* miniaudio only uses priority or exclusive modes. */ #define MA_DSSCL_NORMAL 1 #define MA_DSSCL_PRIORITY 2 #define MA_DSSCL_EXCLUSIVE 3 #define MA_DSSCL_WRITEPRIMARY 4 #define MA_DSCAPS_PRIMARYMONO 0x00000001 #define MA_DSCAPS_PRIMARYSTEREO 0x00000002 #define MA_DSCAPS_PRIMARY8BIT 0x00000004 #define MA_DSCAPS_PRIMARY16BIT 0x00000008 #define MA_DSCAPS_CONTINUOUSRATE 0x00000010 #define MA_DSCAPS_EMULDRIVER 0x00000020 #define MA_DSCAPS_CERTIFIED 0x00000040 #define MA_DSCAPS_SECONDARYMONO 0x00000100 #define MA_DSCAPS_SECONDARYSTEREO 0x00000200 #define MA_DSCAPS_SECONDARY8BIT 0x00000400 #define MA_DSCAPS_SECONDARY16BIT 0x00000800 #define MA_DSBCAPS_PRIMARYBUFFER 0x00000001 #define MA_DSBCAPS_STATIC 0x00000002 #define MA_DSBCAPS_LOCHARDWARE 0x00000004 #define MA_DSBCAPS_LOCSOFTWARE 0x00000008 #define MA_DSBCAPS_CTRL3D 0x00000010 #define MA_DSBCAPS_CTRLFREQUENCY 0x00000020 #define MA_DSBCAPS_CTRLPAN 0x00000040 #define MA_DSBCAPS_CTRLVOLUME 0x00000080 #define MA_DSBCAPS_CTRLPOSITIONNOTIFY 0x00000100 #define MA_DSBCAPS_CTRLFX 0x00000200 #define MA_DSBCAPS_STICKYFOCUS 0x00004000 #define MA_DSBCAPS_GLOBALFOCUS 0x00008000 #define MA_DSBCAPS_GETCURRENTPOSITION2 0x00010000 #define MA_DSBCAPS_MUTE3DATMAXDISTANCE 0x00020000 #define MA_DSBCAPS_LOCDEFER 0x00040000 #define MA_DSBCAPS_TRUEPLAYPOSITION 0x00080000 #define MA_DSBPLAY_LOOPING 0x00000001 #define MA_DSBPLAY_LOCHARDWARE 0x00000002 #define MA_DSBPLAY_LOCSOFTWARE 0x00000004 #define MA_DSBPLAY_TERMINATEBY_TIME 0x00000008 #define MA_DSBPLAY_TERMINATEBY_DISTANCE 0x00000010 #define MA_DSBPLAY_TERMINATEBY_PRIORITY 0x00000020 #define MA_DSBSTATUS_PLAYING 0x00000001 #define MA_DSBSTATUS_BUFFERLOST 0x00000002 #define MA_DSBSTATUS_LOOPING 0x00000004 #define MA_DSBSTATUS_LOCHARDWARE 0x00000008 #define MA_DSBSTATUS_LOCSOFTWARE 0x00000010 #define MA_DSBSTATUS_TERMINATED 0x00000020 #define MA_DSCBSTART_LOOPING 0x00000001 typedef struct { DWORD dwSize; DWORD dwFlags; DWORD dwBufferBytes; DWORD dwReserved; MA_WAVEFORMATEX* lpwfxFormat; GUID guid3DAlgorithm; } MA_DSBUFFERDESC; typedef struct { DWORD dwSize; DWORD dwFlags; DWORD dwBufferBytes; DWORD dwReserved; MA_WAVEFORMATEX* lpwfxFormat; DWORD dwFXCount; void* lpDSCFXDesc; /* <-- miniaudio doesn't use this, so set to void*. */ } MA_DSCBUFFERDESC; typedef struct { DWORD dwSize; DWORD dwFlags; DWORD dwMinSecondarySampleRate; DWORD dwMaxSecondarySampleRate; DWORD dwPrimaryBuffers; DWORD dwMaxHwMixingAllBuffers; DWORD dwMaxHwMixingStaticBuffers; DWORD dwMaxHwMixingStreamingBuffers; DWORD dwFreeHwMixingAllBuffers; DWORD dwFreeHwMixingStaticBuffers; DWORD dwFreeHwMixingStreamingBuffers; DWORD dwMaxHw3DAllBuffers; DWORD dwMaxHw3DStaticBuffers; DWORD dwMaxHw3DStreamingBuffers; DWORD dwFreeHw3DAllBuffers; DWORD dwFreeHw3DStaticBuffers; DWORD dwFreeHw3DStreamingBuffers; DWORD dwTotalHwMemBytes; DWORD dwFreeHwMemBytes; DWORD dwMaxContigFreeHwMemBytes; DWORD dwUnlockTransferRateHwBuffers; DWORD dwPlayCpuOverheadSwBuffers; DWORD dwReserved1; DWORD dwReserved2; } MA_DSCAPS; typedef struct { DWORD dwSize; DWORD dwFlags; DWORD dwBufferBytes; DWORD dwUnlockTransferRate; DWORD dwPlayCpuOverhead; } MA_DSBCAPS; typedef struct { DWORD dwSize; DWORD dwFlags; DWORD dwFormats; DWORD dwChannels; } MA_DSCCAPS; typedef struct { DWORD dwSize; DWORD dwFlags; DWORD dwBufferBytes; DWORD dwReserved; } MA_DSCBCAPS; typedef struct { DWORD dwOffset; HANDLE hEventNotify; } MA_DSBPOSITIONNOTIFY; typedef struct ma_IDirectSound ma_IDirectSound; typedef struct ma_IDirectSoundBuffer ma_IDirectSoundBuffer; typedef struct ma_IDirectSoundCapture ma_IDirectSoundCapture; typedef struct ma_IDirectSoundCaptureBuffer ma_IDirectSoundCaptureBuffer; typedef struct ma_IDirectSoundNotify ma_IDirectSoundNotify; /* COM objects. The way these work is that you have a vtable (a list of function pointers, kind of like how C++ works internally), and then you have a structure with a single member, which is a pointer to the vtable. The vtable is where the methods of the object are defined. Methods need to be in a specific order, and parent classes need to have their methods declared first. */ /* IDirectSound */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSound* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSound* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSound* pThis); /* IDirectSound */ HRESULT (STDMETHODCALLTYPE * CreateSoundBuffer) (ma_IDirectSound* pThis, const MA_DSBUFFERDESC* pDSBufferDesc, ma_IDirectSoundBuffer** ppDSBuffer, void* pUnkOuter); HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSound* pThis, MA_DSCAPS* pDSCaps); HRESULT (STDMETHODCALLTYPE * DuplicateSoundBuffer)(ma_IDirectSound* pThis, ma_IDirectSoundBuffer* pDSBufferOriginal, ma_IDirectSoundBuffer** ppDSBufferDuplicate); HRESULT (STDMETHODCALLTYPE * SetCooperativeLevel) (ma_IDirectSound* pThis, HWND hwnd, DWORD dwLevel); HRESULT (STDMETHODCALLTYPE * Compact) (ma_IDirectSound* pThis); HRESULT (STDMETHODCALLTYPE * GetSpeakerConfig) (ma_IDirectSound* pThis, DWORD* pSpeakerConfig); HRESULT (STDMETHODCALLTYPE * SetSpeakerConfig) (ma_IDirectSound* pThis, DWORD dwSpeakerConfig); HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSound* pThis, const GUID* pGuidDevice); } ma_IDirectSoundVtbl; struct ma_IDirectSound { ma_IDirectSoundVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IDirectSound_QueryInterface(ma_IDirectSound* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IDirectSound_AddRef(ma_IDirectSound* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IDirectSound_Release(ma_IDirectSound* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IDirectSound_CreateSoundBuffer(ma_IDirectSound* pThis, const MA_DSBUFFERDESC* pDSBufferDesc, ma_IDirectSoundBuffer** ppDSBuffer, void* pUnkOuter) { return pThis->lpVtbl->CreateSoundBuffer(pThis, pDSBufferDesc, ppDSBuffer, pUnkOuter); } static MA_INLINE HRESULT ma_IDirectSound_GetCaps(ma_IDirectSound* pThis, MA_DSCAPS* pDSCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCaps); } static MA_INLINE HRESULT ma_IDirectSound_DuplicateSoundBuffer(ma_IDirectSound* pThis, ma_IDirectSoundBuffer* pDSBufferOriginal, ma_IDirectSoundBuffer** ppDSBufferDuplicate) { return pThis->lpVtbl->DuplicateSoundBuffer(pThis, pDSBufferOriginal, ppDSBufferDuplicate); } static MA_INLINE HRESULT ma_IDirectSound_SetCooperativeLevel(ma_IDirectSound* pThis, HWND hwnd, DWORD dwLevel) { return pThis->lpVtbl->SetCooperativeLevel(pThis, hwnd, dwLevel); } static MA_INLINE HRESULT ma_IDirectSound_Compact(ma_IDirectSound* pThis) { return pThis->lpVtbl->Compact(pThis); } static MA_INLINE HRESULT ma_IDirectSound_GetSpeakerConfig(ma_IDirectSound* pThis, DWORD* pSpeakerConfig) { return pThis->lpVtbl->GetSpeakerConfig(pThis, pSpeakerConfig); } static MA_INLINE HRESULT ma_IDirectSound_SetSpeakerConfig(ma_IDirectSound* pThis, DWORD dwSpeakerConfig) { return pThis->lpVtbl->SetSpeakerConfig(pThis, dwSpeakerConfig); } static MA_INLINE HRESULT ma_IDirectSound_Initialize(ma_IDirectSound* pThis, const GUID* pGuidDevice) { return pThis->lpVtbl->Initialize(pThis, pGuidDevice); } /* IDirectSoundBuffer */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundBuffer* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundBuffer* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundBuffer* pThis); /* IDirectSoundBuffer */ HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundBuffer* pThis, MA_DSBCAPS* pDSBufferCaps); HRESULT (STDMETHODCALLTYPE * GetCurrentPosition)(ma_IDirectSoundBuffer* pThis, DWORD* pCurrentPlayCursor, DWORD* pCurrentWriteCursor); HRESULT (STDMETHODCALLTYPE * GetFormat) (ma_IDirectSoundBuffer* pThis, MA_WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten); HRESULT (STDMETHODCALLTYPE * GetVolume) (ma_IDirectSoundBuffer* pThis, LONG* pVolume); HRESULT (STDMETHODCALLTYPE * GetPan) (ma_IDirectSoundBuffer* pThis, LONG* pPan); HRESULT (STDMETHODCALLTYPE * GetFrequency) (ma_IDirectSoundBuffer* pThis, DWORD* pFrequency); HRESULT (STDMETHODCALLTYPE * GetStatus) (ma_IDirectSoundBuffer* pThis, DWORD* pStatus); HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundBuffer* pThis, ma_IDirectSound* pDirectSound, const MA_DSBUFFERDESC* pDSBufferDesc); HRESULT (STDMETHODCALLTYPE * Lock) (ma_IDirectSoundBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags); HRESULT (STDMETHODCALLTYPE * Play) (ma_IDirectSoundBuffer* pThis, DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags); HRESULT (STDMETHODCALLTYPE * SetCurrentPosition)(ma_IDirectSoundBuffer* pThis, DWORD dwNewPosition); HRESULT (STDMETHODCALLTYPE * SetFormat) (ma_IDirectSoundBuffer* pThis, const MA_WAVEFORMATEX* pFormat); HRESULT (STDMETHODCALLTYPE * SetVolume) (ma_IDirectSoundBuffer* pThis, LONG volume); HRESULT (STDMETHODCALLTYPE * SetPan) (ma_IDirectSoundBuffer* pThis, LONG pan); HRESULT (STDMETHODCALLTYPE * SetFrequency) (ma_IDirectSoundBuffer* pThis, DWORD dwFrequency); HRESULT (STDMETHODCALLTYPE * Stop) (ma_IDirectSoundBuffer* pThis); HRESULT (STDMETHODCALLTYPE * Unlock) (ma_IDirectSoundBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2); HRESULT (STDMETHODCALLTYPE * Restore) (ma_IDirectSoundBuffer* pThis); } ma_IDirectSoundBufferVtbl; struct ma_IDirectSoundBuffer { ma_IDirectSoundBufferVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IDirectSoundBuffer_QueryInterface(ma_IDirectSoundBuffer* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IDirectSoundBuffer_AddRef(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IDirectSoundBuffer_Release(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetCaps(ma_IDirectSoundBuffer* pThis, MA_DSBCAPS* pDSBufferCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSBufferCaps); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetCurrentPosition(ma_IDirectSoundBuffer* pThis, DWORD* pCurrentPlayCursor, DWORD* pCurrentWriteCursor) { return pThis->lpVtbl->GetCurrentPosition(pThis, pCurrentPlayCursor, pCurrentWriteCursor); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetFormat(ma_IDirectSoundBuffer* pThis, MA_WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten) { return pThis->lpVtbl->GetFormat(pThis, pFormat, dwSizeAllocated, pSizeWritten); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetVolume(ma_IDirectSoundBuffer* pThis, LONG* pVolume) { return pThis->lpVtbl->GetVolume(pThis, pVolume); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetPan(ma_IDirectSoundBuffer* pThis, LONG* pPan) { return pThis->lpVtbl->GetPan(pThis, pPan); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetFrequency(ma_IDirectSoundBuffer* pThis, DWORD* pFrequency) { return pThis->lpVtbl->GetFrequency(pThis, pFrequency); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetStatus(ma_IDirectSoundBuffer* pThis, DWORD* pStatus) { return pThis->lpVtbl->GetStatus(pThis, pStatus); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_Initialize(ma_IDirectSoundBuffer* pThis, ma_IDirectSound* pDirectSound, const MA_DSBUFFERDESC* pDSBufferDesc) { return pThis->lpVtbl->Initialize(pThis, pDirectSound, pDSBufferDesc); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_Lock(ma_IDirectSoundBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags) { return pThis->lpVtbl->Lock(pThis, dwOffset, dwBytes, ppAudioPtr1, pAudioBytes1, ppAudioPtr2, pAudioBytes2, dwFlags); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_Play(ma_IDirectSoundBuffer* pThis, DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags) { return pThis->lpVtbl->Play(pThis, dwReserved1, dwPriority, dwFlags); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetCurrentPosition(ma_IDirectSoundBuffer* pThis, DWORD dwNewPosition) { return pThis->lpVtbl->SetCurrentPosition(pThis, dwNewPosition); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetFormat(ma_IDirectSoundBuffer* pThis, const MA_WAVEFORMATEX* pFormat) { return pThis->lpVtbl->SetFormat(pThis, pFormat); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetVolume(ma_IDirectSoundBuffer* pThis, LONG volume) { return pThis->lpVtbl->SetVolume(pThis, volume); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetPan(ma_IDirectSoundBuffer* pThis, LONG pan) { return pThis->lpVtbl->SetPan(pThis, pan); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetFrequency(ma_IDirectSoundBuffer* pThis, DWORD dwFrequency) { return pThis->lpVtbl->SetFrequency(pThis, dwFrequency); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_Stop(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Stop(pThis); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_Unlock(ma_IDirectSoundBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2) { return pThis->lpVtbl->Unlock(pThis, pAudioPtr1, dwAudioBytes1, pAudioPtr2, dwAudioBytes2); } static MA_INLINE HRESULT ma_IDirectSoundBuffer_Restore(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Restore(pThis); } /* IDirectSoundCapture */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundCapture* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundCapture* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundCapture* pThis); /* IDirectSoundCapture */ HRESULT (STDMETHODCALLTYPE * CreateCaptureBuffer)(ma_IDirectSoundCapture* pThis, const MA_DSCBUFFERDESC* pDSCBufferDesc, ma_IDirectSoundCaptureBuffer** ppDSCBuffer, void* pUnkOuter); HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundCapture* pThis, MA_DSCCAPS* pDSCCaps); HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundCapture* pThis, const GUID* pGuidDevice); } ma_IDirectSoundCaptureVtbl; struct ma_IDirectSoundCapture { ma_IDirectSoundCaptureVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IDirectSoundCapture_QueryInterface (ma_IDirectSoundCapture* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IDirectSoundCapture_AddRef (ma_IDirectSoundCapture* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IDirectSoundCapture_Release (ma_IDirectSoundCapture* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IDirectSoundCapture_CreateCaptureBuffer(ma_IDirectSoundCapture* pThis, const MA_DSCBUFFERDESC* pDSCBufferDesc, ma_IDirectSoundCaptureBuffer** ppDSCBuffer, void* pUnkOuter) { return pThis->lpVtbl->CreateCaptureBuffer(pThis, pDSCBufferDesc, ppDSCBuffer, pUnkOuter); } static MA_INLINE HRESULT ma_IDirectSoundCapture_GetCaps (ma_IDirectSoundCapture* pThis, MA_DSCCAPS* pDSCCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCCaps); } static MA_INLINE HRESULT ma_IDirectSoundCapture_Initialize (ma_IDirectSoundCapture* pThis, const GUID* pGuidDevice) { return pThis->lpVtbl->Initialize(pThis, pGuidDevice); } /* IDirectSoundCaptureBuffer */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundCaptureBuffer* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundCaptureBuffer* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundCaptureBuffer* pThis); /* IDirectSoundCaptureBuffer */ HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundCaptureBuffer* pThis, MA_DSCBCAPS* pDSCBCaps); HRESULT (STDMETHODCALLTYPE * GetCurrentPosition)(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pCapturePosition, DWORD* pReadPosition); HRESULT (STDMETHODCALLTYPE * GetFormat) (ma_IDirectSoundCaptureBuffer* pThis, MA_WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten); HRESULT (STDMETHODCALLTYPE * GetStatus) (ma_IDirectSoundCaptureBuffer* pThis, DWORD* pStatus); HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundCaptureBuffer* pThis, ma_IDirectSoundCapture* pDirectSoundCapture, const MA_DSCBUFFERDESC* pDSCBufferDesc); HRESULT (STDMETHODCALLTYPE * Lock) (ma_IDirectSoundCaptureBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags); HRESULT (STDMETHODCALLTYPE * Start) (ma_IDirectSoundCaptureBuffer* pThis, DWORD dwFlags); HRESULT (STDMETHODCALLTYPE * Stop) (ma_IDirectSoundCaptureBuffer* pThis); HRESULT (STDMETHODCALLTYPE * Unlock) (ma_IDirectSoundCaptureBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2); } ma_IDirectSoundCaptureBufferVtbl; struct ma_IDirectSoundCaptureBuffer { ma_IDirectSoundCaptureBufferVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_QueryInterface(ma_IDirectSoundCaptureBuffer* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IDirectSoundCaptureBuffer_AddRef(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IDirectSoundCaptureBuffer_Release(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetCaps(ma_IDirectSoundCaptureBuffer* pThis, MA_DSCBCAPS* pDSCBCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCBCaps); } static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetCurrentPosition(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pCapturePosition, DWORD* pReadPosition) { return pThis->lpVtbl->GetCurrentPosition(pThis, pCapturePosition, pReadPosition); } static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetFormat(ma_IDirectSoundCaptureBuffer* pThis, MA_WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten) { return pThis->lpVtbl->GetFormat(pThis, pFormat, dwSizeAllocated, pSizeWritten); } static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetStatus(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pStatus) { return pThis->lpVtbl->GetStatus(pThis, pStatus); } static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Initialize(ma_IDirectSoundCaptureBuffer* pThis, ma_IDirectSoundCapture* pDirectSoundCapture, const MA_DSCBUFFERDESC* pDSCBufferDesc) { return pThis->lpVtbl->Initialize(pThis, pDirectSoundCapture, pDSCBufferDesc); } static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Lock(ma_IDirectSoundCaptureBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags) { return pThis->lpVtbl->Lock(pThis, dwOffset, dwBytes, ppAudioPtr1, pAudioBytes1, ppAudioPtr2, pAudioBytes2, dwFlags); } static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Start(ma_IDirectSoundCaptureBuffer* pThis, DWORD dwFlags) { return pThis->lpVtbl->Start(pThis, dwFlags); } static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Stop(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->Stop(pThis); } static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Unlock(ma_IDirectSoundCaptureBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2) { return pThis->lpVtbl->Unlock(pThis, pAudioPtr1, dwAudioBytes1, pAudioPtr2, dwAudioBytes2); } /* IDirectSoundNotify */ typedef struct { /* IUnknown */ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundNotify* pThis, const IID* const riid, void** ppObject); ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundNotify* pThis); ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundNotify* pThis); /* IDirectSoundNotify */ HRESULT (STDMETHODCALLTYPE * SetNotificationPositions)(ma_IDirectSoundNotify* pThis, DWORD dwPositionNotifies, const MA_DSBPOSITIONNOTIFY* pPositionNotifies); } ma_IDirectSoundNotifyVtbl; struct ma_IDirectSoundNotify { ma_IDirectSoundNotifyVtbl* lpVtbl; }; static MA_INLINE HRESULT ma_IDirectSoundNotify_QueryInterface(ma_IDirectSoundNotify* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } static MA_INLINE ULONG ma_IDirectSoundNotify_AddRef(ma_IDirectSoundNotify* pThis) { return pThis->lpVtbl->AddRef(pThis); } static MA_INLINE ULONG ma_IDirectSoundNotify_Release(ma_IDirectSoundNotify* pThis) { return pThis->lpVtbl->Release(pThis); } static MA_INLINE HRESULT ma_IDirectSoundNotify_SetNotificationPositions(ma_IDirectSoundNotify* pThis, DWORD dwPositionNotifies, const MA_DSBPOSITIONNOTIFY* pPositionNotifies) { return pThis->lpVtbl->SetNotificationPositions(pThis, dwPositionNotifies, pPositionNotifies); } typedef BOOL (CALLBACK * ma_DSEnumCallbackAProc) (GUID* pDeviceGUID, const char* pDeviceDescription, const char* pModule, void* pContext); typedef HRESULT (WINAPI * ma_DirectSoundCreateProc) (const GUID* pcGuidDevice, ma_IDirectSound** ppDS8, ma_IUnknown* pUnkOuter); typedef HRESULT (WINAPI * ma_DirectSoundEnumerateAProc) (ma_DSEnumCallbackAProc pDSEnumCallback, void* pContext); typedef HRESULT (WINAPI * ma_DirectSoundCaptureCreateProc) (const GUID* pcGuidDevice, ma_IDirectSoundCapture** ppDSC8, ma_IUnknown* pUnkOuter); typedef HRESULT (WINAPI * ma_DirectSoundCaptureEnumerateAProc)(ma_DSEnumCallbackAProc pDSEnumCallback, void* pContext); static ma_uint32 ma_get_best_sample_rate_within_range(ma_uint32 sampleRateMin, ma_uint32 sampleRateMax) { /* Normalize the range in case we were given something stupid. */ if (sampleRateMin < (ma_uint32)ma_standard_sample_rate_min) { sampleRateMin = (ma_uint32)ma_standard_sample_rate_min; } if (sampleRateMax > (ma_uint32)ma_standard_sample_rate_max) { sampleRateMax = (ma_uint32)ma_standard_sample_rate_max; } if (sampleRateMin > sampleRateMax) { sampleRateMin = sampleRateMax; } if (sampleRateMin == sampleRateMax) { return sampleRateMax; } else { size_t iStandardRate; for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) { ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate]; if (standardRate >= sampleRateMin && standardRate <= sampleRateMax) { return standardRate; } } } /* Should never get here. */ MA_ASSERT(MA_FALSE); return 0; } /* Retrieves the channel count and channel map for the given speaker configuration. If the speaker configuration is unknown, the channel count and channel map will be left unmodified. */ static void ma_get_channels_from_speaker_config__dsound(DWORD speakerConfig, WORD* pChannelsOut, DWORD* pChannelMapOut) { WORD channels; DWORD channelMap; channels = 0; if (pChannelsOut != NULL) { channels = *pChannelsOut; } channelMap = 0; if (pChannelMapOut != NULL) { channelMap = *pChannelMapOut; } /* The speaker configuration is a combination of speaker config and speaker geometry. The lower 8 bits is what we care about. The upper 16 bits is for the geometry. */ switch ((BYTE)(speakerConfig)) { case 1 /*DSSPEAKER_HEADPHONE*/: channels = 2; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break; case 2 /*DSSPEAKER_MONO*/: channels = 1; channelMap = SPEAKER_FRONT_CENTER; break; case 3 /*DSSPEAKER_QUAD*/: channels = 4; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break; case 4 /*DSSPEAKER_STEREO*/: channels = 2; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break; case 5 /*DSSPEAKER_SURROUND*/: channels = 4; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER; break; case 6 /*DSSPEAKER_5POINT1_BACK*/ /*DSSPEAKER_5POINT1*/: channels = 6; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break; case 7 /*DSSPEAKER_7POINT1_WIDE*/ /*DSSPEAKER_7POINT1*/: channels = 8; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER; break; case 8 /*DSSPEAKER_7POINT1_SURROUND*/: channels = 8; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; case 9 /*DSSPEAKER_5POINT1_SURROUND*/: channels = 6; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; default: break; } if (pChannelsOut != NULL) { *pChannelsOut = channels; } if (pChannelMapOut != NULL) { *pChannelMapOut = channelMap; } } static ma_result ma_context_create_IDirectSound__dsound(ma_context* pContext, ma_share_mode shareMode, const ma_device_id* pDeviceID, ma_IDirectSound** ppDirectSound) { ma_IDirectSound* pDirectSound; HWND hWnd; HRESULT hr; MA_ASSERT(pContext != NULL); MA_ASSERT(ppDirectSound != NULL); *ppDirectSound = NULL; pDirectSound = NULL; if (FAILED(((ma_DirectSoundCreateProc)pContext->dsound.DirectSoundCreate)((pDeviceID == NULL) ? NULL : (const GUID*)pDeviceID->dsound, &pDirectSound, NULL))) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[DirectSound] DirectSoundCreate() failed for playback device."); return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } /* The cooperative level must be set before doing anything else. */ hWnd = (HWND)pContext->dsound.hWnd; if (hWnd == 0) { hWnd = ((MA_PFN_GetForegroundWindow)pContext->win32.GetForegroundWindow)(); if (hWnd == 0) { hWnd = ((MA_PFN_GetDesktopWindow)pContext->win32.GetDesktopWindow)(); } } hr = ma_IDirectSound_SetCooperativeLevel(pDirectSound, hWnd, (shareMode == ma_share_mode_exclusive) ? MA_DSSCL_EXCLUSIVE : MA_DSSCL_PRIORITY); if (FAILED(hr)) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_SetCooperateiveLevel() failed for playback device."); return ma_result_from_HRESULT(hr); } *ppDirectSound = pDirectSound; return MA_SUCCESS; } static ma_result ma_context_create_IDirectSoundCapture__dsound(ma_context* pContext, ma_share_mode shareMode, const ma_device_id* pDeviceID, ma_IDirectSoundCapture** ppDirectSoundCapture) { ma_IDirectSoundCapture* pDirectSoundCapture; HRESULT hr; MA_ASSERT(pContext != NULL); MA_ASSERT(ppDirectSoundCapture != NULL); /* DirectSound does not support exclusive mode for capture. */ if (shareMode == ma_share_mode_exclusive) { return MA_SHARE_MODE_NOT_SUPPORTED; } *ppDirectSoundCapture = NULL; pDirectSoundCapture = NULL; hr = ((ma_DirectSoundCaptureCreateProc)pContext->dsound.DirectSoundCaptureCreate)((pDeviceID == NULL) ? NULL : (const GUID*)pDeviceID->dsound, &pDirectSoundCapture, NULL); if (FAILED(hr)) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[DirectSound] DirectSoundCaptureCreate() failed for capture device."); return ma_result_from_HRESULT(hr); } *ppDirectSoundCapture = pDirectSoundCapture; return MA_SUCCESS; } static ma_result ma_context_get_format_info_for_IDirectSoundCapture__dsound(ma_context* pContext, ma_IDirectSoundCapture* pDirectSoundCapture, WORD* pChannels, WORD* pBitsPerSample, DWORD* pSampleRate) { HRESULT hr; MA_DSCCAPS caps; WORD bitsPerSample; DWORD sampleRate; MA_ASSERT(pContext != NULL); MA_ASSERT(pDirectSoundCapture != NULL); if (pChannels) { *pChannels = 0; } if (pBitsPerSample) { *pBitsPerSample = 0; } if (pSampleRate) { *pSampleRate = 0; } MA_ZERO_OBJECT(&caps); caps.dwSize = sizeof(caps); hr = ma_IDirectSoundCapture_GetCaps(pDirectSoundCapture, &caps); if (FAILED(hr)) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCapture_GetCaps() failed for capture device."); return ma_result_from_HRESULT(hr); } if (pChannels) { *pChannels = (WORD)caps.dwChannels; } /* The device can support multiple formats. We just go through the different formats in order of priority and pick the first one. This the same type of system as the WinMM backend. */ bitsPerSample = 16; sampleRate = 48000; if (caps.dwChannels == 1) { if ((caps.dwFormats & WAVE_FORMAT_48M16) != 0) { sampleRate = 48000; } else if ((caps.dwFormats & WAVE_FORMAT_44M16) != 0) { sampleRate = 44100; } else if ((caps.dwFormats & WAVE_FORMAT_2M16) != 0) { sampleRate = 22050; } else if ((caps.dwFormats & WAVE_FORMAT_1M16) != 0) { sampleRate = 11025; } else if ((caps.dwFormats & WAVE_FORMAT_96M16) != 0) { sampleRate = 96000; } else { bitsPerSample = 8; if ((caps.dwFormats & WAVE_FORMAT_48M08) != 0) { sampleRate = 48000; } else if ((caps.dwFormats & WAVE_FORMAT_44M08) != 0) { sampleRate = 44100; } else if ((caps.dwFormats & WAVE_FORMAT_2M08) != 0) { sampleRate = 22050; } else if ((caps.dwFormats & WAVE_FORMAT_1M08) != 0) { sampleRate = 11025; } else if ((caps.dwFormats & WAVE_FORMAT_96M08) != 0) { sampleRate = 96000; } else { bitsPerSample = 16; /* Didn't find it. Just fall back to 16-bit. */ } } } else if (caps.dwChannels == 2) { if ((caps.dwFormats & WAVE_FORMAT_48S16) != 0) { sampleRate = 48000; } else if ((caps.dwFormats & WAVE_FORMAT_44S16) != 0) { sampleRate = 44100; } else if ((caps.dwFormats & WAVE_FORMAT_2S16) != 0) { sampleRate = 22050; } else if ((caps.dwFormats & WAVE_FORMAT_1S16) != 0) { sampleRate = 11025; } else if ((caps.dwFormats & WAVE_FORMAT_96S16) != 0) { sampleRate = 96000; } else { bitsPerSample = 8; if ((caps.dwFormats & WAVE_FORMAT_48S08) != 0) { sampleRate = 48000; } else if ((caps.dwFormats & WAVE_FORMAT_44S08) != 0) { sampleRate = 44100; } else if ((caps.dwFormats & WAVE_FORMAT_2S08) != 0) { sampleRate = 22050; } else if ((caps.dwFormats & WAVE_FORMAT_1S08) != 0) { sampleRate = 11025; } else if ((caps.dwFormats & WAVE_FORMAT_96S08) != 0) { sampleRate = 96000; } else { bitsPerSample = 16; /* Didn't find it. Just fall back to 16-bit. */ } } } if (pBitsPerSample) { *pBitsPerSample = bitsPerSample; } if (pSampleRate) { *pSampleRate = sampleRate; } return MA_SUCCESS; } typedef struct { ma_context* pContext; ma_device_type deviceType; ma_enum_devices_callback_proc callback; void* pUserData; ma_bool32 terminated; } ma_context_enumerate_devices_callback_data__dsound; static BOOL CALLBACK ma_context_enumerate_devices_callback__dsound(GUID* lpGuid, const char* lpcstrDescription, const char* lpcstrModule, void* lpContext) { ma_context_enumerate_devices_callback_data__dsound* pData = (ma_context_enumerate_devices_callback_data__dsound*)lpContext; ma_device_info deviceInfo; (void)lpcstrModule; MA_ZERO_OBJECT(&deviceInfo); /* ID. */ if (lpGuid != NULL) { MA_COPY_MEMORY(deviceInfo.id.dsound, lpGuid, 16); } else { MA_ZERO_MEMORY(deviceInfo.id.dsound, 16); deviceInfo.isDefault = MA_TRUE; } /* Name / Description */ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), lpcstrDescription, (size_t)-1); /* Call the callback function, but make sure we stop enumerating if the callee requested so. */ MA_ASSERT(pData != NULL); pData->terminated = (pData->callback(pData->pContext, pData->deviceType, &deviceInfo, pData->pUserData) == MA_FALSE); if (pData->terminated) { return FALSE; /* Stop enumeration. */ } else { return TRUE; /* Continue enumeration. */ } } static ma_result ma_context_enumerate_devices__dsound(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { ma_context_enumerate_devices_callback_data__dsound data; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); data.pContext = pContext; data.callback = callback; data.pUserData = pUserData; data.terminated = MA_FALSE; /* Playback. */ if (!data.terminated) { data.deviceType = ma_device_type_playback; ((ma_DirectSoundEnumerateAProc)pContext->dsound.DirectSoundEnumerateA)(ma_context_enumerate_devices_callback__dsound, &data); } /* Capture. */ if (!data.terminated) { data.deviceType = ma_device_type_capture; ((ma_DirectSoundCaptureEnumerateAProc)pContext->dsound.DirectSoundCaptureEnumerateA)(ma_context_enumerate_devices_callback__dsound, &data); } return MA_SUCCESS; } typedef struct { const ma_device_id* pDeviceID; ma_device_info* pDeviceInfo; ma_bool32 found; } ma_context_get_device_info_callback_data__dsound; static BOOL CALLBACK ma_context_get_device_info_callback__dsound(GUID* lpGuid, const char* lpcstrDescription, const char* lpcstrModule, void* lpContext) { ma_context_get_device_info_callback_data__dsound* pData = (ma_context_get_device_info_callback_data__dsound*)lpContext; MA_ASSERT(pData != NULL); if ((pData->pDeviceID == NULL || ma_is_guid_null(pData->pDeviceID->dsound)) && (lpGuid == NULL || ma_is_guid_null(lpGuid))) { /* Default device. */ ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), lpcstrDescription, (size_t)-1); pData->pDeviceInfo->isDefault = MA_TRUE; pData->found = MA_TRUE; return FALSE; /* Stop enumeration. */ } else { /* Not the default device. */ if (lpGuid != NULL && pData->pDeviceID != NULL) { if (memcmp(pData->pDeviceID->dsound, lpGuid, sizeof(pData->pDeviceID->dsound)) == 0) { ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), lpcstrDescription, (size_t)-1); pData->found = MA_TRUE; return FALSE; /* Stop enumeration. */ } } } (void)lpcstrModule; return TRUE; } static ma_result ma_context_get_device_info__dsound(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { ma_result result; HRESULT hr; if (pDeviceID != NULL) { ma_context_get_device_info_callback_data__dsound data; /* ID. */ MA_COPY_MEMORY(pDeviceInfo->id.dsound, pDeviceID->dsound, 16); /* Name / Description. This is retrieved by enumerating over each device until we find that one that matches the input ID. */ data.pDeviceID = pDeviceID; data.pDeviceInfo = pDeviceInfo; data.found = MA_FALSE; if (deviceType == ma_device_type_playback) { ((ma_DirectSoundEnumerateAProc)pContext->dsound.DirectSoundEnumerateA)(ma_context_get_device_info_callback__dsound, &data); } else { ((ma_DirectSoundCaptureEnumerateAProc)pContext->dsound.DirectSoundCaptureEnumerateA)(ma_context_get_device_info_callback__dsound, &data); } if (!data.found) { return MA_NO_DEVICE; } } else { /* I don't think there's a way to get the name of the default device with DirectSound. In this case we just need to use defaults. */ /* ID */ MA_ZERO_MEMORY(pDeviceInfo->id.dsound, 16); /* Name / Description */ if (deviceType == ma_device_type_playback) { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); } else { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); } pDeviceInfo->isDefault = MA_TRUE; } /* Retrieving detailed information is slightly different depending on the device type. */ if (deviceType == ma_device_type_playback) { /* Playback. */ ma_IDirectSound* pDirectSound; MA_DSCAPS caps; WORD channels; result = ma_context_create_IDirectSound__dsound(pContext, ma_share_mode_shared, pDeviceID, &pDirectSound); if (result != MA_SUCCESS) { return result; } MA_ZERO_OBJECT(&caps); caps.dwSize = sizeof(caps); hr = ma_IDirectSound_GetCaps(pDirectSound, &caps); if (FAILED(hr)) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_GetCaps() failed for playback device."); return ma_result_from_HRESULT(hr); } /* Channels. Only a single channel count is reported for DirectSound. */ if ((caps.dwFlags & MA_DSCAPS_PRIMARYSTEREO) != 0) { /* It supports at least stereo, but could support more. */ DWORD speakerConfig; channels = 2; /* Look at the speaker configuration to get a better idea on the channel count. */ hr = ma_IDirectSound_GetSpeakerConfig(pDirectSound, &speakerConfig); if (SUCCEEDED(hr)) { ma_get_channels_from_speaker_config__dsound(speakerConfig, &channels, NULL); } } else { /* It does not support stereo, which means we are stuck with mono. */ channels = 1; } /* In DirectSound, our native formats are centered around sample rates. All formats are supported, and we're only reporting a single channel count. However, DirectSound can report a range of supported sample rates. We're only going to include standard rates known by miniaudio in order to keep the size of this within reason. */ if ((caps.dwFlags & MA_DSCAPS_CONTINUOUSRATE) != 0) { /* Multiple sample rates are supported. We'll report in order of our preferred sample rates. */ size_t iStandardSampleRate; for (iStandardSampleRate = 0; iStandardSampleRate < ma_countof(g_maStandardSampleRatePriorities); iStandardSampleRate += 1) { ma_uint32 sampleRate = g_maStandardSampleRatePriorities[iStandardSampleRate]; if (sampleRate >= caps.dwMinSecondarySampleRate && sampleRate <= caps.dwMaxSecondarySampleRate) { pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = ma_format_unknown; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0; pDeviceInfo->nativeDataFormatCount += 1; } } } else { /* Only a single sample rate is supported. */ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = ma_format_unknown; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = caps.dwMaxSecondarySampleRate; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0; pDeviceInfo->nativeDataFormatCount += 1; } ma_IDirectSound_Release(pDirectSound); } else { /* Capture. This is a little different to playback due to the say the supported formats are reported. Technically capture devices can support a number of different formats, but for simplicity and consistency with ma_device_init() I'm just reporting the best format. */ ma_IDirectSoundCapture* pDirectSoundCapture; WORD channels; WORD bitsPerSample; DWORD sampleRate; result = ma_context_create_IDirectSoundCapture__dsound(pContext, ma_share_mode_shared, pDeviceID, &pDirectSoundCapture); if (result != MA_SUCCESS) { return result; } result = ma_context_get_format_info_for_IDirectSoundCapture__dsound(pContext, pDirectSoundCapture, &channels, &bitsPerSample, &sampleRate); if (result != MA_SUCCESS) { ma_IDirectSoundCapture_Release(pDirectSoundCapture); return result; } ma_IDirectSoundCapture_Release(pDirectSoundCapture); /* The format is always an integer format and is based on the bits per sample. */ if (bitsPerSample == 8) { pDeviceInfo->nativeDataFormats[0].format = ma_format_u8; } else if (bitsPerSample == 16) { pDeviceInfo->nativeDataFormats[0].format = ma_format_s16; } else if (bitsPerSample == 24) { pDeviceInfo->nativeDataFormats[0].format = ma_format_s24; } else if (bitsPerSample == 32) { pDeviceInfo->nativeDataFormats[0].format = ma_format_s32; } else { return MA_FORMAT_NOT_SUPPORTED; } pDeviceInfo->nativeDataFormats[0].channels = channels; pDeviceInfo->nativeDataFormats[0].sampleRate = sampleRate; pDeviceInfo->nativeDataFormats[0].flags = 0; pDeviceInfo->nativeDataFormatCount = 1; } return MA_SUCCESS; } static ma_result ma_device_uninit__dsound(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->dsound.pCaptureBuffer != NULL) { ma_IDirectSoundCaptureBuffer_Release((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); } if (pDevice->dsound.pCapture != NULL) { ma_IDirectSoundCapture_Release((ma_IDirectSoundCapture*)pDevice->dsound.pCapture); } if (pDevice->dsound.pPlaybackBuffer != NULL) { ma_IDirectSoundBuffer_Release((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer); } if (pDevice->dsound.pPlaybackPrimaryBuffer != NULL) { ma_IDirectSoundBuffer_Release((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer); } if (pDevice->dsound.pPlayback != NULL) { ma_IDirectSound_Release((ma_IDirectSound*)pDevice->dsound.pPlayback); } return MA_SUCCESS; } static ma_result ma_config_to_WAVEFORMATEXTENSIBLE(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* pChannelMap, MA_WAVEFORMATEXTENSIBLE* pWF) { GUID subformat; if (format == ma_format_unknown) { format = MA_DEFAULT_FORMAT; } if (channels == 0) { channels = MA_DEFAULT_CHANNELS; } if (sampleRate == 0) { sampleRate = MA_DEFAULT_SAMPLE_RATE; } switch (format) { case ma_format_u8: case ma_format_s16: case ma_format_s24: /*case ma_format_s24_32:*/ case ma_format_s32: { subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM; } break; case ma_format_f32: { subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; } break; default: return MA_FORMAT_NOT_SUPPORTED; } MA_ZERO_OBJECT(pWF); pWF->cbSize = sizeof(*pWF); pWF->wFormatTag = WAVE_FORMAT_EXTENSIBLE; pWF->nChannels = (WORD)channels; pWF->nSamplesPerSec = (DWORD)sampleRate; pWF->wBitsPerSample = (WORD)(ma_get_bytes_per_sample(format)*8); pWF->nBlockAlign = (WORD)(pWF->nChannels * pWF->wBitsPerSample / 8); pWF->nAvgBytesPerSec = pWF->nBlockAlign * pWF->nSamplesPerSec; pWF->Samples.wValidBitsPerSample = pWF->wBitsPerSample; pWF->dwChannelMask = ma_channel_map_to_channel_mask__win32(pChannelMap, channels); pWF->SubFormat = subformat; return MA_SUCCESS; } static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__dsound(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile) { /* DirectSound has a minimum period size of 20ms. In practice, this doesn't seem to be enough for reliable glitch-free processing so going to use 30ms instead. */ ma_uint32 minPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(30, nativeSampleRate); ma_uint32 periodSizeInFrames; periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, nativeSampleRate, performanceProfile); if (periodSizeInFrames < minPeriodSizeInFrames) { periodSizeInFrames = minPeriodSizeInFrames; } return periodSizeInFrames; } static ma_result ma_device_init__dsound(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { ma_result result; HRESULT hr; MA_ASSERT(pDevice != NULL); MA_ZERO_OBJECT(&pDevice->dsound); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } /* Unfortunately DirectSound uses different APIs and data structures for playback and capture devices. We need to initialize the capture device first because we'll want to match its buffer size and period count on the playback side if we're using full-duplex mode. */ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { MA_WAVEFORMATEXTENSIBLE wf; MA_DSCBUFFERDESC descDS; ma_uint32 periodSizeInFrames; ma_uint32 periodCount; char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */ MA_WAVEFORMATEXTENSIBLE* pActualFormat; result = ma_config_to_WAVEFORMATEXTENSIBLE(pDescriptorCapture->format, pDescriptorCapture->channels, pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, &wf); if (result != MA_SUCCESS) { return result; } result = ma_context_create_IDirectSoundCapture__dsound(pDevice->pContext, pDescriptorCapture->shareMode, pDescriptorCapture->pDeviceID, (ma_IDirectSoundCapture**)&pDevice->dsound.pCapture); if (result != MA_SUCCESS) { ma_device_uninit__dsound(pDevice); return result; } result = ma_context_get_format_info_for_IDirectSoundCapture__dsound(pDevice->pContext, (ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &wf.nChannels, &wf.wBitsPerSample, &wf.nSamplesPerSec); if (result != MA_SUCCESS) { ma_device_uninit__dsound(pDevice); return result; } wf.nBlockAlign = (WORD)(wf.nChannels * wf.wBitsPerSample / 8); wf.nAvgBytesPerSec = wf.nBlockAlign * wf.nSamplesPerSec; wf.Samples.wValidBitsPerSample = wf.wBitsPerSample; wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM; /* The size of the buffer must be a clean multiple of the period count. */ periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__dsound(pDescriptorCapture, wf.nSamplesPerSec, pConfig->performanceProfile); periodCount = (pDescriptorCapture->periodCount > 0) ? pDescriptorCapture->periodCount : MA_DEFAULT_PERIODS; MA_ZERO_OBJECT(&descDS); descDS.dwSize = sizeof(descDS); descDS.dwFlags = 0; descDS.dwBufferBytes = periodSizeInFrames * periodCount * wf.nBlockAlign; descDS.lpwfxFormat = (MA_WAVEFORMATEX*)&wf; hr = ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL); if (FAILED(hr)) { ma_device_uninit__dsound(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCapture_CreateCaptureBuffer() failed for capture device."); return ma_result_from_HRESULT(hr); } /* Get the _actual_ properties of the buffer. */ pActualFormat = (MA_WAVEFORMATEXTENSIBLE*)rawdata; hr = ma_IDirectSoundCaptureBuffer_GetFormat((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, (MA_WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL); if (FAILED(hr)) { ma_device_uninit__dsound(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the capture device's buffer."); return ma_result_from_HRESULT(hr); } /* We can now start setting the output data formats. */ pDescriptorCapture->format = ma_format_from_WAVEFORMATEX((MA_WAVEFORMATEX*)pActualFormat); pDescriptorCapture->channels = pActualFormat->nChannels; pDescriptorCapture->sampleRate = pActualFormat->nSamplesPerSec; /* Get the native channel map based on the channel mask. */ if (pActualFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE) { ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDescriptorCapture->channels, pDescriptorCapture->channelMap); } else { ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDescriptorCapture->channels, pDescriptorCapture->channelMap); } /* After getting the actual format the size of the buffer in frames may have actually changed. However, we want this to be as close to what the user has asked for as possible, so let's go ahead and release the old capture buffer and create a new one in this case. */ if (periodSizeInFrames != (descDS.dwBufferBytes / ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) / periodCount)) { descDS.dwBufferBytes = periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) * periodCount; ma_IDirectSoundCaptureBuffer_Release((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); hr = ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL); if (FAILED(hr)) { ma_device_uninit__dsound(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Second attempt at IDirectSoundCapture_CreateCaptureBuffer() failed for capture device."); return ma_result_from_HRESULT(hr); } } /* DirectSound should give us a buffer exactly the size we asked for. */ pDescriptorCapture->periodSizeInFrames = periodSizeInFrames; pDescriptorCapture->periodCount = periodCount; } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { MA_WAVEFORMATEXTENSIBLE wf; MA_DSBUFFERDESC descDSPrimary; MA_DSCAPS caps; char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */ MA_WAVEFORMATEXTENSIBLE* pActualFormat; ma_uint32 periodSizeInFrames; ma_uint32 periodCount; MA_DSBUFFERDESC descDS; WORD nativeChannelCount; DWORD nativeChannelMask = 0; result = ma_config_to_WAVEFORMATEXTENSIBLE(pDescriptorPlayback->format, pDescriptorPlayback->channels, pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, &wf); if (result != MA_SUCCESS) { return result; } result = ma_context_create_IDirectSound__dsound(pDevice->pContext, pDescriptorPlayback->shareMode, pDescriptorPlayback->pDeviceID, (ma_IDirectSound**)&pDevice->dsound.pPlayback); if (result != MA_SUCCESS) { ma_device_uninit__dsound(pDevice); return result; } MA_ZERO_OBJECT(&descDSPrimary); descDSPrimary.dwSize = sizeof(MA_DSBUFFERDESC); descDSPrimary.dwFlags = MA_DSBCAPS_PRIMARYBUFFER | MA_DSBCAPS_CTRLVOLUME; hr = ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDSPrimary, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackPrimaryBuffer, NULL); if (FAILED(hr)) { ma_device_uninit__dsound(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's primary buffer."); return ma_result_from_HRESULT(hr); } /* We may want to make some adjustments to the format if we are using defaults. */ MA_ZERO_OBJECT(&caps); caps.dwSize = sizeof(caps); hr = ma_IDirectSound_GetCaps((ma_IDirectSound*)pDevice->dsound.pPlayback, &caps); if (FAILED(hr)) { ma_device_uninit__dsound(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_GetCaps() failed for playback device."); return ma_result_from_HRESULT(hr); } if ((caps.dwFlags & MA_DSCAPS_PRIMARYSTEREO) != 0) { DWORD speakerConfig; /* It supports at least stereo, but could support more. */ nativeChannelCount = 2; /* Look at the speaker configuration to get a better idea on the channel count. */ if (SUCCEEDED(ma_IDirectSound_GetSpeakerConfig((ma_IDirectSound*)pDevice->dsound.pPlayback, &speakerConfig))) { ma_get_channels_from_speaker_config__dsound(speakerConfig, &nativeChannelCount, &nativeChannelMask); } } else { /* It does not support stereo, which means we are stuck with mono. */ nativeChannelCount = 1; nativeChannelMask = 0x00000001; } if (pDescriptorPlayback->channels == 0) { wf.nChannels = nativeChannelCount; wf.dwChannelMask = nativeChannelMask; } if (pDescriptorPlayback->sampleRate == 0) { /* We base the sample rate on the values returned by GetCaps(). */ if ((caps.dwFlags & MA_DSCAPS_CONTINUOUSRATE) != 0) { wf.nSamplesPerSec = ma_get_best_sample_rate_within_range(caps.dwMinSecondarySampleRate, caps.dwMaxSecondarySampleRate); } else { wf.nSamplesPerSec = caps.dwMaxSecondarySampleRate; } } wf.nBlockAlign = (WORD)(wf.nChannels * wf.wBitsPerSample / 8); wf.nAvgBytesPerSec = wf.nBlockAlign * wf.nSamplesPerSec; /* From MSDN: The method succeeds even if the hardware does not support the requested format; DirectSound sets the buffer to the closest supported format. To determine whether this has happened, an application can call the GetFormat method for the primary buffer and compare the result with the format that was requested with the SetFormat method. */ hr = ma_IDirectSoundBuffer_SetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (MA_WAVEFORMATEX*)&wf); if (FAILED(hr)) { /* If setting of the format failed we'll try again with some fallback settings. On Windows 98 I have observed that IEEE_FLOAT does not work. We'll therefore enforce PCM. I also had issues where a sample rate of 48000 did not work correctly. Not sure if it was a driver issue or not, but will use 44100 for the sample rate. */ wf.cbSize = 18; /* NOTE: Don't use sizeof(MA_WAVEFORMATEX) here because it's got an extra 2 bytes due to padding. */ wf.wFormatTag = WAVE_FORMAT_PCM; wf.wBitsPerSample = 16; wf.nChannels = nativeChannelCount; wf.nSamplesPerSec = 44100; wf.nBlockAlign = wf.nChannels * (wf.wBitsPerSample / 8); wf.nAvgBytesPerSec = wf.nSamplesPerSec * wf.nBlockAlign; hr = ma_IDirectSoundBuffer_SetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (MA_WAVEFORMATEX*)&wf); if (FAILED(hr)) { ma_device_uninit__dsound(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to set format of playback device's primary buffer."); return ma_result_from_HRESULT(hr); } } /* Get the _actual_ properties of the buffer. */ pActualFormat = (MA_WAVEFORMATEXTENSIBLE*)rawdata; hr = ma_IDirectSoundBuffer_GetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (MA_WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL); if (FAILED(hr)) { ma_device_uninit__dsound(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the playback device's primary buffer."); return ma_result_from_HRESULT(hr); } /* We now have enough information to start setting some output properties. */ pDescriptorPlayback->format = ma_format_from_WAVEFORMATEX((MA_WAVEFORMATEX*)pActualFormat); pDescriptorPlayback->channels = pActualFormat->nChannels; pDescriptorPlayback->sampleRate = pActualFormat->nSamplesPerSec; /* Get the internal channel map based on the channel mask. */ if (pActualFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE) { ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDescriptorPlayback->channels, pDescriptorPlayback->channelMap); } else { ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDescriptorPlayback->channels, pDescriptorPlayback->channelMap); } /* The size of the buffer must be a clean multiple of the period count. */ periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__dsound(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile); periodCount = (pDescriptorPlayback->periodCount > 0) ? pDescriptorPlayback->periodCount : MA_DEFAULT_PERIODS; /* Meaning of dwFlags (from MSDN): DSBCAPS_CTRLPOSITIONNOTIFY The buffer has position notification capability. DSBCAPS_GLOBALFOCUS With this flag set, an application using DirectSound can continue to play its buffers if the user switches focus to another application, even if the new application uses DirectSound. DSBCAPS_GETCURRENTPOSITION2 In the first version of DirectSound, the play cursor was significantly ahead of the actual playing sound on emulated sound cards; it was directly behind the write cursor. Now, if the DSBCAPS_GETCURRENTPOSITION2 flag is specified, the application can get a more accurate play cursor. */ MA_ZERO_OBJECT(&descDS); descDS.dwSize = sizeof(descDS); descDS.dwFlags = MA_DSBCAPS_CTRLPOSITIONNOTIFY | MA_DSBCAPS_GLOBALFOCUS | MA_DSBCAPS_GETCURRENTPOSITION2; descDS.dwBufferBytes = periodSizeInFrames * periodCount * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels); descDS.lpwfxFormat = (MA_WAVEFORMATEX*)pActualFormat; hr = ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDS, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackBuffer, NULL); if (FAILED(hr)) { ma_device_uninit__dsound(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's secondary buffer."); return ma_result_from_HRESULT(hr); } /* DirectSound should give us a buffer exactly the size we asked for. */ pDescriptorPlayback->periodSizeInFrames = periodSizeInFrames; pDescriptorPlayback->periodCount = periodCount; } return MA_SUCCESS; } static ma_result ma_device_data_loop__dsound(ma_device* pDevice) { ma_result result = MA_SUCCESS; ma_uint32 bpfDeviceCapture = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); ma_uint32 bpfDevicePlayback = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); HRESULT hr; DWORD lockOffsetInBytesCapture; DWORD lockSizeInBytesCapture; DWORD mappedSizeInBytesCapture; DWORD mappedDeviceFramesProcessedCapture; void* pMappedDeviceBufferCapture; DWORD lockOffsetInBytesPlayback; DWORD lockSizeInBytesPlayback; DWORD mappedSizeInBytesPlayback; void* pMappedDeviceBufferPlayback; DWORD prevReadCursorInBytesCapture = 0; DWORD prevPlayCursorInBytesPlayback = 0; ma_bool32 physicalPlayCursorLoopFlagPlayback = 0; DWORD virtualWriteCursorInBytesPlayback = 0; ma_bool32 virtualWriteCursorLoopFlagPlayback = 0; ma_bool32 isPlaybackDeviceStarted = MA_FALSE; ma_uint32 framesWrittenToPlaybackDevice = 0; /* For knowing whether or not the playback device needs to be started. */ ma_uint32 waitTimeInMilliseconds = 1; DWORD playbackBufferStatus = 0; MA_ASSERT(pDevice != NULL); /* The first thing to do is start the capture device. The playback device is only started after the first period is written. */ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { hr = ma_IDirectSoundCaptureBuffer_Start((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, MA_DSCBSTART_LOOPING); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Start() failed."); return ma_result_from_HRESULT(hr); } } while (ma_device_get_state(pDevice) == ma_device_state_started) { switch (pDevice->type) { case ma_device_type_duplex: { DWORD physicalCaptureCursorInBytes; DWORD physicalReadCursorInBytes; hr = ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes); if (FAILED(hr)) { return ma_result_from_HRESULT(hr); } /* If nothing is available we just sleep for a bit and return from this iteration. */ if (physicalReadCursorInBytes == prevReadCursorInBytesCapture) { ma_sleep(waitTimeInMilliseconds); continue; /* Nothing is available in the capture buffer. */ } /* The current position has moved. We need to map all of the captured samples and write them to the playback device, making sure we don't return until every frame has been copied over. */ if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) { /* The capture position has not looped. This is the simple case. */ lockOffsetInBytesCapture = prevReadCursorInBytesCapture; lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture); } else { /* The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything, do it again from the start. */ if (prevReadCursorInBytesCapture < pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) { /* Lock up to the end of the buffer. */ lockOffsetInBytesCapture = prevReadCursorInBytesCapture; lockSizeInBytesCapture = (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) - prevReadCursorInBytesCapture; } else { /* Lock starting from the start of the buffer. */ lockOffsetInBytesCapture = 0; lockSizeInBytesCapture = physicalReadCursorInBytes; } } if (lockSizeInBytesCapture == 0) { ma_sleep(waitTimeInMilliseconds); continue; /* Nothing is available in the capture buffer. */ } hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedDeviceBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device."); return ma_result_from_HRESULT(hr); } /* At this point we have some input data that we need to output. We do not return until every mapped frame of the input data is written to the playback device. */ mappedDeviceFramesProcessedCapture = 0; for (;;) { /* Keep writing to the playback device. */ ma_uint8 inputFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; ma_uint32 inputFramesInClientFormatCap = sizeof(inputFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); ma_uint8 outputFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; ma_uint32 outputFramesInClientFormatCap = sizeof(outputFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); ma_uint32 outputFramesInClientFormatCount; ma_uint32 outputFramesInClientFormatConsumed = 0; ma_uint64 clientCapturedFramesToProcess = ma_min(inputFramesInClientFormatCap, outputFramesInClientFormatCap); ma_uint64 deviceCapturedFramesToProcess = (mappedSizeInBytesCapture / bpfDeviceCapture) - mappedDeviceFramesProcessedCapture; void* pRunningMappedDeviceBufferCapture = ma_offset_ptr(pMappedDeviceBufferCapture, mappedDeviceFramesProcessedCapture * bpfDeviceCapture); result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningMappedDeviceBufferCapture, &deviceCapturedFramesToProcess, inputFramesInClientFormat, &clientCapturedFramesToProcess); if (result != MA_SUCCESS) { break; } outputFramesInClientFormatCount = (ma_uint32)clientCapturedFramesToProcess; mappedDeviceFramesProcessedCapture += (ma_uint32)deviceCapturedFramesToProcess; ma_device__handle_data_callback(pDevice, outputFramesInClientFormat, inputFramesInClientFormat, (ma_uint32)clientCapturedFramesToProcess); /* At this point we have input and output data in client format. All we need to do now is convert it to the output device format. This may take a few passes. */ for (;;) { ma_uint32 framesWrittenThisIteration; DWORD physicalPlayCursorInBytes; DWORD physicalWriteCursorInBytes; DWORD availableBytesPlayback; DWORD silentPaddingInBytes = 0; /* <-- Must be initialized to 0. */ /* We need the physical play and write cursors. */ if (FAILED(ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes))) { break; } if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) { physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback; } prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes; /* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */ if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */ if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) { availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback; availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */ } else { /* This is an error. */ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Duplex/Playback): Play cursor has moved in front of the write cursor (same loop iteration). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); availableBytesPlayback = 0; } } else { /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */ if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) { availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; } else { /* This is an error. */ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Duplex/Playback): Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); availableBytesPlayback = 0; } } /* If there's no room available for writing we need to wait for more. */ if (availableBytesPlayback == 0) { /* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */ if (!isPlaybackDeviceStarted) { hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING); if (FAILED(hr)) { ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed."); return ma_result_from_HRESULT(hr); } isPlaybackDeviceStarted = MA_TRUE; } else { ma_sleep(waitTimeInMilliseconds); continue; } } /* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */ lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback; if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { /* Same loop iteration. Go up to the end of the buffer. */ lockSizeInBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback; } else { /* Different loop iterations. Go up to the physical play cursor. */ lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; } hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedDeviceBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device."); result = ma_result_from_HRESULT(hr); break; } /* Experiment: If the playback buffer is being starved, pad it with some silence to get it back in sync. This will cause a glitch, but it may prevent endless glitching due to it constantly running out of data. */ if (isPlaybackDeviceStarted) { DWORD bytesQueuedForPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - availableBytesPlayback; if (bytesQueuedForPlayback < (pDevice->playback.internalPeriodSizeInFrames*bpfDevicePlayback)) { silentPaddingInBytes = (pDevice->playback.internalPeriodSizeInFrames*2*bpfDevicePlayback) - bytesQueuedForPlayback; if (silentPaddingInBytes > lockSizeInBytesPlayback) { silentPaddingInBytes = lockSizeInBytesPlayback; } ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Duplex/Playback) Playback buffer starved. availableBytesPlayback=%ld, silentPaddingInBytes=%ld\n", availableBytesPlayback, silentPaddingInBytes); } } /* At this point we have a buffer for output. */ if (silentPaddingInBytes > 0) { MA_ZERO_MEMORY(pMappedDeviceBufferPlayback, silentPaddingInBytes); framesWrittenThisIteration = silentPaddingInBytes/bpfDevicePlayback; } else { ma_uint64 convertedFrameCountIn = (outputFramesInClientFormatCount - outputFramesInClientFormatConsumed); ma_uint64 convertedFrameCountOut = mappedSizeInBytesPlayback/bpfDevicePlayback; void* pConvertedFramesIn = ma_offset_ptr(outputFramesInClientFormat, outputFramesInClientFormatConsumed * bpfDevicePlayback); void* pConvertedFramesOut = pMappedDeviceBufferPlayback; result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pConvertedFramesIn, &convertedFrameCountIn, pConvertedFramesOut, &convertedFrameCountOut); if (result != MA_SUCCESS) { break; } outputFramesInClientFormatConsumed += (ma_uint32)convertedFrameCountOut; framesWrittenThisIteration = (ma_uint32)convertedFrameCountOut; } hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedDeviceBufferPlayback, framesWrittenThisIteration*bpfDevicePlayback, NULL, 0); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device."); result = ma_result_from_HRESULT(hr); break; } virtualWriteCursorInBytesPlayback += framesWrittenThisIteration*bpfDevicePlayback; if ((virtualWriteCursorInBytesPlayback/bpfDevicePlayback) == pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods) { virtualWriteCursorInBytesPlayback = 0; virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback; } /* We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds a bit of a buffer to prevent the playback buffer from getting starved. */ framesWrittenToPlaybackDevice += framesWrittenThisIteration; if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= (pDevice->playback.internalPeriodSizeInFrames*2)) { hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING); if (FAILED(hr)) { ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed."); return ma_result_from_HRESULT(hr); } isPlaybackDeviceStarted = MA_TRUE; } if (framesWrittenThisIteration < mappedSizeInBytesPlayback/bpfDevicePlayback) { break; /* We're finished with the output data.*/ } } if (clientCapturedFramesToProcess == 0) { break; /* We just consumed every input sample. */ } } /* At this point we're done with the mapped portion of the capture buffer. */ hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedDeviceBufferCapture, mappedSizeInBytesCapture, NULL, 0); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device."); return ma_result_from_HRESULT(hr); } prevReadCursorInBytesCapture = (lockOffsetInBytesCapture + mappedSizeInBytesCapture); } break; case ma_device_type_capture: { DWORD physicalCaptureCursorInBytes; DWORD physicalReadCursorInBytes; hr = ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes); if (FAILED(hr)) { return MA_ERROR; } /* If the previous capture position is the same as the current position we need to wait a bit longer. */ if (prevReadCursorInBytesCapture == physicalReadCursorInBytes) { ma_sleep(waitTimeInMilliseconds); continue; } /* Getting here means we have capture data available. */ if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) { /* The capture position has not looped. This is the simple case. */ lockOffsetInBytesCapture = prevReadCursorInBytesCapture; lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture); } else { /* The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything, do it again from the start. */ if (prevReadCursorInBytesCapture < pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) { /* Lock up to the end of the buffer. */ lockOffsetInBytesCapture = prevReadCursorInBytesCapture; lockSizeInBytesCapture = (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) - prevReadCursorInBytesCapture; } else { /* Lock starting from the start of the buffer. */ lockOffsetInBytesCapture = 0; lockSizeInBytesCapture = physicalReadCursorInBytes; } } if (lockSizeInBytesCapture < pDevice->capture.internalPeriodSizeInFrames) { ma_sleep(waitTimeInMilliseconds); continue; /* Nothing is available in the capture buffer. */ } hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedDeviceBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device."); result = ma_result_from_HRESULT(hr); } if (lockSizeInBytesCapture != mappedSizeInBytesCapture) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[DirectSound] (Capture) lockSizeInBytesCapture=%ld != mappedSizeInBytesCapture=%ld\n", lockSizeInBytesCapture, mappedSizeInBytesCapture); } ma_device__send_frames_to_client(pDevice, mappedSizeInBytesCapture/bpfDeviceCapture, pMappedDeviceBufferCapture); hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedDeviceBufferCapture, mappedSizeInBytesCapture, NULL, 0); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device."); return ma_result_from_HRESULT(hr); } prevReadCursorInBytesCapture = lockOffsetInBytesCapture + mappedSizeInBytesCapture; if (prevReadCursorInBytesCapture == (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture)) { prevReadCursorInBytesCapture = 0; } } break; case ma_device_type_playback: { DWORD availableBytesPlayback; DWORD physicalPlayCursorInBytes; DWORD physicalWriteCursorInBytes; hr = ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes); if (FAILED(hr)) { break; } hr = ma_IDirectSoundBuffer_GetStatus((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &playbackBufferStatus); if (SUCCEEDED(hr) && (playbackBufferStatus & MA_DSBSTATUS_PLAYING) == 0 && isPlaybackDeviceStarted) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[DirectSound] Attempting to resume audio due to state: %d.", (int)playbackBufferStatus); hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING); if (FAILED(hr)) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed after attempting to resume from state %d.", (int)playbackBufferStatus); return ma_result_from_HRESULT(hr); } isPlaybackDeviceStarted = MA_TRUE; ma_sleep(waitTimeInMilliseconds); continue; } if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) { physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback; } prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes; /* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */ if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */ if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) { availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback; availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */ } else { /* This is an error. */ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Playback): Play cursor has moved in front of the write cursor (same loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); availableBytesPlayback = 0; } } else { /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */ if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) { availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; } else { /* This is an error. */ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Playback): Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); availableBytesPlayback = 0; } } /* If there's no room available for writing we need to wait for more. */ if (availableBytesPlayback < pDevice->playback.internalPeriodSizeInFrames) { /* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */ if (availableBytesPlayback == 0 && !isPlaybackDeviceStarted) { hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed."); return ma_result_from_HRESULT(hr); } isPlaybackDeviceStarted = MA_TRUE; } else { ma_sleep(waitTimeInMilliseconds); continue; } } /* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */ lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback; if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { /* Same loop iteration. Go up to the end of the buffer. */ lockSizeInBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback; } else { /* Different loop iterations. Go up to the physical play cursor. */ lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; } hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedDeviceBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device."); result = ma_result_from_HRESULT(hr); break; } /* At this point we have a buffer for output. */ ma_device__read_frames_from_client(pDevice, (mappedSizeInBytesPlayback/bpfDevicePlayback), pMappedDeviceBufferPlayback); hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedDeviceBufferPlayback, mappedSizeInBytesPlayback, NULL, 0); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device."); result = ma_result_from_HRESULT(hr); break; } virtualWriteCursorInBytesPlayback += mappedSizeInBytesPlayback; if (virtualWriteCursorInBytesPlayback == pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) { virtualWriteCursorInBytesPlayback = 0; virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback; } /* We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds a bit of a buffer to prevent the playback buffer from getting starved. */ framesWrittenToPlaybackDevice += mappedSizeInBytesPlayback/bpfDevicePlayback; if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= pDevice->playback.internalPeriodSizeInFrames) { hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed."); return ma_result_from_HRESULT(hr); } isPlaybackDeviceStarted = MA_TRUE; } } break; default: return MA_INVALID_ARGS; /* Invalid device type. */ } if (result != MA_SUCCESS) { return result; } } /* Getting here means the device is being stopped. */ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { hr = ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Stop() failed."); return ma_result_from_HRESULT(hr); } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { /* The playback device should be drained before stopping. All we do is wait until the available bytes is equal to the size of the buffer. */ if (isPlaybackDeviceStarted) { for (;;) { DWORD availableBytesPlayback = 0; DWORD physicalPlayCursorInBytes; DWORD physicalWriteCursorInBytes; hr = ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes); if (FAILED(hr)) { break; } if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) { physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback; } prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes; if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */ if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) { availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback; availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */ } else { break; } } else { /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */ if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) { availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; } else { break; } } if (availableBytesPlayback >= (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback)) { break; } ma_sleep(waitTimeInMilliseconds); } } hr = ma_IDirectSoundBuffer_Stop((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer); if (FAILED(hr)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Stop() failed."); return ma_result_from_HRESULT(hr); } ma_IDirectSoundBuffer_SetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0); } return MA_SUCCESS; } static ma_result ma_context_uninit__dsound(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_dsound); ma_dlclose(ma_context_get_log(pContext), pContext->dsound.hDSoundDLL); return MA_SUCCESS; } static ma_result ma_context_init__dsound(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { MA_ASSERT(pContext != NULL); (void)pConfig; pContext->dsound.hDSoundDLL = ma_dlopen(ma_context_get_log(pContext), "dsound.dll"); if (pContext->dsound.hDSoundDLL == NULL) { return MA_API_NOT_FOUND; } pContext->dsound.DirectSoundCreate = ma_dlsym(ma_context_get_log(pContext), pContext->dsound.hDSoundDLL, "DirectSoundCreate"); pContext->dsound.DirectSoundEnumerateA = ma_dlsym(ma_context_get_log(pContext), pContext->dsound.hDSoundDLL, "DirectSoundEnumerateA"); pContext->dsound.DirectSoundCaptureCreate = ma_dlsym(ma_context_get_log(pContext), pContext->dsound.hDSoundDLL, "DirectSoundCaptureCreate"); pContext->dsound.DirectSoundCaptureEnumerateA = ma_dlsym(ma_context_get_log(pContext), pContext->dsound.hDSoundDLL, "DirectSoundCaptureEnumerateA"); /* We need to support all functions or nothing. DirectSound with Windows 95 seems to not work too well in my testing. For example, it's missing DirectSoundCaptureEnumerateA(). This is a convenient place to just disable the DirectSound backend for Windows 95. */ if (pContext->dsound.DirectSoundCreate == NULL || pContext->dsound.DirectSoundEnumerateA == NULL || pContext->dsound.DirectSoundCaptureCreate == NULL || pContext->dsound.DirectSoundCaptureEnumerateA == NULL) { return MA_API_NOT_FOUND; } pContext->dsound.hWnd = pConfig->dsound.hWnd; pCallbacks->onContextInit = ma_context_init__dsound; pCallbacks->onContextUninit = ma_context_uninit__dsound; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__dsound; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__dsound; pCallbacks->onDeviceInit = ma_device_init__dsound; pCallbacks->onDeviceUninit = ma_device_uninit__dsound; pCallbacks->onDeviceStart = NULL; /* Not used. Started in onDeviceDataLoop. */ pCallbacks->onDeviceStop = NULL; /* Not used. Stopped in onDeviceDataLoop. */ pCallbacks->onDeviceRead = NULL; /* Not used. Data is read directly in onDeviceDataLoop. */ pCallbacks->onDeviceWrite = NULL; /* Not used. Data is written directly in onDeviceDataLoop. */ pCallbacks->onDeviceDataLoop = ma_device_data_loop__dsound; return MA_SUCCESS; } #endif /****************************************************************************** WinMM Backend ******************************************************************************/ #ifdef MA_HAS_WINMM /* Some build configurations will exclude the WinMM API. An example is when WIN32_LEAN_AND_MEAN is defined. We need to define the types and functions we need manually. */ #define MA_MMSYSERR_NOERROR 0 #define MA_MMSYSERR_ERROR 1 #define MA_MMSYSERR_BADDEVICEID 2 #define MA_MMSYSERR_INVALHANDLE 5 #define MA_MMSYSERR_NOMEM 7 #define MA_MMSYSERR_INVALFLAG 10 #define MA_MMSYSERR_INVALPARAM 11 #define MA_MMSYSERR_HANDLEBUSY 12 #define MA_CALLBACK_EVENT 0x00050000 #define MA_WAVE_ALLOWSYNC 0x0002 #define MA_WHDR_DONE 0x00000001 #define MA_WHDR_PREPARED 0x00000002 #define MA_WHDR_BEGINLOOP 0x00000004 #define MA_WHDR_ENDLOOP 0x00000008 #define MA_WHDR_INQUEUE 0x00000010 #define MA_MAXPNAMELEN 32 typedef void* MA_HWAVEIN; typedef void* MA_HWAVEOUT; typedef UINT MA_MMRESULT; typedef UINT MA_MMVERSION; typedef struct { WORD wMid; WORD wPid; MA_MMVERSION vDriverVersion; CHAR szPname[MA_MAXPNAMELEN]; DWORD dwFormats; WORD wChannels; WORD wReserved1; } MA_WAVEINCAPSA; typedef struct { WORD wMid; WORD wPid; MA_MMVERSION vDriverVersion; CHAR szPname[MA_MAXPNAMELEN]; DWORD dwFormats; WORD wChannels; WORD wReserved1; DWORD dwSupport; } MA_WAVEOUTCAPSA; typedef struct tagWAVEHDR { char* lpData; DWORD dwBufferLength; DWORD dwBytesRecorded; DWORD_PTR dwUser; DWORD dwFlags; DWORD dwLoops; struct tagWAVEHDR* lpNext; DWORD_PTR reserved; } MA_WAVEHDR; typedef struct { WORD wMid; WORD wPid; MA_MMVERSION vDriverVersion; CHAR szPname[MA_MAXPNAMELEN]; DWORD dwFormats; WORD wChannels; WORD wReserved1; DWORD dwSupport; GUID ManufacturerGuid; GUID ProductGuid; GUID NameGuid; } MA_WAVEOUTCAPS2A; typedef struct { WORD wMid; WORD wPid; MA_MMVERSION vDriverVersion; CHAR szPname[MA_MAXPNAMELEN]; DWORD dwFormats; WORD wChannels; WORD wReserved1; GUID ManufacturerGuid; GUID ProductGuid; GUID NameGuid; } MA_WAVEINCAPS2A; typedef UINT (WINAPI * MA_PFN_waveOutGetNumDevs)(void); typedef MA_MMRESULT (WINAPI * MA_PFN_waveOutGetDevCapsA)(ma_uintptr uDeviceID, MA_WAVEOUTCAPSA* pwoc, UINT cbwoc); typedef MA_MMRESULT (WINAPI * MA_PFN_waveOutOpen)(MA_HWAVEOUT* phwo, UINT uDeviceID, const MA_WAVEFORMATEX* pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen); typedef MA_MMRESULT (WINAPI * MA_PFN_waveOutClose)(MA_HWAVEOUT hwo); typedef MA_MMRESULT (WINAPI * MA_PFN_waveOutPrepareHeader)(MA_HWAVEOUT hwo, MA_WAVEHDR* pwh, UINT cbwh); typedef MA_MMRESULT (WINAPI * MA_PFN_waveOutUnprepareHeader)(MA_HWAVEOUT hwo, MA_WAVEHDR* pwh, UINT cbwh); typedef MA_MMRESULT (WINAPI * MA_PFN_waveOutWrite)(MA_HWAVEOUT hwo, MA_WAVEHDR* pwh, UINT cbwh); typedef MA_MMRESULT (WINAPI * MA_PFN_waveOutReset)(MA_HWAVEOUT hwo); typedef UINT (WINAPI * MA_PFN_waveInGetNumDevs)(void); typedef MA_MMRESULT (WINAPI * MA_PFN_waveInGetDevCapsA)(ma_uintptr uDeviceID, MA_WAVEINCAPSA* pwic, UINT cbwic); typedef MA_MMRESULT (WINAPI * MA_PFN_waveInOpen)(MA_HWAVEIN* phwi, UINT uDeviceID, const MA_WAVEFORMATEX* pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen); typedef MA_MMRESULT (WINAPI * MA_PFN_waveInClose)(MA_HWAVEIN hwi); typedef MA_MMRESULT (WINAPI * MA_PFN_waveInPrepareHeader)(MA_HWAVEIN hwi, MA_WAVEHDR* pwh, UINT cbwh); typedef MA_MMRESULT (WINAPI * MA_PFN_waveInUnprepareHeader)(MA_HWAVEIN hwi, MA_WAVEHDR* pwh, UINT cbwh); typedef MA_MMRESULT (WINAPI * MA_PFN_waveInAddBuffer)(MA_HWAVEIN hwi, MA_WAVEHDR* pwh, UINT cbwh); typedef MA_MMRESULT (WINAPI * MA_PFN_waveInStart)(MA_HWAVEIN hwi); typedef MA_MMRESULT (WINAPI * MA_PFN_waveInReset)(MA_HWAVEIN hwi); static ma_result ma_result_from_MMRESULT(MA_MMRESULT resultMM) { switch (resultMM) { case MA_MMSYSERR_NOERROR: return MA_SUCCESS; case MA_MMSYSERR_BADDEVICEID: return MA_INVALID_ARGS; case MA_MMSYSERR_INVALHANDLE: return MA_INVALID_ARGS; case MA_MMSYSERR_NOMEM: return MA_OUT_OF_MEMORY; case MA_MMSYSERR_INVALFLAG: return MA_INVALID_ARGS; case MA_MMSYSERR_INVALPARAM: return MA_INVALID_ARGS; case MA_MMSYSERR_HANDLEBUSY: return MA_BUSY; case MA_MMSYSERR_ERROR: return MA_ERROR; default: return MA_ERROR; } } static char* ma_find_last_character(char* str, char ch) { char* last; if (str == NULL) { return NULL; } last = NULL; while (*str != '\0') { if (*str == ch) { last = str; } str += 1; } return last; } static ma_uint32 ma_get_period_size_in_bytes(ma_uint32 periodSizeInFrames, ma_format format, ma_uint32 channels) { return periodSizeInFrames * ma_get_bytes_per_frame(format, channels); } /* Our own "WAVECAPS" structure that contains generic information shared between WAVEOUTCAPS2 and WAVEINCAPS2 so we can do things generically and typesafely. Names are being kept the same for consistency. */ typedef struct { CHAR szPname[MA_MAXPNAMELEN]; DWORD dwFormats; WORD wChannels; GUID NameGuid; } MA_WAVECAPSA; static ma_result ma_get_best_info_from_formats_flags__winmm(DWORD dwFormats, WORD channels, WORD* pBitsPerSample, DWORD* pSampleRate) { WORD bitsPerSample = 0; DWORD sampleRate = 0; if (pBitsPerSample) { *pBitsPerSample = 0; } if (pSampleRate) { *pSampleRate = 0; } if (channels == 1) { bitsPerSample = 16; if ((dwFormats & WAVE_FORMAT_48M16) != 0) { sampleRate = 48000; } else if ((dwFormats & WAVE_FORMAT_44M16) != 0) { sampleRate = 44100; } else if ((dwFormats & WAVE_FORMAT_2M16) != 0) { sampleRate = 22050; } else if ((dwFormats & WAVE_FORMAT_1M16) != 0) { sampleRate = 11025; } else if ((dwFormats & WAVE_FORMAT_96M16) != 0) { sampleRate = 96000; } else { bitsPerSample = 8; if ((dwFormats & WAVE_FORMAT_48M08) != 0) { sampleRate = 48000; } else if ((dwFormats & WAVE_FORMAT_44M08) != 0) { sampleRate = 44100; } else if ((dwFormats & WAVE_FORMAT_2M08) != 0) { sampleRate = 22050; } else if ((dwFormats & WAVE_FORMAT_1M08) != 0) { sampleRate = 11025; } else if ((dwFormats & WAVE_FORMAT_96M08) != 0) { sampleRate = 96000; } else { return MA_FORMAT_NOT_SUPPORTED; } } } else { bitsPerSample = 16; if ((dwFormats & WAVE_FORMAT_48S16) != 0) { sampleRate = 48000; } else if ((dwFormats & WAVE_FORMAT_44S16) != 0) { sampleRate = 44100; } else if ((dwFormats & WAVE_FORMAT_2S16) != 0) { sampleRate = 22050; } else if ((dwFormats & WAVE_FORMAT_1S16) != 0) { sampleRate = 11025; } else if ((dwFormats & WAVE_FORMAT_96S16) != 0) { sampleRate = 96000; } else { bitsPerSample = 8; if ((dwFormats & WAVE_FORMAT_48S08) != 0) { sampleRate = 48000; } else if ((dwFormats & WAVE_FORMAT_44S08) != 0) { sampleRate = 44100; } else if ((dwFormats & WAVE_FORMAT_2S08) != 0) { sampleRate = 22050; } else if ((dwFormats & WAVE_FORMAT_1S08) != 0) { sampleRate = 11025; } else if ((dwFormats & WAVE_FORMAT_96S08) != 0) { sampleRate = 96000; } else { return MA_FORMAT_NOT_SUPPORTED; } } } if (pBitsPerSample) { *pBitsPerSample = bitsPerSample; } if (pSampleRate) { *pSampleRate = sampleRate; } return MA_SUCCESS; } static ma_result ma_formats_flags_to_WAVEFORMATEX__winmm(DWORD dwFormats, WORD channels, MA_WAVEFORMATEX* pWF) { ma_result result; MA_ASSERT(pWF != NULL); MA_ZERO_OBJECT(pWF); pWF->cbSize = sizeof(*pWF); pWF->wFormatTag = WAVE_FORMAT_PCM; pWF->nChannels = (WORD)channels; if (pWF->nChannels > 2) { pWF->nChannels = 2; } result = ma_get_best_info_from_formats_flags__winmm(dwFormats, channels, &pWF->wBitsPerSample, &pWF->nSamplesPerSec); if (result != MA_SUCCESS) { return result; } pWF->nBlockAlign = (WORD)(pWF->nChannels * pWF->wBitsPerSample / 8); pWF->nAvgBytesPerSec = pWF->nBlockAlign * pWF->nSamplesPerSec; return MA_SUCCESS; } static ma_result ma_context_get_device_info_from_WAVECAPS(ma_context* pContext, MA_WAVECAPSA* pCaps, ma_device_info* pDeviceInfo) { WORD bitsPerSample; DWORD sampleRate; ma_result result; MA_ASSERT(pContext != NULL); MA_ASSERT(pCaps != NULL); MA_ASSERT(pDeviceInfo != NULL); /* Name / Description Unfortunately the name specified in WAVE(OUT/IN)CAPS2 is limited to 31 characters. This results in an unprofessional looking situation where the names of the devices are truncated. To help work around this, we need to look at the name GUID and try looking in the registry for the full name. If we can't find it there, we need to just fall back to the default name. */ /* Set the default to begin with. */ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), pCaps->szPname, (size_t)-1); /* Now try the registry. There's a few things to consider here: - The name GUID can be null, in which we case we just need to stick to the original 31 characters. - If the name GUID is not present in the registry we'll also need to stick to the original 31 characters. - I like consistency, so I want the returned device names to be consistent with those returned by WASAPI and DirectSound. The problem, however is that WASAPI and DirectSound use "<component> (<name>)" format (such as "Speakers (High Definition Audio)"), but WinMM does not specify the component name. From my admittedly limited testing, I've notice the component name seems to usually fit within the 31 characters of the fixed sized buffer, so what I'm going to do is parse that string for the component name, and then concatenate the name from the registry. */ if (!ma_is_guid_null(&pCaps->NameGuid)) { WCHAR guidStrW[256]; if (((MA_PFN_StringFromGUID2)pContext->win32.StringFromGUID2)(&pCaps->NameGuid, guidStrW, ma_countof(guidStrW)) > 0) { char guidStr[256]; char keyStr[1024]; HKEY hKey; WideCharToMultiByte(CP_UTF8, 0, guidStrW, -1, guidStr, sizeof(guidStr), 0, FALSE); ma_strcpy_s(keyStr, sizeof(keyStr), "SYSTEM\\CurrentControlSet\\Control\\MediaCategories\\"); ma_strcat_s(keyStr, sizeof(keyStr), guidStr); if (((MA_PFN_RegOpenKeyExA)pContext->win32.RegOpenKeyExA)(HKEY_LOCAL_MACHINE, keyStr, 0, KEY_READ, &hKey) == ERROR_SUCCESS) { BYTE nameFromReg[512]; DWORD nameFromRegSize = sizeof(nameFromReg); LONG resultWin32 = ((MA_PFN_RegQueryValueExA)pContext->win32.RegQueryValueExA)(hKey, "Name", 0, NULL, (BYTE*)nameFromReg, (DWORD*)&nameFromRegSize); ((MA_PFN_RegCloseKey)pContext->win32.RegCloseKey)(hKey); if (resultWin32 == ERROR_SUCCESS) { /* We have the value from the registry, so now we need to construct the name string. */ char name[1024]; if (ma_strcpy_s(name, sizeof(name), pDeviceInfo->name) == 0) { char* nameBeg = ma_find_last_character(name, '('); if (nameBeg != NULL) { size_t leadingLen = (nameBeg - name); ma_strncpy_s(nameBeg + 1, sizeof(name) - leadingLen, (const char*)nameFromReg, (size_t)-1); /* The closing ")", if it can fit. */ if (leadingLen + nameFromRegSize < sizeof(name)-1) { ma_strcat_s(name, sizeof(name), ")"); } ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), name, (size_t)-1); } } } } } } result = ma_get_best_info_from_formats_flags__winmm(pCaps->dwFormats, pCaps->wChannels, &bitsPerSample, &sampleRate); if (result != MA_SUCCESS) { return result; } if (bitsPerSample == 8) { pDeviceInfo->nativeDataFormats[0].format = ma_format_u8; } else if (bitsPerSample == 16) { pDeviceInfo->nativeDataFormats[0].format = ma_format_s16; } else if (bitsPerSample == 24) { pDeviceInfo->nativeDataFormats[0].format = ma_format_s24; } else if (bitsPerSample == 32) { pDeviceInfo->nativeDataFormats[0].format = ma_format_s32; } else { return MA_FORMAT_NOT_SUPPORTED; } pDeviceInfo->nativeDataFormats[0].channels = pCaps->wChannels; pDeviceInfo->nativeDataFormats[0].sampleRate = sampleRate; pDeviceInfo->nativeDataFormats[0].flags = 0; pDeviceInfo->nativeDataFormatCount = 1; return MA_SUCCESS; } static ma_result ma_context_get_device_info_from_WAVEOUTCAPS2(ma_context* pContext, MA_WAVEOUTCAPS2A* pCaps, ma_device_info* pDeviceInfo) { MA_WAVECAPSA caps; MA_ASSERT(pContext != NULL); MA_ASSERT(pCaps != NULL); MA_ASSERT(pDeviceInfo != NULL); MA_COPY_MEMORY(caps.szPname, pCaps->szPname, sizeof(caps.szPname)); caps.dwFormats = pCaps->dwFormats; caps.wChannels = pCaps->wChannels; caps.NameGuid = pCaps->NameGuid; return ma_context_get_device_info_from_WAVECAPS(pContext, &caps, pDeviceInfo); } static ma_result ma_context_get_device_info_from_WAVEINCAPS2(ma_context* pContext, MA_WAVEINCAPS2A* pCaps, ma_device_info* pDeviceInfo) { MA_WAVECAPSA caps; MA_ASSERT(pContext != NULL); MA_ASSERT(pCaps != NULL); MA_ASSERT(pDeviceInfo != NULL); MA_COPY_MEMORY(caps.szPname, pCaps->szPname, sizeof(caps.szPname)); caps.dwFormats = pCaps->dwFormats; caps.wChannels = pCaps->wChannels; caps.NameGuid = pCaps->NameGuid; return ma_context_get_device_info_from_WAVECAPS(pContext, &caps, pDeviceInfo); } static ma_result ma_context_enumerate_devices__winmm(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { UINT playbackDeviceCount; UINT captureDeviceCount; UINT iPlaybackDevice; UINT iCaptureDevice; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); /* Playback. */ playbackDeviceCount = ((MA_PFN_waveOutGetNumDevs)pContext->winmm.waveOutGetNumDevs)(); for (iPlaybackDevice = 0; iPlaybackDevice < playbackDeviceCount; ++iPlaybackDevice) { MA_MMRESULT result; MA_WAVEOUTCAPS2A caps; MA_ZERO_OBJECT(&caps); result = ((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(iPlaybackDevice, (MA_WAVEOUTCAPSA*)&caps, sizeof(caps)); if (result == MA_MMSYSERR_NOERROR) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); deviceInfo.id.winmm = iPlaybackDevice; /* The first enumerated device is the default device. */ if (iPlaybackDevice == 0) { deviceInfo.isDefault = MA_TRUE; } if (ma_context_get_device_info_from_WAVEOUTCAPS2(pContext, &caps, &deviceInfo) == MA_SUCCESS) { ma_bool32 cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); if (cbResult == MA_FALSE) { return MA_SUCCESS; /* Enumeration was stopped. */ } } } } /* Capture. */ captureDeviceCount = ((MA_PFN_waveInGetNumDevs)pContext->winmm.waveInGetNumDevs)(); for (iCaptureDevice = 0; iCaptureDevice < captureDeviceCount; ++iCaptureDevice) { MA_MMRESULT result; MA_WAVEINCAPS2A caps; MA_ZERO_OBJECT(&caps); result = ((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(iCaptureDevice, (MA_WAVEINCAPSA*)&caps, sizeof(caps)); if (result == MA_MMSYSERR_NOERROR) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); deviceInfo.id.winmm = iCaptureDevice; /* The first enumerated device is the default device. */ if (iCaptureDevice == 0) { deviceInfo.isDefault = MA_TRUE; } if (ma_context_get_device_info_from_WAVEINCAPS2(pContext, &caps, &deviceInfo) == MA_SUCCESS) { ma_bool32 cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); if (cbResult == MA_FALSE) { return MA_SUCCESS; /* Enumeration was stopped. */ } } } } return MA_SUCCESS; } static ma_result ma_context_get_device_info__winmm(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { UINT winMMDeviceID; MA_ASSERT(pContext != NULL); winMMDeviceID = 0; if (pDeviceID != NULL) { winMMDeviceID = (UINT)pDeviceID->winmm; } pDeviceInfo->id.winmm = winMMDeviceID; /* The first ID is the default device. */ if (winMMDeviceID == 0) { pDeviceInfo->isDefault = MA_TRUE; } if (deviceType == ma_device_type_playback) { MA_MMRESULT result; MA_WAVEOUTCAPS2A caps; MA_ZERO_OBJECT(&caps); result = ((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(winMMDeviceID, (MA_WAVEOUTCAPSA*)&caps, sizeof(caps)); if (result == MA_MMSYSERR_NOERROR) { return ma_context_get_device_info_from_WAVEOUTCAPS2(pContext, &caps, pDeviceInfo); } } else { MA_MMRESULT result; MA_WAVEINCAPS2A caps; MA_ZERO_OBJECT(&caps); result = ((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(winMMDeviceID, (MA_WAVEINCAPSA*)&caps, sizeof(caps)); if (result == MA_MMSYSERR_NOERROR) { return ma_context_get_device_info_from_WAVEINCAPS2(pContext, &caps, pDeviceInfo); } } return MA_NO_DEVICE; } static ma_result ma_device_uninit__winmm(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ((MA_PFN_waveInClose)pDevice->pContext->winmm.waveInClose)((MA_HWAVEIN)pDevice->winmm.hDeviceCapture); CloseHandle((HANDLE)pDevice->winmm.hEventCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((MA_HWAVEOUT)pDevice->winmm.hDevicePlayback); ((MA_PFN_waveOutClose)pDevice->pContext->winmm.waveOutClose)((MA_HWAVEOUT)pDevice->winmm.hDevicePlayback); CloseHandle((HANDLE)pDevice->winmm.hEventPlayback); } ma_free(pDevice->winmm._pHeapData, &pDevice->pContext->allocationCallbacks); MA_ZERO_OBJECT(&pDevice->winmm); /* Safety. */ return MA_SUCCESS; } static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__winmm(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile) { /* WinMM has a minimum period size of 40ms. */ ma_uint32 minPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(40, nativeSampleRate); ma_uint32 periodSizeInFrames; periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, nativeSampleRate, performanceProfile); if (periodSizeInFrames < minPeriodSizeInFrames) { periodSizeInFrames = minPeriodSizeInFrames; } return periodSizeInFrames; } static ma_result ma_device_init__winmm(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { const char* errorMsg = ""; ma_result errorCode = MA_ERROR; ma_result result = MA_SUCCESS; ma_uint32 heapSize; UINT winMMDeviceIDPlayback = 0; UINT winMMDeviceIDCapture = 0; MA_ASSERT(pDevice != NULL); MA_ZERO_OBJECT(&pDevice->winmm); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } /* No exclusive mode with WinMM. */ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) || ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) { return MA_SHARE_MODE_NOT_SUPPORTED; } if (pDescriptorPlayback->pDeviceID != NULL) { winMMDeviceIDPlayback = (UINT)pDescriptorPlayback->pDeviceID->winmm; } if (pDescriptorCapture->pDeviceID != NULL) { winMMDeviceIDCapture = (UINT)pDescriptorCapture->pDeviceID->winmm; } /* The capture device needs to be initialized first. */ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { MA_WAVEINCAPSA caps; MA_WAVEFORMATEX wf; MA_MMRESULT resultMM; /* We use an event to know when a new fragment needs to be enqueued. */ pDevice->winmm.hEventCapture = (ma_handle)CreateEventA(NULL, TRUE, TRUE, NULL); if (pDevice->winmm.hEventCapture == NULL) { errorMsg = "[WinMM] Failed to create event for fragment enqueuing for the capture device.", errorCode = ma_result_from_GetLastError(GetLastError()); goto on_error; } /* The format should be based on the device's actual format. */ if (((MA_PFN_waveInGetDevCapsA)pDevice->pContext->winmm.waveInGetDevCapsA)(winMMDeviceIDCapture, &caps, sizeof(caps)) != MA_MMSYSERR_NOERROR) { errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED; goto on_error; } result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf); if (result != MA_SUCCESS) { errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result; goto on_error; } resultMM = ((MA_PFN_waveInOpen)pDevice->pContext->winmm.waveInOpen)((MA_HWAVEIN*)&pDevice->winmm.hDeviceCapture, winMMDeviceIDCapture, &wf, (DWORD_PTR)pDevice->winmm.hEventCapture, (DWORD_PTR)pDevice, MA_CALLBACK_EVENT | MA_WAVE_ALLOWSYNC); if (resultMM != MA_MMSYSERR_NOERROR) { errorMsg = "[WinMM] Failed to open capture device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE; goto on_error; } pDescriptorCapture->format = ma_format_from_WAVEFORMATEX(&wf); pDescriptorCapture->channels = wf.nChannels; pDescriptorCapture->sampleRate = wf.nSamplesPerSec; ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorCapture->channels); pDescriptorCapture->periodCount = pDescriptorCapture->periodCount; pDescriptorCapture->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__winmm(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile); } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { MA_WAVEOUTCAPSA caps; MA_WAVEFORMATEX wf; MA_MMRESULT resultMM; /* We use an event to know when a new fragment needs to be enqueued. */ pDevice->winmm.hEventPlayback = (ma_handle)CreateEventA(NULL, TRUE, TRUE, NULL); if (pDevice->winmm.hEventPlayback == NULL) { errorMsg = "[WinMM] Failed to create event for fragment enqueuing for the playback device.", errorCode = ma_result_from_GetLastError(GetLastError()); goto on_error; } /* The format should be based on the device's actual format. */ if (((MA_PFN_waveOutGetDevCapsA)pDevice->pContext->winmm.waveOutGetDevCapsA)(winMMDeviceIDPlayback, &caps, sizeof(caps)) != MA_MMSYSERR_NOERROR) { errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED; goto on_error; } result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf); if (result != MA_SUCCESS) { errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result; goto on_error; } resultMM = ((MA_PFN_waveOutOpen)pDevice->pContext->winmm.waveOutOpen)((MA_HWAVEOUT*)&pDevice->winmm.hDevicePlayback, winMMDeviceIDPlayback, &wf, (DWORD_PTR)pDevice->winmm.hEventPlayback, (DWORD_PTR)pDevice, MA_CALLBACK_EVENT | MA_WAVE_ALLOWSYNC); if (resultMM != MA_MMSYSERR_NOERROR) { errorMsg = "[WinMM] Failed to open playback device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE; goto on_error; } pDescriptorPlayback->format = ma_format_from_WAVEFORMATEX(&wf); pDescriptorPlayback->channels = wf.nChannels; pDescriptorPlayback->sampleRate = wf.nSamplesPerSec; ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pDescriptorPlayback->channelMap, ma_countof(pDescriptorPlayback->channelMap), pDescriptorPlayback->channels); pDescriptorPlayback->periodCount = pDescriptorPlayback->periodCount; pDescriptorPlayback->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__winmm(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile); } /* The heap allocated data is allocated like so: [Capture WAVEHDRs][Playback WAVEHDRs][Capture Intermediary Buffer][Playback Intermediary Buffer] */ heapSize = 0; if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { heapSize += sizeof(MA_WAVEHDR)*pDescriptorCapture->periodCount + (pDescriptorCapture->periodSizeInFrames * pDescriptorCapture->periodCount * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels)); } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { heapSize += sizeof(MA_WAVEHDR)*pDescriptorPlayback->periodCount + (pDescriptorPlayback->periodSizeInFrames * pDescriptorPlayback->periodCount * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels)); } pDevice->winmm._pHeapData = (ma_uint8*)ma_calloc(heapSize, &pDevice->pContext->allocationCallbacks); if (pDevice->winmm._pHeapData == NULL) { errorMsg = "[WinMM] Failed to allocate memory for the intermediary buffer.", errorCode = MA_OUT_OF_MEMORY; goto on_error; } MA_ZERO_MEMORY(pDevice->winmm._pHeapData, heapSize); if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { ma_uint32 iPeriod; if (pConfig->deviceType == ma_device_type_capture) { pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData; pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(MA_WAVEHDR)*(pDescriptorCapture->periodCount)); } else { pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData; pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(MA_WAVEHDR)*(pDescriptorCapture->periodCount + pDescriptorPlayback->periodCount)); } /* Prepare headers. */ for (iPeriod = 0; iPeriod < pDescriptorCapture->periodCount; ++iPeriod) { ma_uint32 periodSizeInBytes = ma_get_period_size_in_bytes(pDescriptorCapture->periodSizeInFrames, pDescriptorCapture->format, pDescriptorCapture->channels); ((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].lpData = (char*)(pDevice->winmm.pIntermediaryBufferCapture + (periodSizeInBytes*iPeriod)); ((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwBufferLength = periodSizeInBytes; ((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwFlags = 0L; ((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwLoops = 0L; ((MA_PFN_waveInPrepareHeader)pDevice->pContext->winmm.waveInPrepareHeader)((MA_HWAVEIN)pDevice->winmm.hDeviceCapture, &((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(MA_WAVEHDR)); /* The user data of the MA_WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means it's unlocked and available for writing. A value of 1 means it's locked. */ ((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwUser = 0; } } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { ma_uint32 iPeriod; if (pConfig->deviceType == ma_device_type_playback) { pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData; pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(MA_WAVEHDR)*pDescriptorPlayback->periodCount); } else { pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData + (sizeof(MA_WAVEHDR)*(pDescriptorCapture->periodCount)); pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(MA_WAVEHDR)*(pDescriptorCapture->periodCount + pDescriptorPlayback->periodCount)) + (pDescriptorCapture->periodSizeInFrames*pDescriptorCapture->periodCount*ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels)); } /* Prepare headers. */ for (iPeriod = 0; iPeriod < pDescriptorPlayback->periodCount; ++iPeriod) { ma_uint32 periodSizeInBytes = ma_get_period_size_in_bytes(pDescriptorPlayback->periodSizeInFrames, pDescriptorPlayback->format, pDescriptorPlayback->channels); ((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].lpData = (char*)(pDevice->winmm.pIntermediaryBufferPlayback + (periodSizeInBytes*iPeriod)); ((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwBufferLength = periodSizeInBytes; ((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwFlags = 0L; ((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwLoops = 0L; ((MA_PFN_waveOutPrepareHeader)pDevice->pContext->winmm.waveOutPrepareHeader)((MA_HWAVEOUT)pDevice->winmm.hDevicePlayback, &((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(MA_WAVEHDR)); /* The user data of the MA_WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means it's unlocked and available for writing. A value of 1 means it's locked. */ ((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwUser = 0; } } return MA_SUCCESS; on_error: if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { if (pDevice->winmm.pWAVEHDRCapture != NULL) { ma_uint32 iPeriod; for (iPeriod = 0; iPeriod < pDescriptorCapture->periodCount; ++iPeriod) { ((MA_PFN_waveInUnprepareHeader)pDevice->pContext->winmm.waveInUnprepareHeader)((MA_HWAVEIN)pDevice->winmm.hDeviceCapture, &((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(MA_WAVEHDR)); } } ((MA_PFN_waveInClose)pDevice->pContext->winmm.waveInClose)((MA_HWAVEIN)pDevice->winmm.hDeviceCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { if (pDevice->winmm.pWAVEHDRCapture != NULL) { ma_uint32 iPeriod; for (iPeriod = 0; iPeriod < pDescriptorPlayback->periodCount; ++iPeriod) { ((MA_PFN_waveOutUnprepareHeader)pDevice->pContext->winmm.waveOutUnprepareHeader)((MA_HWAVEOUT)pDevice->winmm.hDevicePlayback, &((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(MA_WAVEHDR)); } } ((MA_PFN_waveOutClose)pDevice->pContext->winmm.waveOutClose)((MA_HWAVEOUT)pDevice->winmm.hDevicePlayback); } ma_free(pDevice->winmm._pHeapData, &pDevice->pContext->allocationCallbacks); if (errorMsg != NULL && errorMsg[0] != '\0') { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "%s", errorMsg); } return errorCode; } static ma_result ma_device_start__winmm(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { MA_MMRESULT resultMM; MA_WAVEHDR* pWAVEHDR; ma_uint32 iPeriod; pWAVEHDR = (MA_WAVEHDR*)pDevice->winmm.pWAVEHDRCapture; /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */ ResetEvent((HANDLE)pDevice->winmm.hEventCapture); /* To start the device we attach all of the buffers and then start it. As the buffers are filled with data we will get notifications. */ for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) { resultMM = ((MA_PFN_waveInAddBuffer)pDevice->pContext->winmm.waveInAddBuffer)((MA_HWAVEIN)pDevice->winmm.hDeviceCapture, &((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(MA_WAVEHDR)); if (resultMM != MA_MMSYSERR_NOERROR) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WinMM] Failed to attach input buffers to capture device in preparation for capture."); return ma_result_from_MMRESULT(resultMM); } /* Make sure all of the buffers start out locked. We don't want to access them until the backend tells us we can. */ pWAVEHDR[iPeriod].dwUser = 1; /* 1 = locked. */ } /* Capture devices need to be explicitly started, unlike playback devices. */ resultMM = ((MA_PFN_waveInStart)pDevice->pContext->winmm.waveInStart)((MA_HWAVEIN)pDevice->winmm.hDeviceCapture); if (resultMM != MA_MMSYSERR_NOERROR) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WinMM] Failed to start backend device."); return ma_result_from_MMRESULT(resultMM); } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { /* Don't need to do anything for playback. It'll be started automatically in ma_device_start__winmm(). */ } return MA_SUCCESS; } static ma_result ma_device_stop__winmm(ma_device* pDevice) { MA_MMRESULT resultMM; MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { if (pDevice->winmm.hDeviceCapture == NULL) { return MA_INVALID_ARGS; } resultMM = ((MA_PFN_waveInReset)pDevice->pContext->winmm.waveInReset)((MA_HWAVEIN)pDevice->winmm.hDeviceCapture); if (resultMM != MA_MMSYSERR_NOERROR) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[WinMM] WARNING: Failed to reset capture device."); } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ma_uint32 iPeriod; MA_WAVEHDR* pWAVEHDR; if (pDevice->winmm.hDevicePlayback == NULL) { return MA_INVALID_ARGS; } /* We need to drain the device. To do this we just loop over each header and if it's locked just wait for the event. */ pWAVEHDR = (MA_WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback; for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; iPeriod += 1) { if (pWAVEHDR[iPeriod].dwUser == 1) { /* 1 = locked. */ if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventPlayback, INFINITE) != WAIT_OBJECT_0) { break; /* An error occurred so just abandon ship and stop the device without draining. */ } pWAVEHDR[iPeriod].dwUser = 0; } } resultMM = ((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((MA_HWAVEOUT)pDevice->winmm.hDevicePlayback); if (resultMM != MA_MMSYSERR_NOERROR) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[WinMM] WARNING: Failed to reset playback device."); } } return MA_SUCCESS; } static ma_result ma_device_write__winmm(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { ma_result result = MA_SUCCESS; MA_MMRESULT resultMM; ma_uint32 totalFramesWritten; MA_WAVEHDR* pWAVEHDR; MA_ASSERT(pDevice != NULL); MA_ASSERT(pPCMFrames != NULL); if (pFramesWritten != NULL) { *pFramesWritten = 0; } pWAVEHDR = (MA_WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback; /* Keep processing as much data as possible. */ totalFramesWritten = 0; while (totalFramesWritten < frameCount) { /* If the current header has some space available we need to write part of it. */ if (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser == 0) { /* 0 = unlocked. */ /* This header has room in it. We copy as much of it as we can. If we end up fully consuming the buffer we need to write it out and move on to the next iteration. */ ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedPlayback; ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesWritten)); const void* pSrc = ma_offset_ptr(pPCMFrames, totalFramesWritten*bpf); void* pDst = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].lpData, pDevice->winmm.headerFramesConsumedPlayback*bpf); MA_COPY_MEMORY(pDst, pSrc, framesToCopy*bpf); pDevice->winmm.headerFramesConsumedPlayback += framesToCopy; totalFramesWritten += framesToCopy; /* If we've consumed the buffer entirely we need to write it out to the device. */ if (pDevice->winmm.headerFramesConsumedPlayback == (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf)) { pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 1; /* 1 = locked. */ pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags &= ~MA_WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */ /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */ ResetEvent((HANDLE)pDevice->winmm.hEventPlayback); /* The device will be started here. */ resultMM = ((MA_PFN_waveOutWrite)pDevice->pContext->winmm.waveOutWrite)((MA_HWAVEOUT)pDevice->winmm.hDevicePlayback, &pWAVEHDR[pDevice->winmm.iNextHeaderPlayback], sizeof(MA_WAVEHDR)); if (resultMM != MA_MMSYSERR_NOERROR) { result = ma_result_from_MMRESULT(resultMM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WinMM] waveOutWrite() failed."); break; } /* Make sure we move to the next header. */ pDevice->winmm.iNextHeaderPlayback = (pDevice->winmm.iNextHeaderPlayback + 1) % pDevice->playback.internalPeriods; pDevice->winmm.headerFramesConsumedPlayback = 0; } /* If at this point we have consumed the entire input buffer we can return. */ MA_ASSERT(totalFramesWritten <= frameCount); if (totalFramesWritten == frameCount) { break; } /* Getting here means there's more to process. */ continue; } /* Getting here means there isn't enough room in the buffer and we need to wait for one to become available. */ if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventPlayback, INFINITE) != WAIT_OBJECT_0) { result = MA_ERROR; break; } /* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */ if ((pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags & MA_WHDR_DONE) != 0) { pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 0; /* 0 = unlocked (make it available for writing). */ pDevice->winmm.headerFramesConsumedPlayback = 0; } /* If the device has been stopped we need to break. */ if (ma_device_get_state(pDevice) != ma_device_state_started) { break; } } if (pFramesWritten != NULL) { *pFramesWritten = totalFramesWritten; } return result; } static ma_result ma_device_read__winmm(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) { ma_result result = MA_SUCCESS; MA_MMRESULT resultMM; ma_uint32 totalFramesRead; MA_WAVEHDR* pWAVEHDR; MA_ASSERT(pDevice != NULL); MA_ASSERT(pPCMFrames != NULL); if (pFramesRead != NULL) { *pFramesRead = 0; } pWAVEHDR = (MA_WAVEHDR*)pDevice->winmm.pWAVEHDRCapture; /* Keep processing as much data as possible. */ totalFramesRead = 0; while (totalFramesRead < frameCount) { /* If the current header has some space available we need to write part of it. */ if (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser == 0) { /* 0 = unlocked. */ /* The buffer is available for reading. If we fully consume it we need to add it back to the buffer. */ ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedCapture; ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesRead)); const void* pSrc = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderCapture].lpData, pDevice->winmm.headerFramesConsumedCapture*bpf); void* pDst = ma_offset_ptr(pPCMFrames, totalFramesRead*bpf); MA_COPY_MEMORY(pDst, pSrc, framesToCopy*bpf); pDevice->winmm.headerFramesConsumedCapture += framesToCopy; totalFramesRead += framesToCopy; /* If we've consumed the buffer entirely we need to add it back to the device. */ if (pDevice->winmm.headerFramesConsumedCapture == (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf)) { pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 1; /* 1 = locked. */ pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags &= ~MA_WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */ /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */ ResetEvent((HANDLE)pDevice->winmm.hEventCapture); /* The device will be started here. */ resultMM = ((MA_PFN_waveInAddBuffer)pDevice->pContext->winmm.waveInAddBuffer)((MA_HWAVEIN)pDevice->winmm.hDeviceCapture, &((MA_WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[pDevice->winmm.iNextHeaderCapture], sizeof(MA_WAVEHDR)); if (resultMM != MA_MMSYSERR_NOERROR) { result = ma_result_from_MMRESULT(resultMM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WinMM] waveInAddBuffer() failed."); break; } /* Make sure we move to the next header. */ pDevice->winmm.iNextHeaderCapture = (pDevice->winmm.iNextHeaderCapture + 1) % pDevice->capture.internalPeriods; pDevice->winmm.headerFramesConsumedCapture = 0; } /* If at this point we have filled the entire input buffer we can return. */ MA_ASSERT(totalFramesRead <= frameCount); if (totalFramesRead == frameCount) { break; } /* Getting here means there's more to process. */ continue; } /* Getting here means there isn't enough any data left to send to the client which means we need to wait for more. */ if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventCapture, INFINITE) != WAIT_OBJECT_0) { result = MA_ERROR; break; } /* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */ if ((pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags & MA_WHDR_DONE) != 0) { pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 0; /* 0 = unlocked (make it available for reading). */ pDevice->winmm.headerFramesConsumedCapture = 0; } /* If the device has been stopped we need to break. */ if (ma_device_get_state(pDevice) != ma_device_state_started) { break; } } if (pFramesRead != NULL) { *pFramesRead = totalFramesRead; } return result; } static ma_result ma_context_uninit__winmm(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_winmm); ma_dlclose(ma_context_get_log(pContext), pContext->winmm.hWinMM); return MA_SUCCESS; } static ma_result ma_context_init__winmm(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { MA_ASSERT(pContext != NULL); (void)pConfig; pContext->winmm.hWinMM = ma_dlopen(ma_context_get_log(pContext), "winmm.dll"); if (pContext->winmm.hWinMM == NULL) { return MA_NO_BACKEND; } pContext->winmm.waveOutGetNumDevs = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveOutGetNumDevs"); pContext->winmm.waveOutGetDevCapsA = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveOutGetDevCapsA"); pContext->winmm.waveOutOpen = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveOutOpen"); pContext->winmm.waveOutClose = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveOutClose"); pContext->winmm.waveOutPrepareHeader = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveOutPrepareHeader"); pContext->winmm.waveOutUnprepareHeader = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveOutUnprepareHeader"); pContext->winmm.waveOutWrite = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveOutWrite"); pContext->winmm.waveOutReset = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveOutReset"); pContext->winmm.waveInGetNumDevs = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveInGetNumDevs"); pContext->winmm.waveInGetDevCapsA = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveInGetDevCapsA"); pContext->winmm.waveInOpen = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveInOpen"); pContext->winmm.waveInClose = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveInClose"); pContext->winmm.waveInPrepareHeader = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveInPrepareHeader"); pContext->winmm.waveInUnprepareHeader = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveInUnprepareHeader"); pContext->winmm.waveInAddBuffer = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveInAddBuffer"); pContext->winmm.waveInStart = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveInStart"); pContext->winmm.waveInReset = ma_dlsym(ma_context_get_log(pContext), pContext->winmm.hWinMM, "waveInReset"); pCallbacks->onContextInit = ma_context_init__winmm; pCallbacks->onContextUninit = ma_context_uninit__winmm; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__winmm; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__winmm; pCallbacks->onDeviceInit = ma_device_init__winmm; pCallbacks->onDeviceUninit = ma_device_uninit__winmm; pCallbacks->onDeviceStart = ma_device_start__winmm; pCallbacks->onDeviceStop = ma_device_stop__winmm; pCallbacks->onDeviceRead = ma_device_read__winmm; pCallbacks->onDeviceWrite = ma_device_write__winmm; pCallbacks->onDeviceDataLoop = NULL; /* This is a blocking read-write API, so this can be NULL since miniaudio will manage the audio thread for us. */ return MA_SUCCESS; } #endif /****************************************************************************** ALSA Backend ******************************************************************************/ #ifdef MA_HAS_ALSA #include <poll.h> /* poll(), struct pollfd */ #include <sys/eventfd.h> /* eventfd() */ #ifdef MA_NO_RUNTIME_LINKING /* asoundlib.h marks some functions with "inline" which isn't always supported. Need to emulate it. */ #if !defined(__cplusplus) #if defined(__STRICT_ANSI__) #if !defined(inline) #define inline __inline__ __attribute__((always_inline)) #define MA_INLINE_DEFINED #endif #endif #endif #include <alsa/asoundlib.h> #if defined(MA_INLINE_DEFINED) #undef inline #undef MA_INLINE_DEFINED #endif typedef snd_pcm_uframes_t ma_snd_pcm_uframes_t; typedef snd_pcm_sframes_t ma_snd_pcm_sframes_t; typedef snd_pcm_stream_t ma_snd_pcm_stream_t; typedef snd_pcm_format_t ma_snd_pcm_format_t; typedef snd_pcm_access_t ma_snd_pcm_access_t; typedef snd_pcm_t ma_snd_pcm_t; typedef snd_pcm_hw_params_t ma_snd_pcm_hw_params_t; typedef snd_pcm_sw_params_t ma_snd_pcm_sw_params_t; typedef snd_pcm_format_mask_t ma_snd_pcm_format_mask_t; typedef snd_pcm_info_t ma_snd_pcm_info_t; typedef snd_pcm_channel_area_t ma_snd_pcm_channel_area_t; typedef snd_pcm_chmap_t ma_snd_pcm_chmap_t; typedef snd_pcm_state_t ma_snd_pcm_state_t; /* snd_pcm_state_t */ #define MA_SND_PCM_STATE_XRUN SND_PCM_STATE_XRUN /* snd_pcm_stream_t */ #define MA_SND_PCM_STREAM_PLAYBACK SND_PCM_STREAM_PLAYBACK #define MA_SND_PCM_STREAM_CAPTURE SND_PCM_STREAM_CAPTURE /* snd_pcm_format_t */ #define MA_SND_PCM_FORMAT_UNKNOWN SND_PCM_FORMAT_UNKNOWN #define MA_SND_PCM_FORMAT_U8 SND_PCM_FORMAT_U8 #define MA_SND_PCM_FORMAT_S16_LE SND_PCM_FORMAT_S16_LE #define MA_SND_PCM_FORMAT_S16_BE SND_PCM_FORMAT_S16_BE #define MA_SND_PCM_FORMAT_S24_LE SND_PCM_FORMAT_S24_LE #define MA_SND_PCM_FORMAT_S24_BE SND_PCM_FORMAT_S24_BE #define MA_SND_PCM_FORMAT_S32_LE SND_PCM_FORMAT_S32_LE #define MA_SND_PCM_FORMAT_S32_BE SND_PCM_FORMAT_S32_BE #define MA_SND_PCM_FORMAT_FLOAT_LE SND_PCM_FORMAT_FLOAT_LE #define MA_SND_PCM_FORMAT_FLOAT_BE SND_PCM_FORMAT_FLOAT_BE #define MA_SND_PCM_FORMAT_FLOAT64_LE SND_PCM_FORMAT_FLOAT64_LE #define MA_SND_PCM_FORMAT_FLOAT64_BE SND_PCM_FORMAT_FLOAT64_BE #define MA_SND_PCM_FORMAT_MU_LAW SND_PCM_FORMAT_MU_LAW #define MA_SND_PCM_FORMAT_A_LAW SND_PCM_FORMAT_A_LAW #define MA_SND_PCM_FORMAT_S24_3LE SND_PCM_FORMAT_S24_3LE #define MA_SND_PCM_FORMAT_S24_3BE SND_PCM_FORMAT_S24_3BE /* ma_snd_pcm_access_t */ #define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED SND_PCM_ACCESS_MMAP_INTERLEAVED #define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED SND_PCM_ACCESS_MMAP_NONINTERLEAVED #define MA_SND_PCM_ACCESS_MMAP_COMPLEX SND_PCM_ACCESS_MMAP_COMPLEX #define MA_SND_PCM_ACCESS_RW_INTERLEAVED SND_PCM_ACCESS_RW_INTERLEAVED #define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED SND_PCM_ACCESS_RW_NONINTERLEAVED /* Channel positions. */ #define MA_SND_CHMAP_UNKNOWN SND_CHMAP_UNKNOWN #define MA_SND_CHMAP_NA SND_CHMAP_NA #define MA_SND_CHMAP_MONO SND_CHMAP_MONO #define MA_SND_CHMAP_FL SND_CHMAP_FL #define MA_SND_CHMAP_FR SND_CHMAP_FR #define MA_SND_CHMAP_RL SND_CHMAP_RL #define MA_SND_CHMAP_RR SND_CHMAP_RR #define MA_SND_CHMAP_FC SND_CHMAP_FC #define MA_SND_CHMAP_LFE SND_CHMAP_LFE #define MA_SND_CHMAP_SL SND_CHMAP_SL #define MA_SND_CHMAP_SR SND_CHMAP_SR #define MA_SND_CHMAP_RC SND_CHMAP_RC #define MA_SND_CHMAP_FLC SND_CHMAP_FLC #define MA_SND_CHMAP_FRC SND_CHMAP_FRC #define MA_SND_CHMAP_RLC SND_CHMAP_RLC #define MA_SND_CHMAP_RRC SND_CHMAP_RRC #define MA_SND_CHMAP_FLW SND_CHMAP_FLW #define MA_SND_CHMAP_FRW SND_CHMAP_FRW #define MA_SND_CHMAP_FLH SND_CHMAP_FLH #define MA_SND_CHMAP_FCH SND_CHMAP_FCH #define MA_SND_CHMAP_FRH SND_CHMAP_FRH #define MA_SND_CHMAP_TC SND_CHMAP_TC #define MA_SND_CHMAP_TFL SND_CHMAP_TFL #define MA_SND_CHMAP_TFR SND_CHMAP_TFR #define MA_SND_CHMAP_TFC SND_CHMAP_TFC #define MA_SND_CHMAP_TRL SND_CHMAP_TRL #define MA_SND_CHMAP_TRR SND_CHMAP_TRR #define MA_SND_CHMAP_TRC SND_CHMAP_TRC #define MA_SND_CHMAP_TFLC SND_CHMAP_TFLC #define MA_SND_CHMAP_TFRC SND_CHMAP_TFRC #define MA_SND_CHMAP_TSL SND_CHMAP_TSL #define MA_SND_CHMAP_TSR SND_CHMAP_TSR #define MA_SND_CHMAP_LLFE SND_CHMAP_LLFE #define MA_SND_CHMAP_RLFE SND_CHMAP_RLFE #define MA_SND_CHMAP_BC SND_CHMAP_BC #define MA_SND_CHMAP_BLC SND_CHMAP_BLC #define MA_SND_CHMAP_BRC SND_CHMAP_BRC /* Open mode flags. */ #define MA_SND_PCM_NO_AUTO_RESAMPLE SND_PCM_NO_AUTO_RESAMPLE #define MA_SND_PCM_NO_AUTO_CHANNELS SND_PCM_NO_AUTO_CHANNELS #define MA_SND_PCM_NO_AUTO_FORMAT SND_PCM_NO_AUTO_FORMAT #else #include <errno.h> /* For EPIPE, etc. */ typedef unsigned long ma_snd_pcm_uframes_t; typedef long ma_snd_pcm_sframes_t; typedef int ma_snd_pcm_stream_t; typedef int ma_snd_pcm_format_t; typedef int ma_snd_pcm_access_t; typedef int ma_snd_pcm_state_t; typedef struct ma_snd_pcm_t ma_snd_pcm_t; typedef struct ma_snd_pcm_hw_params_t ma_snd_pcm_hw_params_t; typedef struct ma_snd_pcm_sw_params_t ma_snd_pcm_sw_params_t; typedef struct ma_snd_pcm_format_mask_t ma_snd_pcm_format_mask_t; typedef struct ma_snd_pcm_info_t ma_snd_pcm_info_t; typedef struct { void* addr; unsigned int first; unsigned int step; } ma_snd_pcm_channel_area_t; typedef struct { unsigned int channels; unsigned int pos[1]; } ma_snd_pcm_chmap_t; /* snd_pcm_state_t */ #define MA_SND_PCM_STATE_OPEN 0 #define MA_SND_PCM_STATE_SETUP 1 #define MA_SND_PCM_STATE_PREPARED 2 #define MA_SND_PCM_STATE_RUNNING 3 #define MA_SND_PCM_STATE_XRUN 4 #define MA_SND_PCM_STATE_DRAINING 5 #define MA_SND_PCM_STATE_PAUSED 6 #define MA_SND_PCM_STATE_SUSPENDED 7 #define MA_SND_PCM_STATE_DISCONNECTED 8 /* snd_pcm_stream_t */ #define MA_SND_PCM_STREAM_PLAYBACK 0 #define MA_SND_PCM_STREAM_CAPTURE 1 /* snd_pcm_format_t */ #define MA_SND_PCM_FORMAT_UNKNOWN -1 #define MA_SND_PCM_FORMAT_U8 1 #define MA_SND_PCM_FORMAT_S16_LE 2 #define MA_SND_PCM_FORMAT_S16_BE 3 #define MA_SND_PCM_FORMAT_S24_LE 6 #define MA_SND_PCM_FORMAT_S24_BE 7 #define MA_SND_PCM_FORMAT_S32_LE 10 #define MA_SND_PCM_FORMAT_S32_BE 11 #define MA_SND_PCM_FORMAT_FLOAT_LE 14 #define MA_SND_PCM_FORMAT_FLOAT_BE 15 #define MA_SND_PCM_FORMAT_FLOAT64_LE 16 #define MA_SND_PCM_FORMAT_FLOAT64_BE 17 #define MA_SND_PCM_FORMAT_MU_LAW 20 #define MA_SND_PCM_FORMAT_A_LAW 21 #define MA_SND_PCM_FORMAT_S24_3LE 32 #define MA_SND_PCM_FORMAT_S24_3BE 33 /* snd_pcm_access_t */ #define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED 0 #define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED 1 #define MA_SND_PCM_ACCESS_MMAP_COMPLEX 2 #define MA_SND_PCM_ACCESS_RW_INTERLEAVED 3 #define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED 4 /* Channel positions. */ #define MA_SND_CHMAP_UNKNOWN 0 #define MA_SND_CHMAP_NA 1 #define MA_SND_CHMAP_MONO 2 #define MA_SND_CHMAP_FL 3 #define MA_SND_CHMAP_FR 4 #define MA_SND_CHMAP_RL 5 #define MA_SND_CHMAP_RR 6 #define MA_SND_CHMAP_FC 7 #define MA_SND_CHMAP_LFE 8 #define MA_SND_CHMAP_SL 9 #define MA_SND_CHMAP_SR 10 #define MA_SND_CHMAP_RC 11 #define MA_SND_CHMAP_FLC 12 #define MA_SND_CHMAP_FRC 13 #define MA_SND_CHMAP_RLC 14 #define MA_SND_CHMAP_RRC 15 #define MA_SND_CHMAP_FLW 16 #define MA_SND_CHMAP_FRW 17 #define MA_SND_CHMAP_FLH 18 #define MA_SND_CHMAP_FCH 19 #define MA_SND_CHMAP_FRH 20 #define MA_SND_CHMAP_TC 21 #define MA_SND_CHMAP_TFL 22 #define MA_SND_CHMAP_TFR 23 #define MA_SND_CHMAP_TFC 24 #define MA_SND_CHMAP_TRL 25 #define MA_SND_CHMAP_TRR 26 #define MA_SND_CHMAP_TRC 27 #define MA_SND_CHMAP_TFLC 28 #define MA_SND_CHMAP_TFRC 29 #define MA_SND_CHMAP_TSL 30 #define MA_SND_CHMAP_TSR 31 #define MA_SND_CHMAP_LLFE 32 #define MA_SND_CHMAP_RLFE 33 #define MA_SND_CHMAP_BC 34 #define MA_SND_CHMAP_BLC 35 #define MA_SND_CHMAP_BRC 36 /* Open mode flags. */ #define MA_SND_PCM_NO_AUTO_RESAMPLE 0x00010000 #define MA_SND_PCM_NO_AUTO_CHANNELS 0x00020000 #define MA_SND_PCM_NO_AUTO_FORMAT 0x00040000 #endif typedef int (* ma_snd_pcm_open_proc) (ma_snd_pcm_t **pcm, const char *name, ma_snd_pcm_stream_t stream, int mode); typedef int (* ma_snd_pcm_close_proc) (ma_snd_pcm_t *pcm); typedef size_t (* ma_snd_pcm_hw_params_sizeof_proc) (void); typedef int (* ma_snd_pcm_hw_params_any_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params); typedef int (* ma_snd_pcm_hw_params_set_format_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t val); typedef int (* ma_snd_pcm_hw_params_set_format_first_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format); typedef void (* ma_snd_pcm_hw_params_get_format_mask_proc) (ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_mask_t *mask); typedef int (* ma_snd_pcm_hw_params_set_channels_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val); typedef int (* ma_snd_pcm_hw_params_set_channels_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val); typedef int (* ma_snd_pcm_hw_params_set_channels_minmax_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *minimum, unsigned int *maximum); typedef int (* ma_snd_pcm_hw_params_set_rate_resample_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val); typedef int (* ma_snd_pcm_hw_params_set_rate_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val, int dir); typedef int (* ma_snd_pcm_hw_params_set_rate_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir); typedef int (* ma_snd_pcm_hw_params_set_rate_minmax_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *min, int *mindir, unsigned int *max, int *maxdir); typedef int (* ma_snd_pcm_hw_params_set_buffer_size_near_proc)(ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val); typedef int (* ma_snd_pcm_hw_params_set_periods_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir); typedef int (* ma_snd_pcm_hw_params_set_access_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t _access); typedef int (* ma_snd_pcm_hw_params_get_format_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format); typedef int (* ma_snd_pcm_hw_params_get_channels_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val); typedef int (* ma_snd_pcm_hw_params_get_channels_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val); typedef int (* ma_snd_pcm_hw_params_get_channels_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val); typedef int (* ma_snd_pcm_hw_params_get_rate_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir); typedef int (* ma_snd_pcm_hw_params_get_rate_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir); typedef int (* ma_snd_pcm_hw_params_get_rate_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir); typedef int (* ma_snd_pcm_hw_params_get_buffer_size_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val); typedef int (* ma_snd_pcm_hw_params_get_periods_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir); typedef int (* ma_snd_pcm_hw_params_get_access_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t *_access); typedef int (* ma_snd_pcm_hw_params_test_format_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t val); typedef int (* ma_snd_pcm_hw_params_test_channels_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val); typedef int (* ma_snd_pcm_hw_params_test_rate_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val, int dir); typedef int (* ma_snd_pcm_hw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params); typedef size_t (* ma_snd_pcm_sw_params_sizeof_proc) (void); typedef int (* ma_snd_pcm_sw_params_current_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params); typedef int (* ma_snd_pcm_sw_params_get_boundary_proc) (const ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t* val); typedef int (* ma_snd_pcm_sw_params_set_avail_min_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val); typedef int (* ma_snd_pcm_sw_params_set_start_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val); typedef int (* ma_snd_pcm_sw_params_set_stop_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val); typedef int (* ma_snd_pcm_sw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params); typedef size_t (* ma_snd_pcm_format_mask_sizeof_proc) (void); typedef int (* ma_snd_pcm_format_mask_test_proc) (const ma_snd_pcm_format_mask_t *mask, ma_snd_pcm_format_t val); typedef ma_snd_pcm_chmap_t * (* ma_snd_pcm_get_chmap_proc) (ma_snd_pcm_t *pcm); typedef ma_snd_pcm_state_t (* ma_snd_pcm_state_proc) (ma_snd_pcm_t *pcm); typedef int (* ma_snd_pcm_prepare_proc) (ma_snd_pcm_t *pcm); typedef int (* ma_snd_pcm_start_proc) (ma_snd_pcm_t *pcm); typedef int (* ma_snd_pcm_drop_proc) (ma_snd_pcm_t *pcm); typedef int (* ma_snd_pcm_drain_proc) (ma_snd_pcm_t *pcm); typedef int (* ma_snd_pcm_reset_proc) (ma_snd_pcm_t *pcm); typedef int (* ma_snd_device_name_hint_proc) (int card, const char *iface, void ***hints); typedef char * (* ma_snd_device_name_get_hint_proc) (const void *hint, const char *id); typedef int (* ma_snd_card_get_index_proc) (const char *name); typedef int (* ma_snd_device_name_free_hint_proc) (void **hints); typedef int (* ma_snd_pcm_mmap_begin_proc) (ma_snd_pcm_t *pcm, const ma_snd_pcm_channel_area_t **areas, ma_snd_pcm_uframes_t *offset, ma_snd_pcm_uframes_t *frames); typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_mmap_commit_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_uframes_t offset, ma_snd_pcm_uframes_t frames); typedef int (* ma_snd_pcm_recover_proc) (ma_snd_pcm_t *pcm, int err, int silent); typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_readi_proc) (ma_snd_pcm_t *pcm, void *buffer, ma_snd_pcm_uframes_t size); typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_writei_proc) (ma_snd_pcm_t *pcm, const void *buffer, ma_snd_pcm_uframes_t size); typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_proc) (ma_snd_pcm_t *pcm); typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_update_proc) (ma_snd_pcm_t *pcm); typedef int (* ma_snd_pcm_wait_proc) (ma_snd_pcm_t *pcm, int timeout); typedef int (* ma_snd_pcm_nonblock_proc) (ma_snd_pcm_t *pcm, int nonblock); typedef int (* ma_snd_pcm_info_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_info_t* info); typedef size_t (* ma_snd_pcm_info_sizeof_proc) (void); typedef const char* (* ma_snd_pcm_info_get_name_proc) (const ma_snd_pcm_info_t* info); typedef int (* ma_snd_pcm_poll_descriptors_proc) (ma_snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space); typedef int (* ma_snd_pcm_poll_descriptors_count_proc) (ma_snd_pcm_t *pcm); typedef int (* ma_snd_pcm_poll_descriptors_revents_proc) (ma_snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents); typedef int (* ma_snd_config_update_free_global_proc) (void); /* This array specifies each of the common devices that can be used for both playback and capture. */ static const char* g_maCommonDeviceNamesALSA[] = { "default", "null", "pulse", "jack" }; /* This array allows us to blacklist specific playback devices. */ static const char* g_maBlacklistedPlaybackDeviceNamesALSA[] = { "" }; /* This array allows us to blacklist specific capture devices. */ static const char* g_maBlacklistedCaptureDeviceNamesALSA[] = { "" }; static ma_snd_pcm_format_t ma_convert_ma_format_to_alsa_format(ma_format format) { ma_snd_pcm_format_t ALSAFormats[] = { MA_SND_PCM_FORMAT_UNKNOWN, /* ma_format_unknown */ MA_SND_PCM_FORMAT_U8, /* ma_format_u8 */ MA_SND_PCM_FORMAT_S16_LE, /* ma_format_s16 */ MA_SND_PCM_FORMAT_S24_3LE, /* ma_format_s24 */ MA_SND_PCM_FORMAT_S32_LE, /* ma_format_s32 */ MA_SND_PCM_FORMAT_FLOAT_LE /* ma_format_f32 */ }; if (ma_is_big_endian()) { ALSAFormats[0] = MA_SND_PCM_FORMAT_UNKNOWN; ALSAFormats[1] = MA_SND_PCM_FORMAT_U8; ALSAFormats[2] = MA_SND_PCM_FORMAT_S16_BE; ALSAFormats[3] = MA_SND_PCM_FORMAT_S24_3BE; ALSAFormats[4] = MA_SND_PCM_FORMAT_S32_BE; ALSAFormats[5] = MA_SND_PCM_FORMAT_FLOAT_BE; } return ALSAFormats[format]; } static ma_format ma_format_from_alsa(ma_snd_pcm_format_t formatALSA) { if (ma_is_little_endian()) { switch (formatALSA) { case MA_SND_PCM_FORMAT_S16_LE: return ma_format_s16; case MA_SND_PCM_FORMAT_S24_3LE: return ma_format_s24; case MA_SND_PCM_FORMAT_S32_LE: return ma_format_s32; case MA_SND_PCM_FORMAT_FLOAT_LE: return ma_format_f32; default: break; } } else { switch (formatALSA) { case MA_SND_PCM_FORMAT_S16_BE: return ma_format_s16; case MA_SND_PCM_FORMAT_S24_3BE: return ma_format_s24; case MA_SND_PCM_FORMAT_S32_BE: return ma_format_s32; case MA_SND_PCM_FORMAT_FLOAT_BE: return ma_format_f32; default: break; } } /* Endian agnostic. */ switch (formatALSA) { case MA_SND_PCM_FORMAT_U8: return ma_format_u8; default: return ma_format_unknown; } } static ma_channel ma_convert_alsa_channel_position_to_ma_channel(unsigned int alsaChannelPos) { switch (alsaChannelPos) { case MA_SND_CHMAP_MONO: return MA_CHANNEL_MONO; case MA_SND_CHMAP_FL: return MA_CHANNEL_FRONT_LEFT; case MA_SND_CHMAP_FR: return MA_CHANNEL_FRONT_RIGHT; case MA_SND_CHMAP_RL: return MA_CHANNEL_BACK_LEFT; case MA_SND_CHMAP_RR: return MA_CHANNEL_BACK_RIGHT; case MA_SND_CHMAP_FC: return MA_CHANNEL_FRONT_CENTER; case MA_SND_CHMAP_LFE: return MA_CHANNEL_LFE; case MA_SND_CHMAP_SL: return MA_CHANNEL_SIDE_LEFT; case MA_SND_CHMAP_SR: return MA_CHANNEL_SIDE_RIGHT; case MA_SND_CHMAP_RC: return MA_CHANNEL_BACK_CENTER; case MA_SND_CHMAP_FLC: return MA_CHANNEL_FRONT_LEFT_CENTER; case MA_SND_CHMAP_FRC: return MA_CHANNEL_FRONT_RIGHT_CENTER; case MA_SND_CHMAP_RLC: return 0; case MA_SND_CHMAP_RRC: return 0; case MA_SND_CHMAP_FLW: return 0; case MA_SND_CHMAP_FRW: return 0; case MA_SND_CHMAP_FLH: return 0; case MA_SND_CHMAP_FCH: return 0; case MA_SND_CHMAP_FRH: return 0; case MA_SND_CHMAP_TC: return MA_CHANNEL_TOP_CENTER; case MA_SND_CHMAP_TFL: return MA_CHANNEL_TOP_FRONT_LEFT; case MA_SND_CHMAP_TFR: return MA_CHANNEL_TOP_FRONT_RIGHT; case MA_SND_CHMAP_TFC: return MA_CHANNEL_TOP_FRONT_CENTER; case MA_SND_CHMAP_TRL: return MA_CHANNEL_TOP_BACK_LEFT; case MA_SND_CHMAP_TRR: return MA_CHANNEL_TOP_BACK_RIGHT; case MA_SND_CHMAP_TRC: return MA_CHANNEL_TOP_BACK_CENTER; default: break; } return 0; } static ma_bool32 ma_is_common_device_name__alsa(const char* name) { size_t iName; for (iName = 0; iName < ma_countof(g_maCommonDeviceNamesALSA); ++iName) { if (ma_strcmp(name, g_maCommonDeviceNamesALSA[iName]) == 0) { return MA_TRUE; } } return MA_FALSE; } static ma_bool32 ma_is_playback_device_blacklisted__alsa(const char* name) { size_t iName; for (iName = 0; iName < ma_countof(g_maBlacklistedPlaybackDeviceNamesALSA); ++iName) { if (ma_strcmp(name, g_maBlacklistedPlaybackDeviceNamesALSA[iName]) == 0) { return MA_TRUE; } } return MA_FALSE; } static ma_bool32 ma_is_capture_device_blacklisted__alsa(const char* name) { size_t iName; for (iName = 0; iName < ma_countof(g_maBlacklistedCaptureDeviceNamesALSA); ++iName) { if (ma_strcmp(name, g_maBlacklistedCaptureDeviceNamesALSA[iName]) == 0) { return MA_TRUE; } } return MA_FALSE; } static ma_bool32 ma_is_device_blacklisted__alsa(ma_device_type deviceType, const char* name) { if (deviceType == ma_device_type_playback) { return ma_is_playback_device_blacklisted__alsa(name); } else { return ma_is_capture_device_blacklisted__alsa(name); } } static const char* ma_find_char(const char* str, char c, int* index) { int i = 0; for (;;) { if (str[i] == '\0') { if (index) *index = -1; return NULL; } if (str[i] == c) { if (index) *index = i; return str + i; } i += 1; } /* Should never get here, but treat it as though the character was not found to make me feel better inside. */ if (index) *index = -1; return NULL; } static ma_bool32 ma_is_device_name_in_hw_format__alsa(const char* hwid) { /* This function is just checking whether or not hwid is in "hw:%d,%d" format. */ int commaPos; const char* dev; int i; if (hwid == NULL) { return MA_FALSE; } if (hwid[0] != 'h' || hwid[1] != 'w' || hwid[2] != ':') { return MA_FALSE; } hwid += 3; dev = ma_find_char(hwid, ',', &commaPos); if (dev == NULL) { return MA_FALSE; } else { dev += 1; /* Skip past the ",". */ } /* Check if the part between the ":" and the "," contains only numbers. If not, return false. */ for (i = 0; i < commaPos; ++i) { if (hwid[i] < '0' || hwid[i] > '9') { return MA_FALSE; } } /* Check if everything after the "," is numeric. If not, return false. */ i = 0; while (dev[i] != '\0') { if (dev[i] < '0' || dev[i] > '9') { return MA_FALSE; } i += 1; } return MA_TRUE; } static int ma_convert_device_name_to_hw_format__alsa(ma_context* pContext, char* dst, size_t dstSize, const char* src) /* Returns 0 on success, non-0 on error. */ { /* src should look something like this: "hw:CARD=I82801AAICH,DEV=0" */ int colonPos; int commaPos; char card[256]; const char* dev; int cardIndex; if (dst == NULL) { return -1; } if (dstSize < 7) { return -1; /* Absolute minimum size of the output buffer is 7 bytes. */ } *dst = '\0'; /* Safety. */ if (src == NULL) { return -1; } /* If the input name is already in "hw:%d,%d" format, just return that verbatim. */ if (ma_is_device_name_in_hw_format__alsa(src)) { return ma_strcpy_s(dst, dstSize, src); } src = ma_find_char(src, ':', &colonPos); if (src == NULL) { return -1; /* Couldn't find a colon */ } dev = ma_find_char(src, ',', &commaPos); if (dev == NULL) { dev = "0"; ma_strncpy_s(card, sizeof(card), src+6, (size_t)-1); /* +6 = ":CARD=" */ } else { dev = dev + 5; /* +5 = ",DEV=" */ ma_strncpy_s(card, sizeof(card), src+6, commaPos-6); /* +6 = ":CARD=" */ } cardIndex = ((ma_snd_card_get_index_proc)pContext->alsa.snd_card_get_index)(card); if (cardIndex < 0) { return -2; /* Failed to retrieve the card index. */ } /* Construction. */ dst[0] = 'h'; dst[1] = 'w'; dst[2] = ':'; if (ma_itoa_s(cardIndex, dst+3, dstSize-3, 10) != 0) { return -3; } if (ma_strcat_s(dst, dstSize, ",") != 0) { return -3; } if (ma_strcat_s(dst, dstSize, dev) != 0) { return -3; } return 0; } static ma_bool32 ma_does_id_exist_in_list__alsa(ma_device_id* pUniqueIDs, ma_uint32 count, const char* pHWID) { ma_uint32 i; MA_ASSERT(pHWID != NULL); for (i = 0; i < count; ++i) { if (ma_strcmp(pUniqueIDs[i].alsa, pHWID) == 0) { return MA_TRUE; } } return MA_FALSE; } static ma_result ma_context_open_pcm__alsa(ma_context* pContext, ma_share_mode shareMode, ma_device_type deviceType, const ma_device_id* pDeviceID, int openMode, ma_snd_pcm_t** ppPCM) { ma_snd_pcm_t* pPCM; ma_snd_pcm_stream_t stream; MA_ASSERT(pContext != NULL); MA_ASSERT(ppPCM != NULL); *ppPCM = NULL; pPCM = NULL; stream = (deviceType == ma_device_type_playback) ? MA_SND_PCM_STREAM_PLAYBACK : MA_SND_PCM_STREAM_CAPTURE; if (pDeviceID == NULL) { ma_bool32 isDeviceOpen; size_t i; /* We're opening the default device. I don't know if trying anything other than "default" is necessary, but it makes me feel better to try as hard as we can get to get _something_ working. */ const char* defaultDeviceNames[] = { "default", NULL, NULL, NULL, NULL, NULL, NULL }; if (shareMode == ma_share_mode_exclusive) { defaultDeviceNames[1] = "hw"; defaultDeviceNames[2] = "hw:0"; defaultDeviceNames[3] = "hw:0,0"; } else { if (deviceType == ma_device_type_playback) { defaultDeviceNames[1] = "dmix"; defaultDeviceNames[2] = "dmix:0"; defaultDeviceNames[3] = "dmix:0,0"; } else { defaultDeviceNames[1] = "dsnoop"; defaultDeviceNames[2] = "dsnoop:0"; defaultDeviceNames[3] = "dsnoop:0,0"; } defaultDeviceNames[4] = "hw"; defaultDeviceNames[5] = "hw:0"; defaultDeviceNames[6] = "hw:0,0"; } isDeviceOpen = MA_FALSE; for (i = 0; i < ma_countof(defaultDeviceNames); ++i) { if (defaultDeviceNames[i] != NULL && defaultDeviceNames[i][0] != '\0') { if (((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, defaultDeviceNames[i], stream, openMode) == 0) { isDeviceOpen = MA_TRUE; break; } } } if (!isDeviceOpen) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_open() failed when trying to open an appropriate default device."); return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } } else { /* We're trying to open a specific device. There's a few things to consider here: miniaudio recognizes a special format of device id that excludes the "hw", "dmix", etc. prefix. It looks like this: ":0,0", ":0,1", etc. When an ID of this format is specified, it indicates to miniaudio that it can try different combinations of plugins ("hw", "dmix", etc.) until it finds an appropriate one that works. This comes in very handy when trying to open a device in shared mode ("dmix"), vs exclusive mode ("hw"). */ /* May end up needing to make small adjustments to the ID, so make a copy. */ ma_device_id deviceID = *pDeviceID; int resultALSA = -ENODEV; if (deviceID.alsa[0] != ':') { /* The ID is not in ":0,0" format. Use the ID exactly as-is. */ resultALSA = ((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, deviceID.alsa, stream, openMode); } else { char hwid[256]; /* The ID is in ":0,0" format. Try different plugins depending on the shared mode. */ if (deviceID.alsa[1] == '\0') { deviceID.alsa[0] = '\0'; /* An ID of ":" should be converted to "". */ } if (shareMode == ma_share_mode_shared) { if (deviceType == ma_device_type_playback) { ma_strcpy_s(hwid, sizeof(hwid), "dmix"); } else { ma_strcpy_s(hwid, sizeof(hwid), "dsnoop"); } if (ma_strcat_s(hwid, sizeof(hwid), deviceID.alsa) == 0) { resultALSA = ((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, hwid, stream, openMode); } } /* If at this point we still don't have an open device it means we're either preferencing exclusive mode or opening with "dmix"/"dsnoop" failed. */ if (resultALSA != 0) { ma_strcpy_s(hwid, sizeof(hwid), "hw"); if (ma_strcat_s(hwid, sizeof(hwid), deviceID.alsa) == 0) { resultALSA = ((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, hwid, stream, openMode); } } } if (resultALSA < 0) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_open() failed."); return ma_result_from_errno(-resultALSA); } } *ppPCM = pPCM; return MA_SUCCESS; } static ma_result ma_context_enumerate_devices__alsa(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { int resultALSA; ma_bool32 cbResult = MA_TRUE; char** ppDeviceHints; ma_device_id* pUniqueIDs = NULL; ma_uint32 uniqueIDCount = 0; char** ppNextDeviceHint; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); ma_mutex_lock(&pContext->alsa.internalDeviceEnumLock); resultALSA = ((ma_snd_device_name_hint_proc)pContext->alsa.snd_device_name_hint)(-1, "pcm", (void***)&ppDeviceHints); if (resultALSA < 0) { ma_mutex_unlock(&pContext->alsa.internalDeviceEnumLock); return ma_result_from_errno(-resultALSA); } ppNextDeviceHint = ppDeviceHints; while (*ppNextDeviceHint != NULL) { char* NAME = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "NAME"); char* DESC = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "DESC"); char* IOID = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "IOID"); ma_device_type deviceType = ma_device_type_playback; ma_bool32 stopEnumeration = MA_FALSE; char hwid[sizeof(pUniqueIDs->alsa)]; ma_device_info deviceInfo; if ((IOID == NULL || ma_strcmp(IOID, "Output") == 0)) { deviceType = ma_device_type_playback; } if ((IOID != NULL && ma_strcmp(IOID, "Input" ) == 0)) { deviceType = ma_device_type_capture; } if (NAME != NULL) { if (pContext->alsa.useVerboseDeviceEnumeration) { /* Verbose mode. Use the name exactly as-is. */ ma_strncpy_s(hwid, sizeof(hwid), NAME, (size_t)-1); } else { /* Simplified mode. Use ":%d,%d" format. */ if (ma_convert_device_name_to_hw_format__alsa(pContext, hwid, sizeof(hwid), NAME) == 0) { /* At this point, hwid looks like "hw:0,0". In simplified enumeration mode, we actually want to strip off the plugin name so it looks like ":0,0". The reason for this is that this special format is detected at device initialization time and is used as an indicator to try to use the most appropriate plugin depending on the device type and sharing mode. */ char* dst = hwid; char* src = hwid+2; while ((*dst++ = *src++)); } else { /* Conversion to "hw:%d,%d" failed. Just use the name as-is. */ ma_strncpy_s(hwid, sizeof(hwid), NAME, (size_t)-1); } if (ma_does_id_exist_in_list__alsa(pUniqueIDs, uniqueIDCount, hwid)) { goto next_device; /* The device has already been enumerated. Move on to the next one. */ } else { /* The device has not yet been enumerated. Make sure it's added to our list so that it's not enumerated again. */ size_t newCapacity = sizeof(*pUniqueIDs) * (uniqueIDCount + 1); ma_device_id* pNewUniqueIDs = (ma_device_id*)ma_realloc(pUniqueIDs, newCapacity, &pContext->allocationCallbacks); if (pNewUniqueIDs == NULL) { goto next_device; /* Failed to allocate memory. */ } pUniqueIDs = pNewUniqueIDs; MA_COPY_MEMORY(pUniqueIDs[uniqueIDCount].alsa, hwid, sizeof(hwid)); uniqueIDCount += 1; } } } else { MA_ZERO_MEMORY(hwid, sizeof(hwid)); } MA_ZERO_OBJECT(&deviceInfo); ma_strncpy_s(deviceInfo.id.alsa, sizeof(deviceInfo.id.alsa), hwid, (size_t)-1); /* There's no good way to determine whether or not a device is the default on Linux. We're just going to do something simple and just use the name of "default" as the indicator. */ if (ma_strcmp(deviceInfo.id.alsa, "default") == 0) { deviceInfo.isDefault = MA_TRUE; } /* DESC is the friendly name. We treat this slightly differently depending on whether or not we are using verbose device enumeration. In verbose mode we want to take the entire description so that the end-user can distinguish between the subdevices of each card/dev pair. In simplified mode, however, we only want the first part of the description. The value in DESC seems to be split into two lines, with the first line being the name of the device and the second line being a description of the device. I don't like having the description be across two lines because it makes formatting ugly and annoying. I'm therefore deciding to put it all on a single line with the second line being put into parentheses. In simplified mode I'm just stripping the second line entirely. */ if (DESC != NULL) { int lfPos; const char* line2 = ma_find_char(DESC, '\n', &lfPos); if (line2 != NULL) { line2 += 1; /* Skip past the new-line character. */ if (pContext->alsa.useVerboseDeviceEnumeration) { /* Verbose mode. Put the second line in brackets. */ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, lfPos); ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), " ("); ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), line2); ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), ")"); } else { /* Simplified mode. Strip the second line entirely. */ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, lfPos); } } else { /* There's no second line. Just copy the whole description. */ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, (size_t)-1); } } if (!ma_is_device_blacklisted__alsa(deviceType, NAME)) { cbResult = callback(pContext, deviceType, &deviceInfo, pUserData); } /* Some devices are both playback and capture, but they are only enumerated by ALSA once. We need to fire the callback again for the other device type in this case. We do this for known devices and where the IOID hint is NULL, which means both Input and Output. */ if (cbResult) { if (ma_is_common_device_name__alsa(NAME) || IOID == NULL) { if (deviceType == ma_device_type_playback) { if (!ma_is_capture_device_blacklisted__alsa(NAME)) { cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); } } else { if (!ma_is_playback_device_blacklisted__alsa(NAME)) { cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } } } } if (cbResult == MA_FALSE) { stopEnumeration = MA_TRUE; } next_device: free(NAME); free(DESC); free(IOID); ppNextDeviceHint += 1; /* We need to stop enumeration if the callback returned false. */ if (stopEnumeration) { break; } } ma_free(pUniqueIDs, &pContext->allocationCallbacks); ((ma_snd_device_name_free_hint_proc)pContext->alsa.snd_device_name_free_hint)((void**)ppDeviceHints); ma_mutex_unlock(&pContext->alsa.internalDeviceEnumLock); return MA_SUCCESS; } typedef struct { ma_device_type deviceType; const ma_device_id* pDeviceID; ma_share_mode shareMode; ma_device_info* pDeviceInfo; ma_bool32 foundDevice; } ma_context_get_device_info_enum_callback_data__alsa; static ma_bool32 ma_context_get_device_info_enum_callback__alsa(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pDeviceInfo, void* pUserData) { ma_context_get_device_info_enum_callback_data__alsa* pData = (ma_context_get_device_info_enum_callback_data__alsa*)pUserData; MA_ASSERT(pData != NULL); (void)pContext; if (pData->pDeviceID == NULL && ma_strcmp(pDeviceInfo->id.alsa, "default") == 0) { ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pDeviceInfo->name, (size_t)-1); pData->foundDevice = MA_TRUE; } else { if (pData->deviceType == deviceType && (pData->pDeviceID != NULL && ma_strcmp(pData->pDeviceID->alsa, pDeviceInfo->id.alsa) == 0)) { ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pDeviceInfo->name, (size_t)-1); pData->foundDevice = MA_TRUE; } } /* Keep enumerating until we have found the device. */ return !pData->foundDevice; } static void ma_context_test_rate_and_add_native_data_format__alsa(ma_context* pContext, ma_snd_pcm_t* pPCM, ma_snd_pcm_hw_params_t* pHWParams, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 flags, ma_device_info* pDeviceInfo) { MA_ASSERT(pPCM != NULL); MA_ASSERT(pHWParams != NULL); MA_ASSERT(pDeviceInfo != NULL); if (pDeviceInfo->nativeDataFormatCount < ma_countof(pDeviceInfo->nativeDataFormats) && ((ma_snd_pcm_hw_params_test_rate_proc)pContext->alsa.snd_pcm_hw_params_test_rate)(pPCM, pHWParams, sampleRate, 0) == 0) { pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = flags; pDeviceInfo->nativeDataFormatCount += 1; } } static void ma_context_iterate_rates_and_add_native_data_format__alsa(ma_context* pContext, ma_snd_pcm_t* pPCM, ma_snd_pcm_hw_params_t* pHWParams, ma_format format, ma_uint32 channels, ma_uint32 flags, ma_device_info* pDeviceInfo) { ma_uint32 iSampleRate; unsigned int minSampleRate; unsigned int maxSampleRate; int sampleRateDir; /* Not used. Just passed into snd_pcm_hw_params_get_rate_min/max(). */ /* There could be a range. */ ((ma_snd_pcm_hw_params_get_rate_min_proc)pContext->alsa.snd_pcm_hw_params_get_rate_min)(pHWParams, &minSampleRate, &sampleRateDir); ((ma_snd_pcm_hw_params_get_rate_max_proc)pContext->alsa.snd_pcm_hw_params_get_rate_max)(pHWParams, &maxSampleRate, &sampleRateDir); /* Make sure our sample rates are clamped to sane values. Stupid devices like "pulse" will reports rates like "1" which is ridiculous. */ minSampleRate = ma_clamp(minSampleRate, (unsigned int)ma_standard_sample_rate_min, (unsigned int)ma_standard_sample_rate_max); maxSampleRate = ma_clamp(maxSampleRate, (unsigned int)ma_standard_sample_rate_min, (unsigned int)ma_standard_sample_rate_max); for (iSampleRate = 0; iSampleRate < ma_countof(g_maStandardSampleRatePriorities); iSampleRate += 1) { ma_uint32 standardSampleRate = g_maStandardSampleRatePriorities[iSampleRate]; if (standardSampleRate >= minSampleRate && standardSampleRate <= maxSampleRate) { ma_context_test_rate_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, standardSampleRate, flags, pDeviceInfo); } } /* Now make sure our min and max rates are included just in case they aren't in the range of our standard rates. */ if (!ma_is_standard_sample_rate(minSampleRate)) { ma_context_test_rate_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, minSampleRate, flags, pDeviceInfo); } if (!ma_is_standard_sample_rate(maxSampleRate) && maxSampleRate != minSampleRate) { ma_context_test_rate_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, maxSampleRate, flags, pDeviceInfo); } } static ma_result ma_context_get_device_info__alsa(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { ma_context_get_device_info_enum_callback_data__alsa data; ma_result result; int resultALSA; ma_snd_pcm_t* pPCM; ma_snd_pcm_hw_params_t* pHWParams; ma_uint32 iFormat; ma_uint32 iChannel; MA_ASSERT(pContext != NULL); /* We just enumerate to find basic information about the device. */ data.deviceType = deviceType; data.pDeviceID = pDeviceID; data.pDeviceInfo = pDeviceInfo; data.foundDevice = MA_FALSE; result = ma_context_enumerate_devices__alsa(pContext, ma_context_get_device_info_enum_callback__alsa, &data); if (result != MA_SUCCESS) { return result; } if (!data.foundDevice) { return MA_NO_DEVICE; } if (ma_strcmp(pDeviceInfo->id.alsa, "default") == 0) { pDeviceInfo->isDefault = MA_TRUE; } /* For detailed info we need to open the device. */ result = ma_context_open_pcm__alsa(pContext, ma_share_mode_shared, deviceType, pDeviceID, 0, &pPCM); if (result != MA_SUCCESS) { return result; } /* We need to initialize a HW parameters object in order to know what formats are supported. */ pHWParams = (ma_snd_pcm_hw_params_t*)ma_calloc(((ma_snd_pcm_hw_params_sizeof_proc)pContext->alsa.snd_pcm_hw_params_sizeof)(), &pContext->allocationCallbacks); if (pHWParams == NULL) { ((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM); return MA_OUT_OF_MEMORY; } resultALSA = ((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams); if (resultALSA < 0) { ma_free(pHWParams, &pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM); ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize hardware parameters. snd_pcm_hw_params_any() failed."); return ma_result_from_errno(-resultALSA); } /* Some ALSA devices can support many permutations of formats, channels and rates. We only support a fixed number of permutations which means we need to employ some strategies to ensure the best combinations are returned. An example is the "pulse" device which can do its own data conversion in software and as a result can support any combination of format, channels and rate. We want to ensure that the first data formats are the best. We have a list of favored sample formats and sample rates, so these will be the basis of our iteration. */ /* Formats. We just iterate over our standard formats and test them, making sure we reset the configuration space each iteration. */ for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); iFormat += 1) { ma_format format = g_maFormatPriorities[iFormat]; /* For each format we need to make sure we reset the configuration space so we don't return channel counts and rates that aren't compatible with a format. */ ((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams); /* Test the format first. If this fails it means the format is not supported and we can skip it. */ if (((ma_snd_pcm_hw_params_test_format_proc)pContext->alsa.snd_pcm_hw_params_test_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(format)) == 0) { /* The format is supported. */ unsigned int minChannels; unsigned int maxChannels; /* The configuration space needs to be restricted to this format so we can get an accurate picture of which sample rates and channel counts are support with this format. */ ((ma_snd_pcm_hw_params_set_format_proc)pContext->alsa.snd_pcm_hw_params_set_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(format)); /* Now we need to check for supported channels. */ ((ma_snd_pcm_hw_params_get_channels_min_proc)pContext->alsa.snd_pcm_hw_params_get_channels_min)(pHWParams, &minChannels); ((ma_snd_pcm_hw_params_get_channels_max_proc)pContext->alsa.snd_pcm_hw_params_get_channels_max)(pHWParams, &maxChannels); if (minChannels > MA_MAX_CHANNELS) { continue; /* Too many channels. */ } if (maxChannels < MA_MIN_CHANNELS) { continue; /* Not enough channels. */ } /* Make sure the channel count is clamped. This is mainly intended for the max channels because some devices can report an unbound maximum. */ minChannels = ma_clamp(minChannels, MA_MIN_CHANNELS, MA_MAX_CHANNELS); maxChannels = ma_clamp(maxChannels, MA_MIN_CHANNELS, MA_MAX_CHANNELS); if (minChannels == MA_MIN_CHANNELS && maxChannels == MA_MAX_CHANNELS) { /* The device supports all channels. Don't iterate over every single one. Instead just set the channels to 0 which means all channels are supported. */ ma_context_iterate_rates_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, 0, 0, pDeviceInfo); /* Intentionally setting the channel count to 0 as that means all channels are supported. */ } else { /* The device only supports a specific set of channels. We need to iterate over all of them. */ for (iChannel = minChannels; iChannel <= maxChannels; iChannel += 1) { /* Test the channel before applying it to the configuration space. */ unsigned int channels = iChannel; /* Make sure our channel range is reset before testing again or else we'll always fail the test. */ ((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams); ((ma_snd_pcm_hw_params_set_format_proc)pContext->alsa.snd_pcm_hw_params_set_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(format)); if (((ma_snd_pcm_hw_params_test_channels_proc)pContext->alsa.snd_pcm_hw_params_test_channels)(pPCM, pHWParams, channels) == 0) { /* The channel count is supported. */ /* The configuration space now needs to be restricted to the channel count before extracting the sample rate. */ ((ma_snd_pcm_hw_params_set_channels_proc)pContext->alsa.snd_pcm_hw_params_set_channels)(pPCM, pHWParams, channels); /* Only after the configuration space has been restricted to the specific channel count should we iterate over our sample rates. */ ma_context_iterate_rates_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, 0, pDeviceInfo); } else { /* The channel count is not supported. Skip. */ } } } } else { /* The format is not supported. Skip. */ } } ma_free(pHWParams, &pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM); return MA_SUCCESS; } static ma_result ma_device_uninit__alsa(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if ((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) { ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture); close(pDevice->alsa.wakeupfdCapture); ma_free(pDevice->alsa.pPollDescriptorsCapture, &pDevice->pContext->allocationCallbacks); } if ((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) { ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback); close(pDevice->alsa.wakeupfdPlayback); ma_free(pDevice->alsa.pPollDescriptorsPlayback, &pDevice->pContext->allocationCallbacks); } return MA_SUCCESS; } static ma_result ma_device_init_by_type__alsa(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType) { ma_result result; int resultALSA; ma_snd_pcm_t* pPCM; ma_bool32 isUsingMMap; ma_snd_pcm_format_t formatALSA; ma_format internalFormat; ma_uint32 internalChannels; ma_uint32 internalSampleRate; ma_channel internalChannelMap[MA_MAX_CHANNELS]; ma_uint32 internalPeriodSizeInFrames; ma_uint32 internalPeriods; int openMode; ma_snd_pcm_hw_params_t* pHWParams; ma_snd_pcm_sw_params_t* pSWParams; ma_snd_pcm_uframes_t bufferBoundary; int pollDescriptorCount; struct pollfd* pPollDescriptors; int wakeupfd; MA_ASSERT(pConfig != NULL); MA_ASSERT(deviceType != ma_device_type_duplex); /* This function should only be called for playback _or_ capture, never duplex. */ MA_ASSERT(pDevice != NULL); formatALSA = ma_convert_ma_format_to_alsa_format(pDescriptor->format); openMode = 0; if (pConfig->alsa.noAutoResample) { openMode |= MA_SND_PCM_NO_AUTO_RESAMPLE; } if (pConfig->alsa.noAutoChannels) { openMode |= MA_SND_PCM_NO_AUTO_CHANNELS; } if (pConfig->alsa.noAutoFormat) { openMode |= MA_SND_PCM_NO_AUTO_FORMAT; } result = ma_context_open_pcm__alsa(pDevice->pContext, pDescriptor->shareMode, deviceType, pDescriptor->pDeviceID, openMode, &pPCM); if (result != MA_SUCCESS) { return result; } /* Hardware parameters. */ pHWParams = (ma_snd_pcm_hw_params_t*)ma_calloc(((ma_snd_pcm_hw_params_sizeof_proc)pDevice->pContext->alsa.snd_pcm_hw_params_sizeof)(), &pDevice->pContext->allocationCallbacks); if (pHWParams == NULL) { ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to allocate memory for hardware parameters."); return MA_OUT_OF_MEMORY; } resultALSA = ((ma_snd_pcm_hw_params_any_proc)pDevice->pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams); if (resultALSA < 0) { ma_free(pHWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize hardware parameters. snd_pcm_hw_params_any() failed."); return ma_result_from_errno(-resultALSA); } /* MMAP Mode. Try using interleaved MMAP access. If this fails, fall back to standard readi/writei. */ isUsingMMap = MA_FALSE; #if 0 /* NOTE: MMAP mode temporarily disabled. */ if (deviceType != ma_device_type_capture) { /* <-- Disabling MMAP mode for capture devices because I apparently do not have a device that supports it which means I can't test it... Contributions welcome. */ if (!pConfig->alsa.noMMap) { if (((ma_snd_pcm_hw_params_set_access_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_MMAP_INTERLEAVED) == 0) { pDevice->alsa.isUsingMMap = MA_TRUE; } } } #endif if (!isUsingMMap) { resultALSA = ((ma_snd_pcm_hw_params_set_access_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_RW_INTERLEAVED); if (resultALSA < 0) { ma_free(pHWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set access mode to neither SND_PCM_ACCESS_MMAP_INTERLEAVED nor SND_PCM_ACCESS_RW_INTERLEAVED. snd_pcm_hw_params_set_access() failed."); return ma_result_from_errno(-resultALSA); } } /* Most important properties first. The documentation for OSS (yes, I know this is ALSA!) recommends format, channels, then sample rate. I can't find any documentation for ALSA specifically, so I'm going to copy the recommendation for OSS. */ /* Format. */ { /* At this point we should have a list of supported formats, so now we need to find the best one. We first check if the requested format is supported, and if so, use that one. If it's not supported, we just run though a list of formats and try to find the best one. */ if (formatALSA == MA_SND_PCM_FORMAT_UNKNOWN || ((ma_snd_pcm_hw_params_test_format_proc)pDevice->pContext->alsa.snd_pcm_hw_params_test_format)(pPCM, pHWParams, formatALSA) != 0) { /* We're either requesting the native format or the specified format is not supported. */ size_t iFormat; formatALSA = MA_SND_PCM_FORMAT_UNKNOWN; for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); ++iFormat) { if (((ma_snd_pcm_hw_params_test_format_proc)pDevice->pContext->alsa.snd_pcm_hw_params_test_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(g_maFormatPriorities[iFormat])) == 0) { formatALSA = ma_convert_ma_format_to_alsa_format(g_maFormatPriorities[iFormat]); break; } } if (formatALSA == MA_SND_PCM_FORMAT_UNKNOWN) { ma_free(pHWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Format not supported. The device does not support any miniaudio formats."); return MA_FORMAT_NOT_SUPPORTED; } } resultALSA = ((ma_snd_pcm_hw_params_set_format_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_format)(pPCM, pHWParams, formatALSA); if (resultALSA < 0) { ma_free(pHWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Format not supported. snd_pcm_hw_params_set_format() failed."); return ma_result_from_errno(-resultALSA); } internalFormat = ma_format_from_alsa(formatALSA); if (internalFormat == ma_format_unknown) { ma_free(pHWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] The chosen format is not supported by miniaudio."); return MA_FORMAT_NOT_SUPPORTED; } } /* Channels. */ { unsigned int channels = pDescriptor->channels; if (channels == 0) { channels = MA_DEFAULT_CHANNELS; } resultALSA = ((ma_snd_pcm_hw_params_set_channels_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_channels_near)(pPCM, pHWParams, &channels); if (resultALSA < 0) { ma_free(pHWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set channel count. snd_pcm_hw_params_set_channels_near() failed."); return ma_result_from_errno(-resultALSA); } internalChannels = (ma_uint32)channels; } /* Sample Rate */ { unsigned int sampleRate; /* It appears there's either a bug in ALSA, a bug in some drivers, or I'm doing something silly; but having resampling enabled causes problems with some device configurations when used in conjunction with MMAP access mode. To fix this problem we need to disable resampling. To reproduce this problem, open the "plug:dmix" device, and set the sample rate to 44100. Internally, it looks like dmix uses a sample rate of 48000. The hardware parameters will get set correctly with no errors, but it looks like the 44100 -> 48000 resampling doesn't work properly - but only with MMAP access mode. You will notice skipping/crackling in the audio, and it'll run at a slightly faster rate. miniaudio has built-in support for sample rate conversion (albeit low quality at the moment), so disabling resampling should be fine for us. The only problem is that it won't be taking advantage of any kind of hardware-accelerated resampling and it won't be very good quality until I get a chance to improve the quality of miniaudio's software sample rate conversion. I don't currently know if the dmix plugin is the only one with this error. Indeed, this is the only one I've been able to reproduce this error with. In the future, we may want to restrict the disabling of resampling to only known bad plugins. */ ((ma_snd_pcm_hw_params_set_rate_resample_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_rate_resample)(pPCM, pHWParams, 0); sampleRate = pDescriptor->sampleRate; if (sampleRate == 0) { sampleRate = MA_DEFAULT_SAMPLE_RATE; } resultALSA = ((ma_snd_pcm_hw_params_set_rate_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_rate_near)(pPCM, pHWParams, &sampleRate, 0); if (resultALSA < 0) { ma_free(pHWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Sample rate not supported. snd_pcm_hw_params_set_rate_near() failed."); return ma_result_from_errno(-resultALSA); } internalSampleRate = (ma_uint32)sampleRate; } /* Periods. */ { ma_uint32 periods = pDescriptor->periodCount; resultALSA = ((ma_snd_pcm_hw_params_set_periods_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_periods_near)(pPCM, pHWParams, &periods, NULL); if (resultALSA < 0) { ma_free(pHWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set period count. snd_pcm_hw_params_set_periods_near() failed."); return ma_result_from_errno(-resultALSA); } internalPeriods = periods; } /* Buffer Size */ { ma_snd_pcm_uframes_t actualBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile) * internalPeriods; resultALSA = ((ma_snd_pcm_hw_params_set_buffer_size_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_buffer_size_near)(pPCM, pHWParams, &actualBufferSizeInFrames); if (resultALSA < 0) { ma_free(pHWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set buffer size for device. snd_pcm_hw_params_set_buffer_size() failed."); return ma_result_from_errno(-resultALSA); } internalPeriodSizeInFrames = actualBufferSizeInFrames / internalPeriods; } /* Apply hardware parameters. */ resultALSA = ((ma_snd_pcm_hw_params_proc)pDevice->pContext->alsa.snd_pcm_hw_params)(pPCM, pHWParams); if (resultALSA < 0) { ma_free(pHWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set hardware parameters. snd_pcm_hw_params() failed."); return ma_result_from_errno(-resultALSA); } ma_free(pHWParams, &pDevice->pContext->allocationCallbacks); pHWParams = NULL; /* Software parameters. */ pSWParams = (ma_snd_pcm_sw_params_t*)ma_calloc(((ma_snd_pcm_sw_params_sizeof_proc)pDevice->pContext->alsa.snd_pcm_sw_params_sizeof)(), &pDevice->pContext->allocationCallbacks); if (pSWParams == NULL) { ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to allocate memory for software parameters."); return MA_OUT_OF_MEMORY; } resultALSA = ((ma_snd_pcm_sw_params_current_proc)pDevice->pContext->alsa.snd_pcm_sw_params_current)(pPCM, pSWParams); if (resultALSA < 0) { ma_free(pSWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize software parameters. snd_pcm_sw_params_current() failed."); return ma_result_from_errno(-resultALSA); } resultALSA = ((ma_snd_pcm_sw_params_set_avail_min_proc)pDevice->pContext->alsa.snd_pcm_sw_params_set_avail_min)(pPCM, pSWParams, ma_prev_power_of_2(internalPeriodSizeInFrames)); if (resultALSA < 0) { ma_free(pSWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_sw_params_set_avail_min() failed."); return ma_result_from_errno(-resultALSA); } resultALSA = ((ma_snd_pcm_sw_params_get_boundary_proc)pDevice->pContext->alsa.snd_pcm_sw_params_get_boundary)(pSWParams, &bufferBoundary); if (resultALSA < 0) { bufferBoundary = internalPeriodSizeInFrames * internalPeriods; } if (deviceType == ma_device_type_playback && !isUsingMMap) { /* Only playback devices in writei/readi mode need a start threshold. */ /* Subtle detail here with the start threshold. When in playback-only mode (no full-duplex) we can set the start threshold to the size of a period. But for full-duplex we need to set it such that it is at least two periods. */ resultALSA = ((ma_snd_pcm_sw_params_set_start_threshold_proc)pDevice->pContext->alsa.snd_pcm_sw_params_set_start_threshold)(pPCM, pSWParams, internalPeriodSizeInFrames*2); if (resultALSA < 0) { ma_free(pSWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set start threshold for playback device. snd_pcm_sw_params_set_start_threshold() failed."); return ma_result_from_errno(-resultALSA); } resultALSA = ((ma_snd_pcm_sw_params_set_stop_threshold_proc)pDevice->pContext->alsa.snd_pcm_sw_params_set_stop_threshold)(pPCM, pSWParams, bufferBoundary); if (resultALSA < 0) { /* Set to boundary to loop instead of stop in the event of an xrun. */ ma_free(pSWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set stop threshold for playback device. snd_pcm_sw_params_set_stop_threshold() failed."); return ma_result_from_errno(-resultALSA); } } resultALSA = ((ma_snd_pcm_sw_params_proc)pDevice->pContext->alsa.snd_pcm_sw_params)(pPCM, pSWParams); if (resultALSA < 0) { ma_free(pSWParams, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set software parameters. snd_pcm_sw_params() failed."); return ma_result_from_errno(-resultALSA); } ma_free(pSWParams, &pDevice->pContext->allocationCallbacks); pSWParams = NULL; /* Grab the internal channel map. For now we're not going to bother trying to change the channel map and instead just do it ourselves. */ { ma_snd_pcm_chmap_t* pChmap = NULL; if (pDevice->pContext->alsa.snd_pcm_get_chmap != NULL) { pChmap = ((ma_snd_pcm_get_chmap_proc)pDevice->pContext->alsa.snd_pcm_get_chmap)(pPCM); } if (pChmap != NULL) { ma_uint32 iChannel; /* There are cases where the returned channel map can have a different channel count than was returned by snd_pcm_hw_params_set_channels_near(). */ if (pChmap->channels >= internalChannels) { /* Drop excess channels. */ for (iChannel = 0; iChannel < internalChannels; ++iChannel) { internalChannelMap[iChannel] = ma_convert_alsa_channel_position_to_ma_channel(pChmap->pos[iChannel]); } } else { ma_uint32 i; /* Excess channels use defaults. Do an initial fill with defaults, overwrite the first pChmap->channels, validate to ensure there are no duplicate channels. If validation fails, fall back to defaults. */ ma_bool32 isValid = MA_TRUE; /* Fill with defaults. */ ma_channel_map_init_standard(ma_standard_channel_map_alsa, internalChannelMap, ma_countof(internalChannelMap), internalChannels); /* Overwrite first pChmap->channels channels. */ for (iChannel = 0; iChannel < pChmap->channels; ++iChannel) { internalChannelMap[iChannel] = ma_convert_alsa_channel_position_to_ma_channel(pChmap->pos[iChannel]); } /* Validate. */ for (i = 0; i < internalChannels && isValid; ++i) { ma_uint32 j; for (j = i+1; j < internalChannels; ++j) { if (internalChannelMap[i] == internalChannelMap[j]) { isValid = MA_FALSE; break; } } } /* If our channel map is invalid, fall back to defaults. */ if (!isValid) { ma_channel_map_init_standard(ma_standard_channel_map_alsa, internalChannelMap, ma_countof(internalChannelMap), internalChannels); } } free(pChmap); pChmap = NULL; } else { /* Could not retrieve the channel map. Fall back to a hard-coded assumption. */ ma_channel_map_init_standard(ma_standard_channel_map_alsa, internalChannelMap, ma_countof(internalChannelMap), internalChannels); } } /* We need to retrieve the poll descriptors so we can use poll() to wait for data to become available for reading or writing. There's no well defined maximum for this so we're just going to allocate this on the heap. */ pollDescriptorCount = ((ma_snd_pcm_poll_descriptors_count_proc)pDevice->pContext->alsa.snd_pcm_poll_descriptors_count)(pPCM); if (pollDescriptorCount <= 0) { ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to retrieve poll descriptors count."); return MA_ERROR; } pPollDescriptors = (struct pollfd*)ma_malloc(sizeof(*pPollDescriptors) * (pollDescriptorCount + 1), &pDevice->pContext->allocationCallbacks); /* +1 because we want room for the wakeup descriptor. */ if (pPollDescriptors == NULL) { ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to allocate memory for poll descriptors."); return MA_OUT_OF_MEMORY; } /* We need an eventfd to wakeup from poll() and avoid a deadlock in situations where the driver never returns from writei() and readi(). This has been observed with the "pulse" device. */ wakeupfd = eventfd(0, 0); if (wakeupfd < 0) { ma_free(pPollDescriptors, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to create eventfd for poll wakeup."); return ma_result_from_errno(errno); } /* We'll place the wakeup fd at the start of the buffer. */ pPollDescriptors[0].fd = wakeupfd; pPollDescriptors[0].events = POLLIN; /* We only care about waiting to read from the wakeup file descriptor. */ pPollDescriptors[0].revents = 0; /* We can now extract the PCM poll descriptors which we place after the wakeup descriptor. */ pollDescriptorCount = ((ma_snd_pcm_poll_descriptors_proc)pDevice->pContext->alsa.snd_pcm_poll_descriptors)(pPCM, pPollDescriptors + 1, pollDescriptorCount); /* +1 because we want to place these descriptors after the wakeup descriptor. */ if (pollDescriptorCount <= 0) { close(wakeupfd); ma_free(pPollDescriptors, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to retrieve poll descriptors."); return MA_ERROR; } if (deviceType == ma_device_type_capture) { pDevice->alsa.pollDescriptorCountCapture = pollDescriptorCount; pDevice->alsa.pPollDescriptorsCapture = pPollDescriptors; pDevice->alsa.wakeupfdCapture = wakeupfd; } else { pDevice->alsa.pollDescriptorCountPlayback = pollDescriptorCount; pDevice->alsa.pPollDescriptorsPlayback = pPollDescriptors; pDevice->alsa.wakeupfdPlayback = wakeupfd; } /* We're done. Prepare the device. */ resultALSA = ((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)(pPCM); if (resultALSA < 0) { close(wakeupfd); ma_free(pPollDescriptors, &pDevice->pContext->allocationCallbacks); ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to prepare device."); return ma_result_from_errno(-resultALSA); } if (deviceType == ma_device_type_capture) { pDevice->alsa.pPCMCapture = (ma_ptr)pPCM; pDevice->alsa.isUsingMMapCapture = isUsingMMap; } else { pDevice->alsa.pPCMPlayback = (ma_ptr)pPCM; pDevice->alsa.isUsingMMapPlayback = isUsingMMap; } pDescriptor->format = internalFormat; pDescriptor->channels = internalChannels; pDescriptor->sampleRate = internalSampleRate; ma_channel_map_copy(pDescriptor->channelMap, internalChannelMap, ma_min(internalChannels, MA_MAX_CHANNELS)); pDescriptor->periodSizeInFrames = internalPeriodSizeInFrames; pDescriptor->periodCount = internalPeriods; return MA_SUCCESS; } static ma_result ma_device_init__alsa(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { MA_ASSERT(pDevice != NULL); MA_ZERO_OBJECT(&pDevice->alsa); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { ma_result result = ma_device_init_by_type__alsa(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture); if (result != MA_SUCCESS) { return result; } } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { ma_result result = ma_device_init_by_type__alsa(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback); if (result != MA_SUCCESS) { return result; } } return MA_SUCCESS; } static ma_result ma_device_start__alsa(ma_device* pDevice) { int resultALSA; if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture); if (resultALSA < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start capture device."); return ma_result_from_errno(-resultALSA); } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { /* When data is written to the device we wait for the device to get ready to receive data with poll(). In my testing I have observed that poll() can sometimes block forever unless the device is started explicitly with snd_pcm_start() or some data is written with snd_pcm_writei(). To resolve this I've decided to do an explicit start with snd_pcm_start(). The problem with this is that the device is started without any data in the internal buffer which will result in an immediate underrun. If instead we were to call into snd_pcm_writei() in an attempt to prevent the underrun, we would run the risk of a weird deadlock issue as documented inside ma_device_write__alsa(). */ resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback); if (resultALSA < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start playback device."); return ma_result_from_errno(-resultALSA); } } return MA_SUCCESS; } static ma_result ma_device_stop__alsa(ma_device* pDevice) { /* The stop callback will get called on the worker thread after read/write__alsa() has returned. At this point there is a small chance that our wakeupfd has not been cleared. We'll clear that out now if applicable. */ int resultPoll; int resultRead; if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Dropping capture device...\n"); ((ma_snd_pcm_drop_proc)pDevice->pContext->alsa.snd_pcm_drop)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture); ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Dropping capture device successful.\n"); /* We need to prepare the device again, otherwise we won't be able to restart the device. */ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing capture device...\n"); if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) < 0) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing capture device failed.\n"); } else { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing capture device successful.\n"); } /* Clear the wakeupfd. */ resultPoll = poll((struct pollfd*)pDevice->alsa.pPollDescriptorsCapture, 1, 0); if (resultPoll > 0) { ma_uint64 t; resultRead = read(((struct pollfd*)pDevice->alsa.pPollDescriptorsCapture)[0].fd, &t, sizeof(t)); if (resultRead != sizeof(t)) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Failed to read from capture wakeupfd. read() = %d\n", resultRead); } } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Dropping playback device...\n"); ((ma_snd_pcm_drop_proc)pDevice->pContext->alsa.snd_pcm_drop)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback); ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Dropping playback device successful.\n"); /* We need to prepare the device again, otherwise we won't be able to restart the device. */ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing playback device...\n"); if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) < 0) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing playback device failed.\n"); } else { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing playback device successful.\n"); } /* Clear the wakeupfd. */ resultPoll = poll((struct pollfd*)pDevice->alsa.pPollDescriptorsPlayback, 1, 0); if (resultPoll > 0) { ma_uint64 t; resultRead = read(((struct pollfd*)pDevice->alsa.pPollDescriptorsPlayback)[0].fd, &t, sizeof(t)); if (resultRead != sizeof(t)) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Failed to read from playback wakeupfd. read() = %d\n", resultRead); } } } return MA_SUCCESS; } static ma_result ma_device_wait__alsa(ma_device* pDevice, ma_snd_pcm_t* pPCM, struct pollfd* pPollDescriptors, int pollDescriptorCount, short requiredEvent) { for (;;) { unsigned short revents; int resultALSA; int resultPoll = poll(pPollDescriptors, pollDescriptorCount, -1); if (resultPoll < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[ALSA] poll() failed.\n"); /* There have been reports that poll() is returning an error randomly and that instead of returning an error, simply trying again will work. I'm experimenting with adopting this advice. */ continue; /*return ma_result_from_errno(errno);*/ } /* Before checking the ALSA poll descriptor flag we need to check if the wakeup descriptor has had it's POLLIN flag set. If so, we need to actually read the data and then exit the function. The wakeup descriptor will be the first item in the descriptors buffer. */ if ((pPollDescriptors[0].revents & POLLIN) != 0) { ma_uint64 t; int resultRead = read(pPollDescriptors[0].fd, &t, sizeof(t)); /* <-- Important that we read here so that the next write() does not block. */ if (resultRead < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] read() failed.\n"); return ma_result_from_errno(errno); } ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] POLLIN set for wakeupfd\n"); return MA_DEVICE_NOT_STARTED; } /* Getting here means that some data should be able to be read. We need to use ALSA to translate the revents flags for us. */ resultALSA = ((ma_snd_pcm_poll_descriptors_revents_proc)pDevice->pContext->alsa.snd_pcm_poll_descriptors_revents)(pPCM, pPollDescriptors + 1, pollDescriptorCount - 1, &revents); /* +1, -1 to ignore the wakeup descriptor. */ if (resultALSA < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_poll_descriptors_revents() failed.\n"); return ma_result_from_errno(-resultALSA); } if ((revents & POLLERR) != 0) { ma_snd_pcm_state_t state = ((ma_snd_pcm_state_proc)pDevice->pContext->alsa.snd_pcm_state)(pPCM); if (state == MA_SND_PCM_STATE_XRUN) { /* The PCM is in a xrun state. This will be recovered from at a higher level. We can disregard this. */ } else { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[ALSA] POLLERR detected. status = %d\n", ((ma_snd_pcm_state_proc)pDevice->pContext->alsa.snd_pcm_state)(pPCM)); } } if ((revents & requiredEvent) == requiredEvent) { break; /* We're done. Data available for reading or writing. */ } } return MA_SUCCESS; } static ma_result ma_device_wait_read__alsa(ma_device* pDevice) { return ma_device_wait__alsa(pDevice, (ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, (struct pollfd*)pDevice->alsa.pPollDescriptorsCapture, pDevice->alsa.pollDescriptorCountCapture + 1, POLLIN); /* +1 to account for the wakeup descriptor. */ } static ma_result ma_device_wait_write__alsa(ma_device* pDevice) { return ma_device_wait__alsa(pDevice, (ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, (struct pollfd*)pDevice->alsa.pPollDescriptorsPlayback, pDevice->alsa.pollDescriptorCountPlayback + 1, POLLOUT); /* +1 to account for the wakeup descriptor. */ } static ma_result ma_device_read__alsa(ma_device* pDevice, void* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead) { ma_snd_pcm_sframes_t resultALSA = 0; MA_ASSERT(pDevice != NULL); MA_ASSERT(pFramesOut != NULL); if (pFramesRead != NULL) { *pFramesRead = 0; } while (ma_device_get_state(pDevice) == ma_device_state_started) { ma_result result; /* The first thing to do is wait for data to become available for reading. This will return an error code if the device has been stopped. */ result = ma_device_wait_read__alsa(pDevice); if (result != MA_SUCCESS) { return result; } /* Getting here means we should have data available. */ resultALSA = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, pFramesOut, frameCount); if (resultALSA >= 0) { break; /* Success. */ } else { if (resultALSA == -EAGAIN) { /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EGAIN (read)\n");*/ continue; /* Try again. */ } else if (resultALSA == -EPIPE) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EPIPE (read)\n"); /* Overrun. Recover and try again. If this fails we need to return an error. */ resultALSA = ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, resultALSA, MA_TRUE); if (resultALSA < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after overrun."); return ma_result_from_errno((int)-resultALSA); } resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture); if (resultALSA < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun."); return ma_result_from_errno((int)-resultALSA); } continue; /* Try reading again. */ } } } if (pFramesRead != NULL) { *pFramesRead = resultALSA; } return MA_SUCCESS; } static ma_result ma_device_write__alsa(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { ma_snd_pcm_sframes_t resultALSA = 0; MA_ASSERT(pDevice != NULL); MA_ASSERT(pFrames != NULL); if (pFramesWritten != NULL) { *pFramesWritten = 0; } while (ma_device_get_state(pDevice) == ma_device_state_started) { ma_result result; /* The first thing to do is wait for space to become available for writing. This will return an error code if the device has been stopped. */ result = ma_device_wait_write__alsa(pDevice); if (result != MA_SUCCESS) { return result; } resultALSA = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, pFrames, frameCount); if (resultALSA >= 0) { break; /* Success. */ } else { if (resultALSA == -EAGAIN) { /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EGAIN (write)\n");*/ continue; /* Try again. */ } else if (resultALSA == -EPIPE) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EPIPE (write)\n"); /* Underrun. Recover and try again. If this fails we need to return an error. */ resultALSA = ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, resultALSA, MA_TRUE); /* MA_TRUE=silent (don't print anything on error). */ if (resultALSA < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after underrun."); return ma_result_from_errno((int)-resultALSA); } /* In my testing I have had a situation where writei() does not automatically restart the device even though I've set it up as such in the software parameters. What will happen is writei() will block indefinitely even though the number of frames is well beyond the auto-start threshold. To work around this I've needed to add an explicit start here. Not sure if this is me just being stupid and not recovering the device properly, but this definitely feels like something isn't quite right here. */ resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback); if (resultALSA < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun."); return ma_result_from_errno((int)-resultALSA); } continue; /* Try writing again. */ } } } if (pFramesWritten != NULL) { *pFramesWritten = resultALSA; } return MA_SUCCESS; } static ma_result ma_device_data_loop_wakeup__alsa(ma_device* pDevice) { ma_uint64 t = 1; int resultWrite = 0; MA_ASSERT(pDevice != NULL); ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Waking up...\n"); /* Write to an eventfd to trigger a wakeup from poll() and abort any reading or writing. */ if (pDevice->alsa.pPollDescriptorsCapture != NULL) { resultWrite = write(pDevice->alsa.wakeupfdCapture, &t, sizeof(t)); } if (pDevice->alsa.pPollDescriptorsPlayback != NULL) { resultWrite = write(pDevice->alsa.wakeupfdPlayback, &t, sizeof(t)); } if (resultWrite < 0) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] write() failed.\n"); return ma_result_from_errno(errno); } ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Waking up completed successfully.\n"); return MA_SUCCESS; } static ma_result ma_context_uninit__alsa(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_alsa); /* Clean up memory for memory leak checkers. */ ((ma_snd_config_update_free_global_proc)pContext->alsa.snd_config_update_free_global)(); #ifndef MA_NO_RUNTIME_LINKING ma_dlclose(ma_context_get_log(pContext), pContext->alsa.asoundSO); #endif ma_mutex_uninit(&pContext->alsa.internalDeviceEnumLock); return MA_SUCCESS; } static ma_result ma_context_init__alsa(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { ma_result result; #ifndef MA_NO_RUNTIME_LINKING const char* libasoundNames[] = { "libasound.so.2", "libasound.so" }; size_t i; for (i = 0; i < ma_countof(libasoundNames); ++i) { pContext->alsa.asoundSO = ma_dlopen(ma_context_get_log(pContext), libasoundNames[i]); if (pContext->alsa.asoundSO != NULL) { break; } } if (pContext->alsa.asoundSO == NULL) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[ALSA] Failed to open shared object.\n"); return MA_NO_BACKEND; } pContext->alsa.snd_pcm_open = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_open"); pContext->alsa.snd_pcm_close = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_close"); pContext->alsa.snd_pcm_hw_params_sizeof = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_sizeof"); pContext->alsa.snd_pcm_hw_params_any = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_any"); pContext->alsa.snd_pcm_hw_params_set_format = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_set_format"); pContext->alsa.snd_pcm_hw_params_set_format_first = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_set_format_first"); pContext->alsa.snd_pcm_hw_params_get_format_mask = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_get_format_mask"); pContext->alsa.snd_pcm_hw_params_set_channels = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_set_channels"); pContext->alsa.snd_pcm_hw_params_set_channels_near = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_set_channels_near"); pContext->alsa.snd_pcm_hw_params_set_channels_minmax = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_set_channels_minmax"); pContext->alsa.snd_pcm_hw_params_set_rate_resample = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate_resample"); pContext->alsa.snd_pcm_hw_params_set_rate = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate"); pContext->alsa.snd_pcm_hw_params_set_rate_near = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate_near"); pContext->alsa.snd_pcm_hw_params_set_buffer_size_near = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_set_buffer_size_near"); pContext->alsa.snd_pcm_hw_params_set_periods_near = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_set_periods_near"); pContext->alsa.snd_pcm_hw_params_set_access = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_set_access"); pContext->alsa.snd_pcm_hw_params_get_format = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_get_format"); pContext->alsa.snd_pcm_hw_params_get_channels = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels"); pContext->alsa.snd_pcm_hw_params_get_channels_min = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels_min"); pContext->alsa.snd_pcm_hw_params_get_channels_max = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels_max"); pContext->alsa.snd_pcm_hw_params_get_rate = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate"); pContext->alsa.snd_pcm_hw_params_get_rate_min = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate_min"); pContext->alsa.snd_pcm_hw_params_get_rate_max = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate_max"); pContext->alsa.snd_pcm_hw_params_get_buffer_size = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_get_buffer_size"); pContext->alsa.snd_pcm_hw_params_get_periods = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_get_periods"); pContext->alsa.snd_pcm_hw_params_get_access = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_get_access"); pContext->alsa.snd_pcm_hw_params_test_format = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_test_format"); pContext->alsa.snd_pcm_hw_params_test_channels = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_test_channels"); pContext->alsa.snd_pcm_hw_params_test_rate = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params_test_rate"); pContext->alsa.snd_pcm_hw_params = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_hw_params"); pContext->alsa.snd_pcm_sw_params_sizeof = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_sw_params_sizeof"); pContext->alsa.snd_pcm_sw_params_current = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_sw_params_current"); pContext->alsa.snd_pcm_sw_params_get_boundary = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_sw_params_get_boundary"); pContext->alsa.snd_pcm_sw_params_set_avail_min = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_sw_params_set_avail_min"); pContext->alsa.snd_pcm_sw_params_set_start_threshold = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_sw_params_set_start_threshold"); pContext->alsa.snd_pcm_sw_params_set_stop_threshold = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_sw_params_set_stop_threshold"); pContext->alsa.snd_pcm_sw_params = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_sw_params"); pContext->alsa.snd_pcm_format_mask_sizeof = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_format_mask_sizeof"); pContext->alsa.snd_pcm_format_mask_test = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_format_mask_test"); pContext->alsa.snd_pcm_get_chmap = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_get_chmap"); pContext->alsa.snd_pcm_state = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_state"); pContext->alsa.snd_pcm_prepare = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_prepare"); pContext->alsa.snd_pcm_start = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_start"); pContext->alsa.snd_pcm_drop = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_drop"); pContext->alsa.snd_pcm_drain = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_drain"); pContext->alsa.snd_pcm_reset = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_reset"); pContext->alsa.snd_device_name_hint = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_device_name_hint"); pContext->alsa.snd_device_name_get_hint = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_device_name_get_hint"); pContext->alsa.snd_card_get_index = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_card_get_index"); pContext->alsa.snd_device_name_free_hint = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_device_name_free_hint"); pContext->alsa.snd_pcm_mmap_begin = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_mmap_begin"); pContext->alsa.snd_pcm_mmap_commit = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_mmap_commit"); pContext->alsa.snd_pcm_recover = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_recover"); pContext->alsa.snd_pcm_readi = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_readi"); pContext->alsa.snd_pcm_writei = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_writei"); pContext->alsa.snd_pcm_avail = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_avail"); pContext->alsa.snd_pcm_avail_update = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_avail_update"); pContext->alsa.snd_pcm_wait = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_wait"); pContext->alsa.snd_pcm_nonblock = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_nonblock"); pContext->alsa.snd_pcm_info = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_info"); pContext->alsa.snd_pcm_info_sizeof = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_info_sizeof"); pContext->alsa.snd_pcm_info_get_name = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_info_get_name"); pContext->alsa.snd_pcm_poll_descriptors = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_poll_descriptors"); pContext->alsa.snd_pcm_poll_descriptors_count = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_poll_descriptors_count"); pContext->alsa.snd_pcm_poll_descriptors_revents = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_pcm_poll_descriptors_revents"); pContext->alsa.snd_config_update_free_global = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->alsa.asoundSO, "snd_config_update_free_global"); #else /* The system below is just for type safety. */ ma_snd_pcm_open_proc _snd_pcm_open = snd_pcm_open; ma_snd_pcm_close_proc _snd_pcm_close = snd_pcm_close; ma_snd_pcm_hw_params_sizeof_proc _snd_pcm_hw_params_sizeof = snd_pcm_hw_params_sizeof; ma_snd_pcm_hw_params_any_proc _snd_pcm_hw_params_any = snd_pcm_hw_params_any; ma_snd_pcm_hw_params_set_format_proc _snd_pcm_hw_params_set_format = snd_pcm_hw_params_set_format; ma_snd_pcm_hw_params_set_format_first_proc _snd_pcm_hw_params_set_format_first = snd_pcm_hw_params_set_format_first; ma_snd_pcm_hw_params_get_format_mask_proc _snd_pcm_hw_params_get_format_mask = snd_pcm_hw_params_get_format_mask; ma_snd_pcm_hw_params_set_channels_proc _snd_pcm_hw_params_set_channels = snd_pcm_hw_params_set_channels; ma_snd_pcm_hw_params_set_channels_near_proc _snd_pcm_hw_params_set_channels_near = snd_pcm_hw_params_set_channels_near; ma_snd_pcm_hw_params_set_channels_minmax_proc _snd_pcm_hw_params_set_channels_minmax = snd_pcm_hw_params_set_channels_minmax; ma_snd_pcm_hw_params_set_rate_resample_proc _snd_pcm_hw_params_set_rate_resample = snd_pcm_hw_params_set_rate_resample; ma_snd_pcm_hw_params_set_rate_proc _snd_pcm_hw_params_set_rate = snd_pcm_hw_params_set_rate; ma_snd_pcm_hw_params_set_rate_near_proc _snd_pcm_hw_params_set_rate_near = snd_pcm_hw_params_set_rate_near; ma_snd_pcm_hw_params_set_rate_minmax_proc _snd_pcm_hw_params_set_rate_minmax = snd_pcm_hw_params_set_rate_minmax; ma_snd_pcm_hw_params_set_buffer_size_near_proc _snd_pcm_hw_params_set_buffer_size_near = snd_pcm_hw_params_set_buffer_size_near; ma_snd_pcm_hw_params_set_periods_near_proc _snd_pcm_hw_params_set_periods_near = snd_pcm_hw_params_set_periods_near; ma_snd_pcm_hw_params_set_access_proc _snd_pcm_hw_params_set_access = snd_pcm_hw_params_set_access; ma_snd_pcm_hw_params_get_format_proc _snd_pcm_hw_params_get_format = snd_pcm_hw_params_get_format; ma_snd_pcm_hw_params_get_channels_proc _snd_pcm_hw_params_get_channels = snd_pcm_hw_params_get_channels; ma_snd_pcm_hw_params_get_channels_min_proc _snd_pcm_hw_params_get_channels_min = snd_pcm_hw_params_get_channels_min; ma_snd_pcm_hw_params_get_channels_max_proc _snd_pcm_hw_params_get_channels_max = snd_pcm_hw_params_get_channels_max; ma_snd_pcm_hw_params_get_rate_proc _snd_pcm_hw_params_get_rate = snd_pcm_hw_params_get_rate; ma_snd_pcm_hw_params_get_rate_min_proc _snd_pcm_hw_params_get_rate_min = snd_pcm_hw_params_get_rate_min; ma_snd_pcm_hw_params_get_rate_max_proc _snd_pcm_hw_params_get_rate_max = snd_pcm_hw_params_get_rate_max; ma_snd_pcm_hw_params_get_buffer_size_proc _snd_pcm_hw_params_get_buffer_size = snd_pcm_hw_params_get_buffer_size; ma_snd_pcm_hw_params_get_periods_proc _snd_pcm_hw_params_get_periods = snd_pcm_hw_params_get_periods; ma_snd_pcm_hw_params_get_access_proc _snd_pcm_hw_params_get_access = snd_pcm_hw_params_get_access; ma_snd_pcm_hw_params_test_format_proc _snd_pcm_hw_params_test_format = snd_pcm_hw_params_test_format; ma_snd_pcm_hw_params_test_channels_proc _snd_pcm_hw_params_test_channels = snd_pcm_hw_params_test_channels; ma_snd_pcm_hw_params_test_rate_proc _snd_pcm_hw_params_test_rate = snd_pcm_hw_params_test_rate; ma_snd_pcm_hw_params_proc _snd_pcm_hw_params = snd_pcm_hw_params; ma_snd_pcm_sw_params_sizeof_proc _snd_pcm_sw_params_sizeof = snd_pcm_sw_params_sizeof; ma_snd_pcm_sw_params_current_proc _snd_pcm_sw_params_current = snd_pcm_sw_params_current; ma_snd_pcm_sw_params_get_boundary_proc _snd_pcm_sw_params_get_boundary = snd_pcm_sw_params_get_boundary; ma_snd_pcm_sw_params_set_avail_min_proc _snd_pcm_sw_params_set_avail_min = snd_pcm_sw_params_set_avail_min; ma_snd_pcm_sw_params_set_start_threshold_proc _snd_pcm_sw_params_set_start_threshold = snd_pcm_sw_params_set_start_threshold; ma_snd_pcm_sw_params_set_stop_threshold_proc _snd_pcm_sw_params_set_stop_threshold = snd_pcm_sw_params_set_stop_threshold; ma_snd_pcm_sw_params_proc _snd_pcm_sw_params = snd_pcm_sw_params; ma_snd_pcm_format_mask_sizeof_proc _snd_pcm_format_mask_sizeof = snd_pcm_format_mask_sizeof; ma_snd_pcm_format_mask_test_proc _snd_pcm_format_mask_test = snd_pcm_format_mask_test; ma_snd_pcm_get_chmap_proc _snd_pcm_get_chmap = snd_pcm_get_chmap; ma_snd_pcm_state_proc _snd_pcm_state = snd_pcm_state; ma_snd_pcm_prepare_proc _snd_pcm_prepare = snd_pcm_prepare; ma_snd_pcm_start_proc _snd_pcm_start = snd_pcm_start; ma_snd_pcm_drop_proc _snd_pcm_drop = snd_pcm_drop; ma_snd_pcm_drain_proc _snd_pcm_drain = snd_pcm_drain; ma_snd_pcm_reset_proc _snd_pcm_reset = snd_pcm_reset; ma_snd_device_name_hint_proc _snd_device_name_hint = snd_device_name_hint; ma_snd_device_name_get_hint_proc _snd_device_name_get_hint = snd_device_name_get_hint; ma_snd_card_get_index_proc _snd_card_get_index = snd_card_get_index; ma_snd_device_name_free_hint_proc _snd_device_name_free_hint = snd_device_name_free_hint; ma_snd_pcm_mmap_begin_proc _snd_pcm_mmap_begin = snd_pcm_mmap_begin; ma_snd_pcm_mmap_commit_proc _snd_pcm_mmap_commit = snd_pcm_mmap_commit; ma_snd_pcm_recover_proc _snd_pcm_recover = snd_pcm_recover; ma_snd_pcm_readi_proc _snd_pcm_readi = snd_pcm_readi; ma_snd_pcm_writei_proc _snd_pcm_writei = snd_pcm_writei; ma_snd_pcm_avail_proc _snd_pcm_avail = snd_pcm_avail; ma_snd_pcm_avail_update_proc _snd_pcm_avail_update = snd_pcm_avail_update; ma_snd_pcm_wait_proc _snd_pcm_wait = snd_pcm_wait; ma_snd_pcm_nonblock_proc _snd_pcm_nonblock = snd_pcm_nonblock; ma_snd_pcm_info_proc _snd_pcm_info = snd_pcm_info; ma_snd_pcm_info_sizeof_proc _snd_pcm_info_sizeof = snd_pcm_info_sizeof; ma_snd_pcm_info_get_name_proc _snd_pcm_info_get_name = snd_pcm_info_get_name; ma_snd_pcm_poll_descriptors_proc _snd_pcm_poll_descriptors = snd_pcm_poll_descriptors; ma_snd_pcm_poll_descriptors_count_proc _snd_pcm_poll_descriptors_count = snd_pcm_poll_descriptors_count; ma_snd_pcm_poll_descriptors_revents_proc _snd_pcm_poll_descriptors_revents = snd_pcm_poll_descriptors_revents; ma_snd_config_update_free_global_proc _snd_config_update_free_global = snd_config_update_free_global; pContext->alsa.snd_pcm_open = (ma_proc)_snd_pcm_open; pContext->alsa.snd_pcm_close = (ma_proc)_snd_pcm_close; pContext->alsa.snd_pcm_hw_params_sizeof = (ma_proc)_snd_pcm_hw_params_sizeof; pContext->alsa.snd_pcm_hw_params_any = (ma_proc)_snd_pcm_hw_params_any; pContext->alsa.snd_pcm_hw_params_set_format = (ma_proc)_snd_pcm_hw_params_set_format; pContext->alsa.snd_pcm_hw_params_set_format_first = (ma_proc)_snd_pcm_hw_params_set_format_first; pContext->alsa.snd_pcm_hw_params_get_format_mask = (ma_proc)_snd_pcm_hw_params_get_format_mask; pContext->alsa.snd_pcm_hw_params_set_channels = (ma_proc)_snd_pcm_hw_params_set_channels; pContext->alsa.snd_pcm_hw_params_set_channels_near = (ma_proc)_snd_pcm_hw_params_set_channels_near; pContext->alsa.snd_pcm_hw_params_set_channels_minmax = (ma_proc)_snd_pcm_hw_params_set_channels_minmax; pContext->alsa.snd_pcm_hw_params_set_rate_resample = (ma_proc)_snd_pcm_hw_params_set_rate_resample; pContext->alsa.snd_pcm_hw_params_set_rate = (ma_proc)_snd_pcm_hw_params_set_rate; pContext->alsa.snd_pcm_hw_params_set_rate_near = (ma_proc)_snd_pcm_hw_params_set_rate_near; pContext->alsa.snd_pcm_hw_params_set_rate_minmax = (ma_proc)_snd_pcm_hw_params_set_rate_minmax; pContext->alsa.snd_pcm_hw_params_set_buffer_size_near = (ma_proc)_snd_pcm_hw_params_set_buffer_size_near; pContext->alsa.snd_pcm_hw_params_set_periods_near = (ma_proc)_snd_pcm_hw_params_set_periods_near; pContext->alsa.snd_pcm_hw_params_set_access = (ma_proc)_snd_pcm_hw_params_set_access; pContext->alsa.snd_pcm_hw_params_get_format = (ma_proc)_snd_pcm_hw_params_get_format; pContext->alsa.snd_pcm_hw_params_get_channels = (ma_proc)_snd_pcm_hw_params_get_channels; pContext->alsa.snd_pcm_hw_params_get_channels_min = (ma_proc)_snd_pcm_hw_params_get_channels_min; pContext->alsa.snd_pcm_hw_params_get_channels_max = (ma_proc)_snd_pcm_hw_params_get_channels_max; pContext->alsa.snd_pcm_hw_params_get_rate = (ma_proc)_snd_pcm_hw_params_get_rate; pContext->alsa.snd_pcm_hw_params_get_rate_min = (ma_proc)_snd_pcm_hw_params_get_rate_min; pContext->alsa.snd_pcm_hw_params_get_rate_max = (ma_proc)_snd_pcm_hw_params_get_rate_max; pContext->alsa.snd_pcm_hw_params_get_buffer_size = (ma_proc)_snd_pcm_hw_params_get_buffer_size; pContext->alsa.snd_pcm_hw_params_get_periods = (ma_proc)_snd_pcm_hw_params_get_periods; pContext->alsa.snd_pcm_hw_params_get_access = (ma_proc)_snd_pcm_hw_params_get_access; pContext->alsa.snd_pcm_hw_params_test_format = (ma_proc)_snd_pcm_hw_params_test_format; pContext->alsa.snd_pcm_hw_params_test_channels = (ma_proc)_snd_pcm_hw_params_test_channels; pContext->alsa.snd_pcm_hw_params_test_rate = (ma_proc)_snd_pcm_hw_params_test_rate; pContext->alsa.snd_pcm_hw_params = (ma_proc)_snd_pcm_hw_params; pContext->alsa.snd_pcm_sw_params_sizeof = (ma_proc)_snd_pcm_sw_params_sizeof; pContext->alsa.snd_pcm_sw_params_current = (ma_proc)_snd_pcm_sw_params_current; pContext->alsa.snd_pcm_sw_params_get_boundary = (ma_proc)_snd_pcm_sw_params_get_boundary; pContext->alsa.snd_pcm_sw_params_set_avail_min = (ma_proc)_snd_pcm_sw_params_set_avail_min; pContext->alsa.snd_pcm_sw_params_set_start_threshold = (ma_proc)_snd_pcm_sw_params_set_start_threshold; pContext->alsa.snd_pcm_sw_params_set_stop_threshold = (ma_proc)_snd_pcm_sw_params_set_stop_threshold; pContext->alsa.snd_pcm_sw_params = (ma_proc)_snd_pcm_sw_params; pContext->alsa.snd_pcm_format_mask_sizeof = (ma_proc)_snd_pcm_format_mask_sizeof; pContext->alsa.snd_pcm_format_mask_test = (ma_proc)_snd_pcm_format_mask_test; pContext->alsa.snd_pcm_get_chmap = (ma_proc)_snd_pcm_get_chmap; pContext->alsa.snd_pcm_state = (ma_proc)_snd_pcm_state; pContext->alsa.snd_pcm_prepare = (ma_proc)_snd_pcm_prepare; pContext->alsa.snd_pcm_start = (ma_proc)_snd_pcm_start; pContext->alsa.snd_pcm_drop = (ma_proc)_snd_pcm_drop; pContext->alsa.snd_pcm_drain = (ma_proc)_snd_pcm_drain; pContext->alsa.snd_pcm_reset = (ma_proc)_snd_pcm_reset; pContext->alsa.snd_device_name_hint = (ma_proc)_snd_device_name_hint; pContext->alsa.snd_device_name_get_hint = (ma_proc)_snd_device_name_get_hint; pContext->alsa.snd_card_get_index = (ma_proc)_snd_card_get_index; pContext->alsa.snd_device_name_free_hint = (ma_proc)_snd_device_name_free_hint; pContext->alsa.snd_pcm_mmap_begin = (ma_proc)_snd_pcm_mmap_begin; pContext->alsa.snd_pcm_mmap_commit = (ma_proc)_snd_pcm_mmap_commit; pContext->alsa.snd_pcm_recover = (ma_proc)_snd_pcm_recover; pContext->alsa.snd_pcm_readi = (ma_proc)_snd_pcm_readi; pContext->alsa.snd_pcm_writei = (ma_proc)_snd_pcm_writei; pContext->alsa.snd_pcm_avail = (ma_proc)_snd_pcm_avail; pContext->alsa.snd_pcm_avail_update = (ma_proc)_snd_pcm_avail_update; pContext->alsa.snd_pcm_wait = (ma_proc)_snd_pcm_wait; pContext->alsa.snd_pcm_nonblock = (ma_proc)_snd_pcm_nonblock; pContext->alsa.snd_pcm_info = (ma_proc)_snd_pcm_info; pContext->alsa.snd_pcm_info_sizeof = (ma_proc)_snd_pcm_info_sizeof; pContext->alsa.snd_pcm_info_get_name = (ma_proc)_snd_pcm_info_get_name; pContext->alsa.snd_pcm_poll_descriptors = (ma_proc)_snd_pcm_poll_descriptors; pContext->alsa.snd_pcm_poll_descriptors_count = (ma_proc)_snd_pcm_poll_descriptors_count; pContext->alsa.snd_pcm_poll_descriptors_revents = (ma_proc)_snd_pcm_poll_descriptors_revents; pContext->alsa.snd_config_update_free_global = (ma_proc)_snd_config_update_free_global; #endif pContext->alsa.useVerboseDeviceEnumeration = pConfig->alsa.useVerboseDeviceEnumeration; result = ma_mutex_init(&pContext->alsa.internalDeviceEnumLock); if (result != MA_SUCCESS) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[ALSA] WARNING: Failed to initialize mutex for internal device enumeration."); return result; } pCallbacks->onContextInit = ma_context_init__alsa; pCallbacks->onContextUninit = ma_context_uninit__alsa; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__alsa; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__alsa; pCallbacks->onDeviceInit = ma_device_init__alsa; pCallbacks->onDeviceUninit = ma_device_uninit__alsa; pCallbacks->onDeviceStart = ma_device_start__alsa; pCallbacks->onDeviceStop = ma_device_stop__alsa; pCallbacks->onDeviceRead = ma_device_read__alsa; pCallbacks->onDeviceWrite = ma_device_write__alsa; pCallbacks->onDeviceDataLoop = NULL; pCallbacks->onDeviceDataLoopWakeup = ma_device_data_loop_wakeup__alsa; return MA_SUCCESS; } #endif /* MA_HAS_ALSA */ /****************************************************************************** PulseAudio Backend ******************************************************************************/ #ifdef MA_HAS_PULSEAUDIO /* The PulseAudio API, along with Apple's Core Audio, is the worst of the mainstream audio APIs. This is a brief description of what's going on in the PulseAudio backend. I apologize if this gets a bit ranty for your liking - you might want to skip this discussion. PulseAudio has something they call the "Simple API", which unfortunately isn't suitable for miniaudio. I've not seen anywhere where it allows you to enumerate over devices, nor does it seem to support the ability to stop and start streams. Looking at the documentation, it appears as though the stream is constantly running and you prevent sound from being emitted or captured by simply not calling the read or write functions. This is not a professional solution as it would be much better to *actually* stop the underlying stream. Perhaps the simple API has some smarts to do this automatically, but I'm not sure. Another limitation with the simple API is that it seems inefficient when you want to have multiple streams to a single context. For these reasons, miniaudio is not using the simple API. Since we're not using the simple API, we're left with the asynchronous API as our only other option. And boy, is this where it starts to get fun, and I don't mean that in a good way... The problems start with the very name of the API - "asynchronous". Yes, this is an asynchronous oriented API which means your commands don't immediately take effect. You instead need to issue your commands, and then wait for them to complete. The waiting mechanism is enabled through the use of a "main loop". In the asynchronous API you cannot get away from the main loop, and the main loop is where almost all of PulseAudio's problems stem from. When you first initialize PulseAudio you need an object referred to as "main loop". You can implement this yourself by defining your own vtable, but it's much easier to just use one of the built-in main loop implementations. There's two generic implementations called pa_mainloop and pa_threaded_mainloop, and another implementation specific to GLib called pa_glib_mainloop. We're using pa_threaded_mainloop because it simplifies management of the worker thread. The idea of the main loop object is pretty self explanatory - you're supposed to use it to implement a worker thread which runs in a loop. The main loop is where operations are actually executed. To initialize the main loop, you just use `pa_threaded_mainloop_new()`. This is the first function you'll call. You can then get a pointer to the vtable with `pa_threaded_mainloop_get_api()` (the main loop vtable is called `pa_mainloop_api`). Again, you can bypass the threaded main loop object entirely and just implement `pa_mainloop_api` directly, but there's no need for it unless you're doing something extremely specialized such as if you want to integrate it into your application's existing main loop infrastructure. (EDIT 2021-01-26: miniaudio is no longer using `pa_threaded_mainloop` due to this issue: https://github.com/mackron/miniaudio/issues/262. It is now using `pa_mainloop` which turns out to be a simpler solution anyway. The rest of this rant still applies, however.) Once you have your main loop vtable (the `pa_mainloop_api` object) you can create the PulseAudio context. This is very similar to miniaudio's context and they map to each other quite well. You have one context to many streams, which is basically the same as miniaudio's one `ma_context` to many `ma_device`s. Here's where it starts to get annoying, however. When you first create the PulseAudio context, which is done with `pa_context_new()`, it's not actually connected to anything. When you connect, you call `pa_context_connect()`. However, if you remember, PulseAudio is an asynchronous API. That means you cannot just assume the context is connected after `pa_context_context()` has returned. You instead need to wait for it to connect. To do this, you need to either wait for a callback to get fired, which you can set with `pa_context_set_state_callback()`, or you can continuously poll the context's state. Either way, you need to run this in a loop. All objects from here out are created from the context, and, I believe, you can't be creating these objects until the context is connected. This waiting loop is therefore unavoidable. In order for the waiting to ever complete, however, the main loop needs to be running. Before attempting to connect the context, the main loop needs to be started with `pa_threaded_mainloop_start()`. The reason for this asynchronous design is to support cases where you're connecting to a remote server, say through a local network or an internet connection. However, the *VAST* majority of cases don't involve this at all - they just connect to a local "server" running on the host machine. The fact that this would be the default rather than making `pa_context_connect()` synchronous tends to boggle the mind. Once the context has been created and connected you can start creating a stream. A PulseAudio stream is analogous to miniaudio's device. The initialization of a stream is fairly standard - you configure some attributes (analogous to miniaudio's device config) and then call `pa_stream_new()` to actually create it. Here is where we start to get into "operations". When configuring the stream, you can get information about the source (such as sample format, sample rate, etc.), however it's not synchronous. Instead, a `pa_operation` object is returned from `pa_context_get_source_info_by_name()` (capture) or `pa_context_get_sink_info_by_name()` (playback). Then, you need to run a loop (again!) to wait for the operation to complete which you can determine via a callback or polling, just like we did with the context. Then, as an added bonus, you need to decrement the reference counter of the `pa_operation` object to ensure memory is cleaned up. All of that just to retrieve basic information about a device! Once the basic information about the device has been retrieved, miniaudio can now create the stream with `ma_stream_new()`. Like the context, this needs to be connected. But we need to be careful here, because we're now about to introduce one of the most horrific design choices in PulseAudio. PulseAudio allows you to specify a callback that is fired when data can be written to or read from a stream. The language is important here because PulseAudio takes it literally, specifically the "can be". You would think these callbacks would be appropriate as the place for writing and reading data to and from the stream, and that would be right, except when it's not. When you initialize the stream, you can set a flag that tells PulseAudio to not start the stream automatically. This is required because miniaudio does not auto-start devices straight after initialization - you need to call `ma_device_start()` manually. The problem is that even when this flag is specified, PulseAudio will immediately fire its write or read callback. This is *technically* correct (based on the wording in the documentation) because indeed, data *can* be written at this point. The problem is that it's not *practical*. It makes sense that the write/read callback would be where a program will want to write or read data to or from the stream, but when it's called before the application has even requested that the stream be started, it's just not practical because the program probably isn't ready for any kind of data delivery at that point (it may still need to load files or whatnot). Instead, this callback should only be fired when the application requests the stream be started which is how it works with literally *every* other callback-based audio API. Since miniaudio forbids firing of the data callback until the device has been started (as it should be with *all* callback based APIs), logic needs to be added to ensure miniaudio doesn't just blindly fire the application-defined data callback from within the PulseAudio callback before the stream has actually been started. The device state is used for this - if the state is anything other than `ma_device_state_starting` or `ma_device_state_started`, the main data callback is not fired. This, unfortunately, is not the end of the problems with the PulseAudio write callback. Any normal callback based audio API will continuously fire the callback at regular intervals based on the size of the internal buffer. This will only ever be fired when the device is running, and will be fired regardless of whether or not the user actually wrote anything to the device/stream. This not the case in PulseAudio. In PulseAudio, the data callback will *only* be called if you wrote something to it previously. That means, if you don't call `pa_stream_write()`, the callback will not get fired. On the surface you wouldn't think this would matter because you should be always writing data, and if you don't have anything to write, just write silence. That's fine until you want to drain the stream. You see, if you're continuously writing data to the stream, the stream will never get drained! That means in order to drain the stream, you need to *not* write data to it! But remember, when you don't write data to the stream, the callback won't get fired again! Why is draining important? Because that's how we've defined stopping to work in miniaudio. In miniaudio, stopping the device requires it to be drained before returning from ma_device_stop(). So we've stopped the device, which requires us to drain, but draining requires us to *not* write data to the stream (or else it won't ever complete draining), but not writing to the stream means the callback won't get fired again! This becomes a problem when stopping and then restarting the device. When the device is stopped, it's drained, which requires us to *not* write anything to the stream. But then, since we didn't write anything to it, the write callback will *never* get called again if we just resume the stream naively. This means that starting the stream requires us to write data to the stream from outside the callback. This disconnect is something PulseAudio has got seriously wrong - there should only ever be a single source of data delivery, that being the callback. (I have tried using `pa_stream_flush()` to trigger the write callback to fire, but this just doesn't work for some reason.) Once you've created the stream, you need to connect it which involves the whole waiting procedure. This is the same process as the context, only this time you'll poll for the state with `pa_stream_get_status()`. The starting and stopping of a streaming is referred to as "corking" in PulseAudio. The analogy is corking a barrel. To start the stream, you uncork it, to stop it you cork it. Personally I think it's silly - why would you not just call it "starting" and "stopping" like any other normal audio API? Anyway, the act of corking is, you guessed it, asynchronous. This means you'll need our waiting loop as usual. Again, why this asynchronous design is the default is absolutely beyond me. Would it really be that hard to just make it run synchronously? Teardown is pretty simple (what?!). It's just a matter of calling the relevant `_unref()` function on each object in reverse order that they were initialized in. That's about it from the PulseAudio side. A bit ranty, I know, but they really need to fix that main loop and callback system. They're embarrassingly unpractical. The main loop thing is an easy fix - have synchronous versions of all APIs. If an application wants these to run asynchronously, they can execute them in a separate thread themselves. The desire to run these asynchronously is such a niche requirement - it makes no sense to make it the default. The stream write callback needs to be change, or an alternative provided, that is constantly fired, regardless of whether or not `pa_stream_write()` has been called, and it needs to take a pointer to a buffer as a parameter which the program just writes to directly rather than having to call `pa_stream_writable_size()` and `pa_stream_write()`. These changes alone will change PulseAudio from one of the worst audio APIs to one of the best. */ /* It is assumed pulseaudio.h is available when linking at compile time. When linking at compile time, we use the declarations in the header to check for type safety. We cannot do this when linking at run time because the header might not be available. */ #ifdef MA_NO_RUNTIME_LINKING /* pulseaudio.h marks some functions with "inline" which isn't always supported. Need to emulate it. */ #if !defined(__cplusplus) #if defined(__STRICT_ANSI__) #if !defined(inline) #define inline __inline__ __attribute__((always_inline)) #define MA_INLINE_DEFINED #endif #endif #endif #include <pulse/pulseaudio.h> #if defined(MA_INLINE_DEFINED) #undef inline #undef MA_INLINE_DEFINED #endif #define MA_PA_OK PA_OK #define MA_PA_ERR_ACCESS PA_ERR_ACCESS #define MA_PA_ERR_INVALID PA_ERR_INVALID #define MA_PA_ERR_NOENTITY PA_ERR_NOENTITY #define MA_PA_ERR_NOTSUPPORTED PA_ERR_NOTSUPPORTED #define MA_PA_CHANNELS_MAX PA_CHANNELS_MAX #define MA_PA_RATE_MAX PA_RATE_MAX typedef pa_context_flags_t ma_pa_context_flags_t; #define MA_PA_CONTEXT_NOFLAGS PA_CONTEXT_NOFLAGS #define MA_PA_CONTEXT_NOAUTOSPAWN PA_CONTEXT_NOAUTOSPAWN #define MA_PA_CONTEXT_NOFAIL PA_CONTEXT_NOFAIL typedef pa_stream_flags_t ma_pa_stream_flags_t; #define MA_PA_STREAM_NOFLAGS PA_STREAM_NOFLAGS #define MA_PA_STREAM_START_CORKED PA_STREAM_START_CORKED #define MA_PA_STREAM_INTERPOLATE_TIMING PA_STREAM_INTERPOLATE_TIMING #define MA_PA_STREAM_NOT_MONOTONIC PA_STREAM_NOT_MONOTONIC #define MA_PA_STREAM_AUTO_TIMING_UPDATE PA_STREAM_AUTO_TIMING_UPDATE #define MA_PA_STREAM_NO_REMAP_CHANNELS PA_STREAM_NO_REMAP_CHANNELS #define MA_PA_STREAM_NO_REMIX_CHANNELS PA_STREAM_NO_REMIX_CHANNELS #define MA_PA_STREAM_FIX_FORMAT PA_STREAM_FIX_FORMAT #define MA_PA_STREAM_FIX_RATE PA_STREAM_FIX_RATE #define MA_PA_STREAM_FIX_CHANNELS PA_STREAM_FIX_CHANNELS #define MA_PA_STREAM_DONT_MOVE PA_STREAM_DONT_MOVE #define MA_PA_STREAM_VARIABLE_RATE PA_STREAM_VARIABLE_RATE #define MA_PA_STREAM_PEAK_DETECT PA_STREAM_PEAK_DETECT #define MA_PA_STREAM_START_MUTED PA_STREAM_START_MUTED #define MA_PA_STREAM_ADJUST_LATENCY PA_STREAM_ADJUST_LATENCY #define MA_PA_STREAM_EARLY_REQUESTS PA_STREAM_EARLY_REQUESTS #define MA_PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND #define MA_PA_STREAM_START_UNMUTED PA_STREAM_START_UNMUTED #define MA_PA_STREAM_FAIL_ON_SUSPEND PA_STREAM_FAIL_ON_SUSPEND #define MA_PA_STREAM_RELATIVE_VOLUME PA_STREAM_RELATIVE_VOLUME #define MA_PA_STREAM_PASSTHROUGH PA_STREAM_PASSTHROUGH typedef pa_sink_flags_t ma_pa_sink_flags_t; #define MA_PA_SINK_NOFLAGS PA_SINK_NOFLAGS #define MA_PA_SINK_HW_VOLUME_CTRL PA_SINK_HW_VOLUME_CTRL #define MA_PA_SINK_LATENCY PA_SINK_LATENCY #define MA_PA_SINK_HARDWARE PA_SINK_HARDWARE #define MA_PA_SINK_NETWORK PA_SINK_NETWORK #define MA_PA_SINK_HW_MUTE_CTRL PA_SINK_HW_MUTE_CTRL #define MA_PA_SINK_DECIBEL_VOLUME PA_SINK_DECIBEL_VOLUME #define MA_PA_SINK_FLAT_VOLUME PA_SINK_FLAT_VOLUME #define MA_PA_SINK_DYNAMIC_LATENCY PA_SINK_DYNAMIC_LATENCY #define MA_PA_SINK_SET_FORMATS PA_SINK_SET_FORMATS typedef pa_source_flags_t ma_pa_source_flags_t; #define MA_PA_SOURCE_NOFLAGS PA_SOURCE_NOFLAGS #define MA_PA_SOURCE_HW_VOLUME_CTRL PA_SOURCE_HW_VOLUME_CTRL #define MA_PA_SOURCE_LATENCY PA_SOURCE_LATENCY #define MA_PA_SOURCE_HARDWARE PA_SOURCE_HARDWARE #define MA_PA_SOURCE_NETWORK PA_SOURCE_NETWORK #define MA_PA_SOURCE_HW_MUTE_CTRL PA_SOURCE_HW_MUTE_CTRL #define MA_PA_SOURCE_DECIBEL_VOLUME PA_SOURCE_DECIBEL_VOLUME #define MA_PA_SOURCE_DYNAMIC_LATENCY PA_SOURCE_DYNAMIC_LATENCY #define MA_PA_SOURCE_FLAT_VOLUME PA_SOURCE_FLAT_VOLUME typedef pa_context_state_t ma_pa_context_state_t; #define MA_PA_CONTEXT_UNCONNECTED PA_CONTEXT_UNCONNECTED #define MA_PA_CONTEXT_CONNECTING PA_CONTEXT_CONNECTING #define MA_PA_CONTEXT_AUTHORIZING PA_CONTEXT_AUTHORIZING #define MA_PA_CONTEXT_SETTING_NAME PA_CONTEXT_SETTING_NAME #define MA_PA_CONTEXT_READY PA_CONTEXT_READY #define MA_PA_CONTEXT_FAILED PA_CONTEXT_FAILED #define MA_PA_CONTEXT_TERMINATED PA_CONTEXT_TERMINATED typedef pa_stream_state_t ma_pa_stream_state_t; #define MA_PA_STREAM_UNCONNECTED PA_STREAM_UNCONNECTED #define MA_PA_STREAM_CREATING PA_STREAM_CREATING #define MA_PA_STREAM_READY PA_STREAM_READY #define MA_PA_STREAM_FAILED PA_STREAM_FAILED #define MA_PA_STREAM_TERMINATED PA_STREAM_TERMINATED typedef pa_operation_state_t ma_pa_operation_state_t; #define MA_PA_OPERATION_RUNNING PA_OPERATION_RUNNING #define MA_PA_OPERATION_DONE PA_OPERATION_DONE #define MA_PA_OPERATION_CANCELLED PA_OPERATION_CANCELLED typedef pa_sink_state_t ma_pa_sink_state_t; #define MA_PA_SINK_INVALID_STATE PA_SINK_INVALID_STATE #define MA_PA_SINK_RUNNING PA_SINK_RUNNING #define MA_PA_SINK_IDLE PA_SINK_IDLE #define MA_PA_SINK_SUSPENDED PA_SINK_SUSPENDED typedef pa_source_state_t ma_pa_source_state_t; #define MA_PA_SOURCE_INVALID_STATE PA_SOURCE_INVALID_STATE #define MA_PA_SOURCE_RUNNING PA_SOURCE_RUNNING #define MA_PA_SOURCE_IDLE PA_SOURCE_IDLE #define MA_PA_SOURCE_SUSPENDED PA_SOURCE_SUSPENDED typedef pa_seek_mode_t ma_pa_seek_mode_t; #define MA_PA_SEEK_RELATIVE PA_SEEK_RELATIVE #define MA_PA_SEEK_ABSOLUTE PA_SEEK_ABSOLUTE #define MA_PA_SEEK_RELATIVE_ON_READ PA_SEEK_RELATIVE_ON_READ #define MA_PA_SEEK_RELATIVE_END PA_SEEK_RELATIVE_END typedef pa_channel_position_t ma_pa_channel_position_t; #define MA_PA_CHANNEL_POSITION_INVALID PA_CHANNEL_POSITION_INVALID #define MA_PA_CHANNEL_POSITION_MONO PA_CHANNEL_POSITION_MONO #define MA_PA_CHANNEL_POSITION_FRONT_LEFT PA_CHANNEL_POSITION_FRONT_LEFT #define MA_PA_CHANNEL_POSITION_FRONT_RIGHT PA_CHANNEL_POSITION_FRONT_RIGHT #define MA_PA_CHANNEL_POSITION_FRONT_CENTER PA_CHANNEL_POSITION_FRONT_CENTER #define MA_PA_CHANNEL_POSITION_REAR_CENTER PA_CHANNEL_POSITION_REAR_CENTER #define MA_PA_CHANNEL_POSITION_REAR_LEFT PA_CHANNEL_POSITION_REAR_LEFT #define MA_PA_CHANNEL_POSITION_REAR_RIGHT PA_CHANNEL_POSITION_REAR_RIGHT #define MA_PA_CHANNEL_POSITION_LFE PA_CHANNEL_POSITION_LFE #define MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER #define MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER #define MA_PA_CHANNEL_POSITION_SIDE_LEFT PA_CHANNEL_POSITION_SIDE_LEFT #define MA_PA_CHANNEL_POSITION_SIDE_RIGHT PA_CHANNEL_POSITION_SIDE_RIGHT #define MA_PA_CHANNEL_POSITION_AUX0 PA_CHANNEL_POSITION_AUX0 #define MA_PA_CHANNEL_POSITION_AUX1 PA_CHANNEL_POSITION_AUX1 #define MA_PA_CHANNEL_POSITION_AUX2 PA_CHANNEL_POSITION_AUX2 #define MA_PA_CHANNEL_POSITION_AUX3 PA_CHANNEL_POSITION_AUX3 #define MA_PA_CHANNEL_POSITION_AUX4 PA_CHANNEL_POSITION_AUX4 #define MA_PA_CHANNEL_POSITION_AUX5 PA_CHANNEL_POSITION_AUX5 #define MA_PA_CHANNEL_POSITION_AUX6 PA_CHANNEL_POSITION_AUX6 #define MA_PA_CHANNEL_POSITION_AUX7 PA_CHANNEL_POSITION_AUX7 #define MA_PA_CHANNEL_POSITION_AUX8 PA_CHANNEL_POSITION_AUX8 #define MA_PA_CHANNEL_POSITION_AUX9 PA_CHANNEL_POSITION_AUX9 #define MA_PA_CHANNEL_POSITION_AUX10 PA_CHANNEL_POSITION_AUX10 #define MA_PA_CHANNEL_POSITION_AUX11 PA_CHANNEL_POSITION_AUX11 #define MA_PA_CHANNEL_POSITION_AUX12 PA_CHANNEL_POSITION_AUX12 #define MA_PA_CHANNEL_POSITION_AUX13 PA_CHANNEL_POSITION_AUX13 #define MA_PA_CHANNEL_POSITION_AUX14 PA_CHANNEL_POSITION_AUX14 #define MA_PA_CHANNEL_POSITION_AUX15 PA_CHANNEL_POSITION_AUX15 #define MA_PA_CHANNEL_POSITION_AUX16 PA_CHANNEL_POSITION_AUX16 #define MA_PA_CHANNEL_POSITION_AUX17 PA_CHANNEL_POSITION_AUX17 #define MA_PA_CHANNEL_POSITION_AUX18 PA_CHANNEL_POSITION_AUX18 #define MA_PA_CHANNEL_POSITION_AUX19 PA_CHANNEL_POSITION_AUX19 #define MA_PA_CHANNEL_POSITION_AUX20 PA_CHANNEL_POSITION_AUX20 #define MA_PA_CHANNEL_POSITION_AUX21 PA_CHANNEL_POSITION_AUX21 #define MA_PA_CHANNEL_POSITION_AUX22 PA_CHANNEL_POSITION_AUX22 #define MA_PA_CHANNEL_POSITION_AUX23 PA_CHANNEL_POSITION_AUX23 #define MA_PA_CHANNEL_POSITION_AUX24 PA_CHANNEL_POSITION_AUX24 #define MA_PA_CHANNEL_POSITION_AUX25 PA_CHANNEL_POSITION_AUX25 #define MA_PA_CHANNEL_POSITION_AUX26 PA_CHANNEL_POSITION_AUX26 #define MA_PA_CHANNEL_POSITION_AUX27 PA_CHANNEL_POSITION_AUX27 #define MA_PA_CHANNEL_POSITION_AUX28 PA_CHANNEL_POSITION_AUX28 #define MA_PA_CHANNEL_POSITION_AUX29 PA_CHANNEL_POSITION_AUX29 #define MA_PA_CHANNEL_POSITION_AUX30 PA_CHANNEL_POSITION_AUX30 #define MA_PA_CHANNEL_POSITION_AUX31 PA_CHANNEL_POSITION_AUX31 #define MA_PA_CHANNEL_POSITION_TOP_CENTER PA_CHANNEL_POSITION_TOP_CENTER #define MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT PA_CHANNEL_POSITION_TOP_FRONT_LEFT #define MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT PA_CHANNEL_POSITION_TOP_FRONT_RIGHT #define MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER PA_CHANNEL_POSITION_TOP_FRONT_CENTER #define MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT PA_CHANNEL_POSITION_TOP_REAR_LEFT #define MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT PA_CHANNEL_POSITION_TOP_REAR_RIGHT #define MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER PA_CHANNEL_POSITION_TOP_REAR_CENTER #define MA_PA_CHANNEL_POSITION_LEFT PA_CHANNEL_POSITION_LEFT #define MA_PA_CHANNEL_POSITION_RIGHT PA_CHANNEL_POSITION_RIGHT #define MA_PA_CHANNEL_POSITION_CENTER PA_CHANNEL_POSITION_CENTER #define MA_PA_CHANNEL_POSITION_SUBWOOFER PA_CHANNEL_POSITION_SUBWOOFER typedef pa_channel_map_def_t ma_pa_channel_map_def_t; #define MA_PA_CHANNEL_MAP_AIFF PA_CHANNEL_MAP_AIFF #define MA_PA_CHANNEL_MAP_ALSA PA_CHANNEL_MAP_ALSA #define MA_PA_CHANNEL_MAP_AUX PA_CHANNEL_MAP_AUX #define MA_PA_CHANNEL_MAP_WAVEEX PA_CHANNEL_MAP_WAVEEX #define MA_PA_CHANNEL_MAP_OSS PA_CHANNEL_MAP_OSS #define MA_PA_CHANNEL_MAP_DEFAULT PA_CHANNEL_MAP_DEFAULT typedef pa_sample_format_t ma_pa_sample_format_t; #define MA_PA_SAMPLE_INVALID PA_SAMPLE_INVALID #define MA_PA_SAMPLE_U8 PA_SAMPLE_U8 #define MA_PA_SAMPLE_ALAW PA_SAMPLE_ALAW #define MA_PA_SAMPLE_ULAW PA_SAMPLE_ULAW #define MA_PA_SAMPLE_S16LE PA_SAMPLE_S16LE #define MA_PA_SAMPLE_S16BE PA_SAMPLE_S16BE #define MA_PA_SAMPLE_FLOAT32LE PA_SAMPLE_FLOAT32LE #define MA_PA_SAMPLE_FLOAT32BE PA_SAMPLE_FLOAT32BE #define MA_PA_SAMPLE_S32LE PA_SAMPLE_S32LE #define MA_PA_SAMPLE_S32BE PA_SAMPLE_S32BE #define MA_PA_SAMPLE_S24LE PA_SAMPLE_S24LE #define MA_PA_SAMPLE_S24BE PA_SAMPLE_S24BE #define MA_PA_SAMPLE_S24_32LE PA_SAMPLE_S24_32LE #define MA_PA_SAMPLE_S24_32BE PA_SAMPLE_S24_32BE typedef pa_mainloop ma_pa_mainloop; typedef pa_threaded_mainloop ma_pa_threaded_mainloop; typedef pa_mainloop_api ma_pa_mainloop_api; typedef pa_context ma_pa_context; typedef pa_operation ma_pa_operation; typedef pa_stream ma_pa_stream; typedef pa_spawn_api ma_pa_spawn_api; typedef pa_buffer_attr ma_pa_buffer_attr; typedef pa_channel_map ma_pa_channel_map; typedef pa_cvolume ma_pa_cvolume; typedef pa_sample_spec ma_pa_sample_spec; typedef pa_sink_info ma_pa_sink_info; typedef pa_source_info ma_pa_source_info; typedef pa_context_notify_cb_t ma_pa_context_notify_cb_t; typedef pa_sink_info_cb_t ma_pa_sink_info_cb_t; typedef pa_source_info_cb_t ma_pa_source_info_cb_t; typedef pa_stream_success_cb_t ma_pa_stream_success_cb_t; typedef pa_stream_request_cb_t ma_pa_stream_request_cb_t; typedef pa_stream_notify_cb_t ma_pa_stream_notify_cb_t; typedef pa_free_cb_t ma_pa_free_cb_t; #else #define MA_PA_OK 0 #define MA_PA_ERR_ACCESS 1 #define MA_PA_ERR_INVALID 2 #define MA_PA_ERR_NOENTITY 5 #define MA_PA_ERR_NOTSUPPORTED 19 #define MA_PA_CHANNELS_MAX 32 #define MA_PA_RATE_MAX 384000 typedef int ma_pa_context_flags_t; #define MA_PA_CONTEXT_NOFLAGS 0x00000000 #define MA_PA_CONTEXT_NOAUTOSPAWN 0x00000001 #define MA_PA_CONTEXT_NOFAIL 0x00000002 typedef int ma_pa_stream_flags_t; #define MA_PA_STREAM_NOFLAGS 0x00000000 #define MA_PA_STREAM_START_CORKED 0x00000001 #define MA_PA_STREAM_INTERPOLATE_TIMING 0x00000002 #define MA_PA_STREAM_NOT_MONOTONIC 0x00000004 #define MA_PA_STREAM_AUTO_TIMING_UPDATE 0x00000008 #define MA_PA_STREAM_NO_REMAP_CHANNELS 0x00000010 #define MA_PA_STREAM_NO_REMIX_CHANNELS 0x00000020 #define MA_PA_STREAM_FIX_FORMAT 0x00000040 #define MA_PA_STREAM_FIX_RATE 0x00000080 #define MA_PA_STREAM_FIX_CHANNELS 0x00000100 #define MA_PA_STREAM_DONT_MOVE 0x00000200 #define MA_PA_STREAM_VARIABLE_RATE 0x00000400 #define MA_PA_STREAM_PEAK_DETECT 0x00000800 #define MA_PA_STREAM_START_MUTED 0x00001000 #define MA_PA_STREAM_ADJUST_LATENCY 0x00002000 #define MA_PA_STREAM_EARLY_REQUESTS 0x00004000 #define MA_PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND 0x00008000 #define MA_PA_STREAM_START_UNMUTED 0x00010000 #define MA_PA_STREAM_FAIL_ON_SUSPEND 0x00020000 #define MA_PA_STREAM_RELATIVE_VOLUME 0x00040000 #define MA_PA_STREAM_PASSTHROUGH 0x00080000 typedef int ma_pa_sink_flags_t; #define MA_PA_SINK_NOFLAGS 0x00000000 #define MA_PA_SINK_HW_VOLUME_CTRL 0x00000001 #define MA_PA_SINK_LATENCY 0x00000002 #define MA_PA_SINK_HARDWARE 0x00000004 #define MA_PA_SINK_NETWORK 0x00000008 #define MA_PA_SINK_HW_MUTE_CTRL 0x00000010 #define MA_PA_SINK_DECIBEL_VOLUME 0x00000020 #define MA_PA_SINK_FLAT_VOLUME 0x00000040 #define MA_PA_SINK_DYNAMIC_LATENCY 0x00000080 #define MA_PA_SINK_SET_FORMATS 0x00000100 typedef int ma_pa_source_flags_t; #define MA_PA_SOURCE_NOFLAGS 0x00000000 #define MA_PA_SOURCE_HW_VOLUME_CTRL 0x00000001 #define MA_PA_SOURCE_LATENCY 0x00000002 #define MA_PA_SOURCE_HARDWARE 0x00000004 #define MA_PA_SOURCE_NETWORK 0x00000008 #define MA_PA_SOURCE_HW_MUTE_CTRL 0x00000010 #define MA_PA_SOURCE_DECIBEL_VOLUME 0x00000020 #define MA_PA_SOURCE_DYNAMIC_LATENCY 0x00000040 #define MA_PA_SOURCE_FLAT_VOLUME 0x00000080 typedef int ma_pa_context_state_t; #define MA_PA_CONTEXT_UNCONNECTED 0 #define MA_PA_CONTEXT_CONNECTING 1 #define MA_PA_CONTEXT_AUTHORIZING 2 #define MA_PA_CONTEXT_SETTING_NAME 3 #define MA_PA_CONTEXT_READY 4 #define MA_PA_CONTEXT_FAILED 5 #define MA_PA_CONTEXT_TERMINATED 6 typedef int ma_pa_stream_state_t; #define MA_PA_STREAM_UNCONNECTED 0 #define MA_PA_STREAM_CREATING 1 #define MA_PA_STREAM_READY 2 #define MA_PA_STREAM_FAILED 3 #define MA_PA_STREAM_TERMINATED 4 typedef int ma_pa_operation_state_t; #define MA_PA_OPERATION_RUNNING 0 #define MA_PA_OPERATION_DONE 1 #define MA_PA_OPERATION_CANCELLED 2 typedef int ma_pa_sink_state_t; #define MA_PA_SINK_INVALID_STATE -1 #define MA_PA_SINK_RUNNING 0 #define MA_PA_SINK_IDLE 1 #define MA_PA_SINK_SUSPENDED 2 typedef int ma_pa_source_state_t; #define MA_PA_SOURCE_INVALID_STATE -1 #define MA_PA_SOURCE_RUNNING 0 #define MA_PA_SOURCE_IDLE 1 #define MA_PA_SOURCE_SUSPENDED 2 typedef int ma_pa_seek_mode_t; #define MA_PA_SEEK_RELATIVE 0 #define MA_PA_SEEK_ABSOLUTE 1 #define MA_PA_SEEK_RELATIVE_ON_READ 2 #define MA_PA_SEEK_RELATIVE_END 3 typedef int ma_pa_channel_position_t; #define MA_PA_CHANNEL_POSITION_INVALID -1 #define MA_PA_CHANNEL_POSITION_MONO 0 #define MA_PA_CHANNEL_POSITION_FRONT_LEFT 1 #define MA_PA_CHANNEL_POSITION_FRONT_RIGHT 2 #define MA_PA_CHANNEL_POSITION_FRONT_CENTER 3 #define MA_PA_CHANNEL_POSITION_REAR_CENTER 4 #define MA_PA_CHANNEL_POSITION_REAR_LEFT 5 #define MA_PA_CHANNEL_POSITION_REAR_RIGHT 6 #define MA_PA_CHANNEL_POSITION_LFE 7 #define MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER 8 #define MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER 9 #define MA_PA_CHANNEL_POSITION_SIDE_LEFT 10 #define MA_PA_CHANNEL_POSITION_SIDE_RIGHT 11 #define MA_PA_CHANNEL_POSITION_AUX0 12 #define MA_PA_CHANNEL_POSITION_AUX1 13 #define MA_PA_CHANNEL_POSITION_AUX2 14 #define MA_PA_CHANNEL_POSITION_AUX3 15 #define MA_PA_CHANNEL_POSITION_AUX4 16 #define MA_PA_CHANNEL_POSITION_AUX5 17 #define MA_PA_CHANNEL_POSITION_AUX6 18 #define MA_PA_CHANNEL_POSITION_AUX7 19 #define MA_PA_CHANNEL_POSITION_AUX8 20 #define MA_PA_CHANNEL_POSITION_AUX9 21 #define MA_PA_CHANNEL_POSITION_AUX10 22 #define MA_PA_CHANNEL_POSITION_AUX11 23 #define MA_PA_CHANNEL_POSITION_AUX12 24 #define MA_PA_CHANNEL_POSITION_AUX13 25 #define MA_PA_CHANNEL_POSITION_AUX14 26 #define MA_PA_CHANNEL_POSITION_AUX15 27 #define MA_PA_CHANNEL_POSITION_AUX16 28 #define MA_PA_CHANNEL_POSITION_AUX17 29 #define MA_PA_CHANNEL_POSITION_AUX18 30 #define MA_PA_CHANNEL_POSITION_AUX19 31 #define MA_PA_CHANNEL_POSITION_AUX20 32 #define MA_PA_CHANNEL_POSITION_AUX21 33 #define MA_PA_CHANNEL_POSITION_AUX22 34 #define MA_PA_CHANNEL_POSITION_AUX23 35 #define MA_PA_CHANNEL_POSITION_AUX24 36 #define MA_PA_CHANNEL_POSITION_AUX25 37 #define MA_PA_CHANNEL_POSITION_AUX26 38 #define MA_PA_CHANNEL_POSITION_AUX27 39 #define MA_PA_CHANNEL_POSITION_AUX28 40 #define MA_PA_CHANNEL_POSITION_AUX29 41 #define MA_PA_CHANNEL_POSITION_AUX30 42 #define MA_PA_CHANNEL_POSITION_AUX31 43 #define MA_PA_CHANNEL_POSITION_TOP_CENTER 44 #define MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT 45 #define MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT 46 #define MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER 47 #define MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT 48 #define MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT 49 #define MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER 50 #define MA_PA_CHANNEL_POSITION_LEFT MA_PA_CHANNEL_POSITION_FRONT_LEFT #define MA_PA_CHANNEL_POSITION_RIGHT MA_PA_CHANNEL_POSITION_FRONT_RIGHT #define MA_PA_CHANNEL_POSITION_CENTER MA_PA_CHANNEL_POSITION_FRONT_CENTER #define MA_PA_CHANNEL_POSITION_SUBWOOFER MA_PA_CHANNEL_POSITION_LFE typedef int ma_pa_channel_map_def_t; #define MA_PA_CHANNEL_MAP_AIFF 0 #define MA_PA_CHANNEL_MAP_ALSA 1 #define MA_PA_CHANNEL_MAP_AUX 2 #define MA_PA_CHANNEL_MAP_WAVEEX 3 #define MA_PA_CHANNEL_MAP_OSS 4 #define MA_PA_CHANNEL_MAP_DEFAULT MA_PA_CHANNEL_MAP_AIFF typedef int ma_pa_sample_format_t; #define MA_PA_SAMPLE_INVALID -1 #define MA_PA_SAMPLE_U8 0 #define MA_PA_SAMPLE_ALAW 1 #define MA_PA_SAMPLE_ULAW 2 #define MA_PA_SAMPLE_S16LE 3 #define MA_PA_SAMPLE_S16BE 4 #define MA_PA_SAMPLE_FLOAT32LE 5 #define MA_PA_SAMPLE_FLOAT32BE 6 #define MA_PA_SAMPLE_S32LE 7 #define MA_PA_SAMPLE_S32BE 8 #define MA_PA_SAMPLE_S24LE 9 #define MA_PA_SAMPLE_S24BE 10 #define MA_PA_SAMPLE_S24_32LE 11 #define MA_PA_SAMPLE_S24_32BE 12 typedef struct ma_pa_mainloop ma_pa_mainloop; typedef struct ma_pa_threaded_mainloop ma_pa_threaded_mainloop; typedef struct ma_pa_mainloop_api ma_pa_mainloop_api; typedef struct ma_pa_context ma_pa_context; typedef struct ma_pa_operation ma_pa_operation; typedef struct ma_pa_stream ma_pa_stream; typedef struct ma_pa_spawn_api ma_pa_spawn_api; typedef struct { ma_uint32 maxlength; ma_uint32 tlength; ma_uint32 prebuf; ma_uint32 minreq; ma_uint32 fragsize; } ma_pa_buffer_attr; typedef struct { ma_uint8 channels; ma_pa_channel_position_t map[MA_PA_CHANNELS_MAX]; } ma_pa_channel_map; typedef struct { ma_uint8 channels; ma_uint32 values[MA_PA_CHANNELS_MAX]; } ma_pa_cvolume; typedef struct { ma_pa_sample_format_t format; ma_uint32 rate; ma_uint8 channels; } ma_pa_sample_spec; typedef struct { const char* name; ma_uint32 index; const char* description; ma_pa_sample_spec sample_spec; ma_pa_channel_map channel_map; ma_uint32 owner_module; ma_pa_cvolume volume; int mute; ma_uint32 monitor_source; const char* monitor_source_name; ma_uint64 latency; const char* driver; ma_pa_sink_flags_t flags; void* proplist; ma_uint64 configured_latency; ma_uint32 base_volume; ma_pa_sink_state_t state; ma_uint32 n_volume_steps; ma_uint32 card; ma_uint32 n_ports; void** ports; void* active_port; ma_uint8 n_formats; void** formats; } ma_pa_sink_info; typedef struct { const char *name; ma_uint32 index; const char *description; ma_pa_sample_spec sample_spec; ma_pa_channel_map channel_map; ma_uint32 owner_module; ma_pa_cvolume volume; int mute; ma_uint32 monitor_of_sink; const char *monitor_of_sink_name; ma_uint64 latency; const char *driver; ma_pa_source_flags_t flags; void* proplist; ma_uint64 configured_latency; ma_uint32 base_volume; ma_pa_source_state_t state; ma_uint32 n_volume_steps; ma_uint32 card; ma_uint32 n_ports; void** ports; void* active_port; ma_uint8 n_formats; void** formats; } ma_pa_source_info; typedef void (* ma_pa_context_notify_cb_t)(ma_pa_context* c, void* userdata); typedef void (* ma_pa_sink_info_cb_t) (ma_pa_context* c, const ma_pa_sink_info* i, int eol, void* userdata); typedef void (* ma_pa_source_info_cb_t) (ma_pa_context* c, const ma_pa_source_info* i, int eol, void* userdata); typedef void (* ma_pa_stream_success_cb_t)(ma_pa_stream* s, int success, void* userdata); typedef void (* ma_pa_stream_request_cb_t)(ma_pa_stream* s, size_t nbytes, void* userdata); typedef void (* ma_pa_stream_notify_cb_t) (ma_pa_stream* s, void* userdata); typedef void (* ma_pa_free_cb_t) (void* p); #endif typedef ma_pa_mainloop* (* ma_pa_mainloop_new_proc) (void); typedef void (* ma_pa_mainloop_free_proc) (ma_pa_mainloop* m); typedef void (* ma_pa_mainloop_quit_proc) (ma_pa_mainloop* m, int retval); typedef ma_pa_mainloop_api* (* ma_pa_mainloop_get_api_proc) (ma_pa_mainloop* m); typedef int (* ma_pa_mainloop_iterate_proc) (ma_pa_mainloop* m, int block, int* retval); typedef void (* ma_pa_mainloop_wakeup_proc) (ma_pa_mainloop* m); typedef ma_pa_threaded_mainloop* (* ma_pa_threaded_mainloop_new_proc) (void); typedef void (* ma_pa_threaded_mainloop_free_proc) (ma_pa_threaded_mainloop* m); typedef int (* ma_pa_threaded_mainloop_start_proc) (ma_pa_threaded_mainloop* m); typedef void (* ma_pa_threaded_mainloop_stop_proc) (ma_pa_threaded_mainloop* m); typedef void (* ma_pa_threaded_mainloop_lock_proc) (ma_pa_threaded_mainloop* m); typedef void (* ma_pa_threaded_mainloop_unlock_proc) (ma_pa_threaded_mainloop* m); typedef void (* ma_pa_threaded_mainloop_wait_proc) (ma_pa_threaded_mainloop* m); typedef void (* ma_pa_threaded_mainloop_signal_proc) (ma_pa_threaded_mainloop* m, int wait_for_accept); typedef void (* ma_pa_threaded_mainloop_accept_proc) (ma_pa_threaded_mainloop* m); typedef int (* ma_pa_threaded_mainloop_get_retval_proc) (const ma_pa_threaded_mainloop* m); typedef ma_pa_mainloop_api* (* ma_pa_threaded_mainloop_get_api_proc) (ma_pa_threaded_mainloop* m); typedef int (* ma_pa_threaded_mainloop_in_thread_proc) (ma_pa_threaded_mainloop* m); typedef void (* ma_pa_threaded_mainloop_set_name_proc) (ma_pa_threaded_mainloop* m, const char* name); typedef ma_pa_context* (* ma_pa_context_new_proc) (ma_pa_mainloop_api* mainloop, const char* name); typedef void (* ma_pa_context_unref_proc) (ma_pa_context* c); typedef int (* ma_pa_context_connect_proc) (ma_pa_context* c, const char* server, ma_pa_context_flags_t flags, const ma_pa_spawn_api* api); typedef void (* ma_pa_context_disconnect_proc) (ma_pa_context* c); typedef void (* ma_pa_context_set_state_callback_proc) (ma_pa_context* c, ma_pa_context_notify_cb_t cb, void* userdata); typedef ma_pa_context_state_t (* ma_pa_context_get_state_proc) (const ma_pa_context* c); typedef ma_pa_operation* (* ma_pa_context_get_sink_info_list_proc) (ma_pa_context* c, ma_pa_sink_info_cb_t cb, void* userdata); typedef ma_pa_operation* (* ma_pa_context_get_source_info_list_proc) (ma_pa_context* c, ma_pa_source_info_cb_t cb, void* userdata); typedef ma_pa_operation* (* ma_pa_context_get_sink_info_by_name_proc) (ma_pa_context* c, const char* name, ma_pa_sink_info_cb_t cb, void* userdata); typedef ma_pa_operation* (* ma_pa_context_get_source_info_by_name_proc)(ma_pa_context* c, const char* name, ma_pa_source_info_cb_t cb, void* userdata); typedef void (* ma_pa_operation_unref_proc) (ma_pa_operation* o); typedef ma_pa_operation_state_t (* ma_pa_operation_get_state_proc) (const ma_pa_operation* o); typedef ma_pa_channel_map* (* ma_pa_channel_map_init_extend_proc) (ma_pa_channel_map* m, unsigned channels, ma_pa_channel_map_def_t def); typedef int (* ma_pa_channel_map_valid_proc) (const ma_pa_channel_map* m); typedef int (* ma_pa_channel_map_compatible_proc) (const ma_pa_channel_map* m, const ma_pa_sample_spec* ss); typedef ma_pa_stream* (* ma_pa_stream_new_proc) (ma_pa_context* c, const char* name, const ma_pa_sample_spec* ss, const ma_pa_channel_map* map); typedef void (* ma_pa_stream_unref_proc) (ma_pa_stream* s); typedef int (* ma_pa_stream_connect_playback_proc) (ma_pa_stream* s, const char* dev, const ma_pa_buffer_attr* attr, ma_pa_stream_flags_t flags, const ma_pa_cvolume* volume, ma_pa_stream* sync_stream); typedef int (* ma_pa_stream_connect_record_proc) (ma_pa_stream* s, const char* dev, const ma_pa_buffer_attr* attr, ma_pa_stream_flags_t flags); typedef int (* ma_pa_stream_disconnect_proc) (ma_pa_stream* s); typedef ma_pa_stream_state_t (* ma_pa_stream_get_state_proc) (const ma_pa_stream* s); typedef const ma_pa_sample_spec* (* ma_pa_stream_get_sample_spec_proc) (ma_pa_stream* s); typedef const ma_pa_channel_map* (* ma_pa_stream_get_channel_map_proc) (ma_pa_stream* s); typedef const ma_pa_buffer_attr* (* ma_pa_stream_get_buffer_attr_proc) (ma_pa_stream* s); typedef ma_pa_operation* (* ma_pa_stream_set_buffer_attr_proc) (ma_pa_stream* s, const ma_pa_buffer_attr* attr, ma_pa_stream_success_cb_t cb, void* userdata); typedef const char* (* ma_pa_stream_get_device_name_proc) (const ma_pa_stream* s); typedef void (* ma_pa_stream_set_write_callback_proc) (ma_pa_stream* s, ma_pa_stream_request_cb_t cb, void* userdata); typedef void (* ma_pa_stream_set_read_callback_proc) (ma_pa_stream* s, ma_pa_stream_request_cb_t cb, void* userdata); typedef void (* ma_pa_stream_set_suspended_callback_proc) (ma_pa_stream* s, ma_pa_stream_notify_cb_t cb, void* userdata); typedef void (* ma_pa_stream_set_moved_callback_proc) (ma_pa_stream* s, ma_pa_stream_notify_cb_t cb, void* userdata); typedef int (* ma_pa_stream_is_suspended_proc) (const ma_pa_stream* s); typedef ma_pa_operation* (* ma_pa_stream_flush_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata); typedef ma_pa_operation* (* ma_pa_stream_drain_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata); typedef int (* ma_pa_stream_is_corked_proc) (const ma_pa_stream* s); typedef ma_pa_operation* (* ma_pa_stream_cork_proc) (ma_pa_stream* s, int b, ma_pa_stream_success_cb_t cb, void* userdata); typedef ma_pa_operation* (* ma_pa_stream_trigger_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata); typedef int (* ma_pa_stream_begin_write_proc) (ma_pa_stream* s, void** data, size_t* nbytes); typedef int (* ma_pa_stream_write_proc) (ma_pa_stream* s, const void* data, size_t nbytes, ma_pa_free_cb_t free_cb, int64_t offset, ma_pa_seek_mode_t seek); typedef int (* ma_pa_stream_peek_proc) (ma_pa_stream* s, const void** data, size_t* nbytes); typedef int (* ma_pa_stream_drop_proc) (ma_pa_stream* s); typedef size_t (* ma_pa_stream_writable_size_proc) (const ma_pa_stream* s); typedef size_t (* ma_pa_stream_readable_size_proc) (const ma_pa_stream* s); typedef struct { ma_uint32 count; ma_uint32 capacity; ma_device_info* pInfo; } ma_pulse_device_enum_data; static ma_result ma_result_from_pulse(int result) { if (result < 0) { return MA_ERROR; } switch (result) { case MA_PA_OK: return MA_SUCCESS; case MA_PA_ERR_ACCESS: return MA_ACCESS_DENIED; case MA_PA_ERR_INVALID: return MA_INVALID_ARGS; case MA_PA_ERR_NOENTITY: return MA_NO_DEVICE; default: return MA_ERROR; } } #if 0 static ma_pa_sample_format_t ma_format_to_pulse(ma_format format) { if (ma_is_little_endian()) { switch (format) { case ma_format_s16: return MA_PA_SAMPLE_S16LE; case ma_format_s24: return MA_PA_SAMPLE_S24LE; case ma_format_s32: return MA_PA_SAMPLE_S32LE; case ma_format_f32: return MA_PA_SAMPLE_FLOAT32LE; default: break; } } else { switch (format) { case ma_format_s16: return MA_PA_SAMPLE_S16BE; case ma_format_s24: return MA_PA_SAMPLE_S24BE; case ma_format_s32: return MA_PA_SAMPLE_S32BE; case ma_format_f32: return MA_PA_SAMPLE_FLOAT32BE; default: break; } } /* Endian agnostic. */ switch (format) { case ma_format_u8: return MA_PA_SAMPLE_U8; default: return MA_PA_SAMPLE_INVALID; } } #endif static ma_format ma_format_from_pulse(ma_pa_sample_format_t format) { if (ma_is_little_endian()) { switch (format) { case MA_PA_SAMPLE_S16LE: return ma_format_s16; case MA_PA_SAMPLE_S24LE: return ma_format_s24; case MA_PA_SAMPLE_S32LE: return ma_format_s32; case MA_PA_SAMPLE_FLOAT32LE: return ma_format_f32; default: break; } } else { switch (format) { case MA_PA_SAMPLE_S16BE: return ma_format_s16; case MA_PA_SAMPLE_S24BE: return ma_format_s24; case MA_PA_SAMPLE_S32BE: return ma_format_s32; case MA_PA_SAMPLE_FLOAT32BE: return ma_format_f32; default: break; } } /* Endian agnostic. */ switch (format) { case MA_PA_SAMPLE_U8: return ma_format_u8; default: return ma_format_unknown; } } static ma_channel ma_channel_position_from_pulse(ma_pa_channel_position_t position) { switch (position) { case MA_PA_CHANNEL_POSITION_INVALID: return MA_CHANNEL_NONE; case MA_PA_CHANNEL_POSITION_MONO: return MA_CHANNEL_MONO; case MA_PA_CHANNEL_POSITION_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT; case MA_PA_CHANNEL_POSITION_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT; case MA_PA_CHANNEL_POSITION_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER; case MA_PA_CHANNEL_POSITION_REAR_CENTER: return MA_CHANNEL_BACK_CENTER; case MA_PA_CHANNEL_POSITION_REAR_LEFT: return MA_CHANNEL_BACK_LEFT; case MA_PA_CHANNEL_POSITION_REAR_RIGHT: return MA_CHANNEL_BACK_RIGHT; case MA_PA_CHANNEL_POSITION_LFE: return MA_CHANNEL_LFE; case MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER; case MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER; case MA_PA_CHANNEL_POSITION_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT; case MA_PA_CHANNEL_POSITION_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT; case MA_PA_CHANNEL_POSITION_AUX0: return MA_CHANNEL_AUX_0; case MA_PA_CHANNEL_POSITION_AUX1: return MA_CHANNEL_AUX_1; case MA_PA_CHANNEL_POSITION_AUX2: return MA_CHANNEL_AUX_2; case MA_PA_CHANNEL_POSITION_AUX3: return MA_CHANNEL_AUX_3; case MA_PA_CHANNEL_POSITION_AUX4: return MA_CHANNEL_AUX_4; case MA_PA_CHANNEL_POSITION_AUX5: return MA_CHANNEL_AUX_5; case MA_PA_CHANNEL_POSITION_AUX6: return MA_CHANNEL_AUX_6; case MA_PA_CHANNEL_POSITION_AUX7: return MA_CHANNEL_AUX_7; case MA_PA_CHANNEL_POSITION_AUX8: return MA_CHANNEL_AUX_8; case MA_PA_CHANNEL_POSITION_AUX9: return MA_CHANNEL_AUX_9; case MA_PA_CHANNEL_POSITION_AUX10: return MA_CHANNEL_AUX_10; case MA_PA_CHANNEL_POSITION_AUX11: return MA_CHANNEL_AUX_11; case MA_PA_CHANNEL_POSITION_AUX12: return MA_CHANNEL_AUX_12; case MA_PA_CHANNEL_POSITION_AUX13: return MA_CHANNEL_AUX_13; case MA_PA_CHANNEL_POSITION_AUX14: return MA_CHANNEL_AUX_14; case MA_PA_CHANNEL_POSITION_AUX15: return MA_CHANNEL_AUX_15; case MA_PA_CHANNEL_POSITION_AUX16: return MA_CHANNEL_AUX_16; case MA_PA_CHANNEL_POSITION_AUX17: return MA_CHANNEL_AUX_17; case MA_PA_CHANNEL_POSITION_AUX18: return MA_CHANNEL_AUX_18; case MA_PA_CHANNEL_POSITION_AUX19: return MA_CHANNEL_AUX_19; case MA_PA_CHANNEL_POSITION_AUX20: return MA_CHANNEL_AUX_20; case MA_PA_CHANNEL_POSITION_AUX21: return MA_CHANNEL_AUX_21; case MA_PA_CHANNEL_POSITION_AUX22: return MA_CHANNEL_AUX_22; case MA_PA_CHANNEL_POSITION_AUX23: return MA_CHANNEL_AUX_23; case MA_PA_CHANNEL_POSITION_AUX24: return MA_CHANNEL_AUX_24; case MA_PA_CHANNEL_POSITION_AUX25: return MA_CHANNEL_AUX_25; case MA_PA_CHANNEL_POSITION_AUX26: return MA_CHANNEL_AUX_26; case MA_PA_CHANNEL_POSITION_AUX27: return MA_CHANNEL_AUX_27; case MA_PA_CHANNEL_POSITION_AUX28: return MA_CHANNEL_AUX_28; case MA_PA_CHANNEL_POSITION_AUX29: return MA_CHANNEL_AUX_29; case MA_PA_CHANNEL_POSITION_AUX30: return MA_CHANNEL_AUX_30; case MA_PA_CHANNEL_POSITION_AUX31: return MA_CHANNEL_AUX_31; case MA_PA_CHANNEL_POSITION_TOP_CENTER: return MA_CHANNEL_TOP_CENTER; case MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT; case MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT; case MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER; case MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT: return MA_CHANNEL_TOP_BACK_LEFT; case MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT; case MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER: return MA_CHANNEL_TOP_BACK_CENTER; default: return MA_CHANNEL_NONE; } } #if 0 static ma_pa_channel_position_t ma_channel_position_to_pulse(ma_channel position) { switch (position) { case MA_CHANNEL_NONE: return MA_PA_CHANNEL_POSITION_INVALID; case MA_CHANNEL_FRONT_LEFT: return MA_PA_CHANNEL_POSITION_FRONT_LEFT; case MA_CHANNEL_FRONT_RIGHT: return MA_PA_CHANNEL_POSITION_FRONT_RIGHT; case MA_CHANNEL_FRONT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_CENTER; case MA_CHANNEL_LFE: return MA_PA_CHANNEL_POSITION_LFE; case MA_CHANNEL_BACK_LEFT: return MA_PA_CHANNEL_POSITION_REAR_LEFT; case MA_CHANNEL_BACK_RIGHT: return MA_PA_CHANNEL_POSITION_REAR_RIGHT; case MA_CHANNEL_FRONT_LEFT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER; case MA_CHANNEL_FRONT_RIGHT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER; case MA_CHANNEL_BACK_CENTER: return MA_PA_CHANNEL_POSITION_REAR_CENTER; case MA_CHANNEL_SIDE_LEFT: return MA_PA_CHANNEL_POSITION_SIDE_LEFT; case MA_CHANNEL_SIDE_RIGHT: return MA_PA_CHANNEL_POSITION_SIDE_RIGHT; case MA_CHANNEL_TOP_CENTER: return MA_PA_CHANNEL_POSITION_TOP_CENTER; case MA_CHANNEL_TOP_FRONT_LEFT: return MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT; case MA_CHANNEL_TOP_FRONT_CENTER: return MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER; case MA_CHANNEL_TOP_FRONT_RIGHT: return MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT; case MA_CHANNEL_TOP_BACK_LEFT: return MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT; case MA_CHANNEL_TOP_BACK_CENTER: return MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER; case MA_CHANNEL_TOP_BACK_RIGHT: return MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT; case MA_CHANNEL_19: return MA_PA_CHANNEL_POSITION_AUX18; case MA_CHANNEL_20: return MA_PA_CHANNEL_POSITION_AUX19; case MA_CHANNEL_21: return MA_PA_CHANNEL_POSITION_AUX20; case MA_CHANNEL_22: return MA_PA_CHANNEL_POSITION_AUX21; case MA_CHANNEL_23: return MA_PA_CHANNEL_POSITION_AUX22; case MA_CHANNEL_24: return MA_PA_CHANNEL_POSITION_AUX23; case MA_CHANNEL_25: return MA_PA_CHANNEL_POSITION_AUX24; case MA_CHANNEL_26: return MA_PA_CHANNEL_POSITION_AUX25; case MA_CHANNEL_27: return MA_PA_CHANNEL_POSITION_AUX26; case MA_CHANNEL_28: return MA_PA_CHANNEL_POSITION_AUX27; case MA_CHANNEL_29: return MA_PA_CHANNEL_POSITION_AUX28; case MA_CHANNEL_30: return MA_PA_CHANNEL_POSITION_AUX29; case MA_CHANNEL_31: return MA_PA_CHANNEL_POSITION_AUX30; case MA_CHANNEL_32: return MA_PA_CHANNEL_POSITION_AUX31; default: return (ma_pa_channel_position_t)position; } } #endif static ma_result ma_wait_for_operation__pulse(ma_context* pContext, ma_ptr pMainLoop, ma_pa_operation* pOP) { int resultPA; ma_pa_operation_state_t state; MA_ASSERT(pContext != NULL); MA_ASSERT(pOP != NULL); for (;;) { state = ((ma_pa_operation_get_state_proc)pContext->pulse.pa_operation_get_state)(pOP); if (state != MA_PA_OPERATION_RUNNING) { break; /* Done. */ } resultPA = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pMainLoop, 1, NULL); if (resultPA < 0) { return ma_result_from_pulse(resultPA); } } return MA_SUCCESS; } static ma_result ma_wait_for_operation_and_unref__pulse(ma_context* pContext, ma_ptr pMainLoop, ma_pa_operation* pOP) { ma_result result; if (pOP == NULL) { return MA_INVALID_ARGS; } result = ma_wait_for_operation__pulse(pContext, pMainLoop, pOP); ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); return result; } static ma_result ma_wait_for_pa_context_to_connect__pulse(ma_context* pContext, ma_ptr pMainLoop, ma_ptr pPulseContext) { int resultPA; ma_pa_context_state_t state; for (;;) { state = ((ma_pa_context_get_state_proc)pContext->pulse.pa_context_get_state)((ma_pa_context*)pPulseContext); if (state == MA_PA_CONTEXT_READY) { break; /* Done. */ } if (state == MA_PA_CONTEXT_FAILED || state == MA_PA_CONTEXT_TERMINATED) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while connecting the PulseAudio context."); return MA_ERROR; } resultPA = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pMainLoop, 1, NULL); if (resultPA < 0) { return ma_result_from_pulse(resultPA); } } /* Should never get here. */ return MA_SUCCESS; } static ma_result ma_wait_for_pa_stream_to_connect__pulse(ma_context* pContext, ma_ptr pMainLoop, ma_ptr pStream) { int resultPA; ma_pa_stream_state_t state; for (;;) { state = ((ma_pa_stream_get_state_proc)pContext->pulse.pa_stream_get_state)((ma_pa_stream*)pStream); if (state == MA_PA_STREAM_READY) { break; /* Done. */ } if (state == MA_PA_STREAM_FAILED || state == MA_PA_STREAM_TERMINATED) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while connecting the PulseAudio stream."); return MA_ERROR; } resultPA = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pMainLoop, 1, NULL); if (resultPA < 0) { return ma_result_from_pulse(resultPA); } } return MA_SUCCESS; } static ma_result ma_init_pa_mainloop_and_pa_context__pulse(ma_context* pContext, const char* pApplicationName, const char* pServerName, ma_bool32 tryAutoSpawn, ma_ptr* ppMainLoop, ma_ptr* ppPulseContext) { ma_result result; ma_ptr pMainLoop; ma_ptr pPulseContext; MA_ASSERT(ppMainLoop != NULL); MA_ASSERT(ppPulseContext != NULL); /* The PulseAudio context maps well to miniaudio's notion of a context. The pa_context object will be initialized as part of the ma_context. */ pMainLoop = ((ma_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)(); if (pMainLoop == NULL) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create mainloop."); return MA_FAILED_TO_INIT_BACKEND; } pPulseContext = ((ma_pa_context_new_proc)pContext->pulse.pa_context_new)(((ma_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)((ma_pa_mainloop*)pMainLoop), pApplicationName); if (pPulseContext == NULL) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio context."); ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)(pMainLoop)); return MA_FAILED_TO_INIT_BACKEND; } /* Now we need to connect to the context. Everything is asynchronous so we need to wait for it to connect before returning. */ result = ma_result_from_pulse(((ma_pa_context_connect_proc)pContext->pulse.pa_context_connect)((ma_pa_context*)pPulseContext, pServerName, (tryAutoSpawn) ? MA_PA_CONTEXT_NOFLAGS : MA_PA_CONTEXT_NOAUTOSPAWN, NULL)); if (result != MA_SUCCESS) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio context."); ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)(pPulseContext)); ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)(pMainLoop)); return result; } /* Since ma_context_init() runs synchronously we need to wait for the PulseAudio context to connect before we return. */ result = ma_wait_for_pa_context_to_connect__pulse(pContext, pMainLoop, pPulseContext); if (result != MA_SUCCESS) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] Waiting for connection failed."); ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)(pPulseContext)); ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)(pMainLoop)); return result; } *ppMainLoop = pMainLoop; *ppPulseContext = pPulseContext; return MA_SUCCESS; } static void ma_device_sink_info_callback(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData) { ma_pa_sink_info* pInfoOut; if (endOfList > 0) { return; } /* There has been a report that indicates that pInfo can be null which results in a null pointer dereference below. We'll check for this for safety. */ if (pInfo == NULL) { return; } pInfoOut = (ma_pa_sink_info*)pUserData; MA_ASSERT(pInfoOut != NULL); *pInfoOut = *pInfo; (void)pPulseContext; /* Unused. */ } static void ma_device_source_info_callback(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData) { ma_pa_source_info* pInfoOut; if (endOfList > 0) { return; } /* There has been a report that indicates that pInfo can be null which results in a null pointer dereference below. We'll check for this for safety. */ if (pInfo == NULL) { return; } pInfoOut = (ma_pa_source_info*)pUserData; MA_ASSERT(pInfoOut != NULL); *pInfoOut = *pInfo; (void)pPulseContext; /* Unused. */ } #if 0 static void ma_device_sink_name_callback(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData) { ma_device* pDevice; if (endOfList > 0) { return; } pDevice = (ma_device*)pUserData; MA_ASSERT(pDevice != NULL); ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), pInfo->description, (size_t)-1); (void)pPulseContext; /* Unused. */ } static void ma_device_source_name_callback(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData) { ma_device* pDevice; if (endOfList > 0) { return; } pDevice = (ma_device*)pUserData; MA_ASSERT(pDevice != NULL); ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), pInfo->description, (size_t)-1); (void)pPulseContext; /* Unused. */ } #endif static ma_result ma_context_get_sink_info__pulse(ma_context* pContext, const char* pDeviceName, ma_pa_sink_info* pSinkInfo) { ma_pa_operation* pOP; pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)pContext->pulse.pPulseContext, pDeviceName, ma_device_sink_info_callback, pSinkInfo); if (pOP == NULL) { return MA_ERROR; } return ma_wait_for_operation_and_unref__pulse(pContext, pContext->pulse.pMainLoop, pOP); } static ma_result ma_context_get_source_info__pulse(ma_context* pContext, const char* pDeviceName, ma_pa_source_info* pSourceInfo) { ma_pa_operation* pOP; pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)pContext->pulse.pPulseContext, pDeviceName, ma_device_source_info_callback, pSourceInfo); if (pOP == NULL) { return MA_ERROR; } return ma_wait_for_operation_and_unref__pulse(pContext, pContext->pulse.pMainLoop, pOP); } static ma_result ma_context_get_default_device_index__pulse(ma_context* pContext, ma_device_type deviceType, ma_uint32* pIndex) { ma_result result; MA_ASSERT(pContext != NULL); MA_ASSERT(pIndex != NULL); if (pIndex != NULL) { *pIndex = (ma_uint32)-1; } if (deviceType == ma_device_type_playback) { ma_pa_sink_info sinkInfo; result = ma_context_get_sink_info__pulse(pContext, NULL, &sinkInfo); if (result != MA_SUCCESS) { return result; } if (pIndex != NULL) { *pIndex = sinkInfo.index; } } if (deviceType == ma_device_type_capture) { ma_pa_source_info sourceInfo; result = ma_context_get_source_info__pulse(pContext, NULL, &sourceInfo); if (result != MA_SUCCESS) { return result; } if (pIndex != NULL) { *pIndex = sourceInfo.index; } } return MA_SUCCESS; } typedef struct { ma_context* pContext; ma_enum_devices_callback_proc callback; void* pUserData; ma_bool32 isTerminated; ma_uint32 defaultDeviceIndexPlayback; ma_uint32 defaultDeviceIndexCapture; } ma_context_enumerate_devices_callback_data__pulse; static void ma_context_enumerate_devices_sink_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_sink_info* pSinkInfo, int endOfList, void* pUserData) { ma_context_enumerate_devices_callback_data__pulse* pData = (ma_context_enumerate_devices_callback_data__pulse*)pUserData; ma_device_info deviceInfo; MA_ASSERT(pData != NULL); if (endOfList || pData->isTerminated) { return; } MA_ZERO_OBJECT(&deviceInfo); /* The name from PulseAudio is the ID for miniaudio. */ if (pSinkInfo->name != NULL) { ma_strncpy_s(deviceInfo.id.pulse, sizeof(deviceInfo.id.pulse), pSinkInfo->name, (size_t)-1); } /* The description from PulseAudio is the name for miniaudio. */ if (pSinkInfo->description != NULL) { ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), pSinkInfo->description, (size_t)-1); } if (pSinkInfo->index == pData->defaultDeviceIndexPlayback) { deviceInfo.isDefault = MA_TRUE; } pData->isTerminated = !pData->callback(pData->pContext, ma_device_type_playback, &deviceInfo, pData->pUserData); (void)pPulseContext; /* Unused. */ } static void ma_context_enumerate_devices_source_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_source_info* pSourceInfo, int endOfList, void* pUserData) { ma_context_enumerate_devices_callback_data__pulse* pData = (ma_context_enumerate_devices_callback_data__pulse*)pUserData; ma_device_info deviceInfo; MA_ASSERT(pData != NULL); if (endOfList || pData->isTerminated) { return; } MA_ZERO_OBJECT(&deviceInfo); /* The name from PulseAudio is the ID for miniaudio. */ if (pSourceInfo->name != NULL) { ma_strncpy_s(deviceInfo.id.pulse, sizeof(deviceInfo.id.pulse), pSourceInfo->name, (size_t)-1); } /* The description from PulseAudio is the name for miniaudio. */ if (pSourceInfo->description != NULL) { ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), pSourceInfo->description, (size_t)-1); } if (pSourceInfo->index == pData->defaultDeviceIndexCapture) { deviceInfo.isDefault = MA_TRUE; } pData->isTerminated = !pData->callback(pData->pContext, ma_device_type_capture, &deviceInfo, pData->pUserData); (void)pPulseContext; /* Unused. */ } static ma_result ma_context_enumerate_devices__pulse(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { ma_result result = MA_SUCCESS; ma_context_enumerate_devices_callback_data__pulse callbackData; ma_pa_operation* pOP = NULL; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); callbackData.pContext = pContext; callbackData.callback = callback; callbackData.pUserData = pUserData; callbackData.isTerminated = MA_FALSE; callbackData.defaultDeviceIndexPlayback = (ma_uint32)-1; callbackData.defaultDeviceIndexCapture = (ma_uint32)-1; /* We need to get the index of the default devices. */ ma_context_get_default_device_index__pulse(pContext, ma_device_type_playback, &callbackData.defaultDeviceIndexPlayback); ma_context_get_default_device_index__pulse(pContext, ma_device_type_capture, &callbackData.defaultDeviceIndexCapture); /* Playback. */ if (!callbackData.isTerminated) { pOP = ((ma_pa_context_get_sink_info_list_proc)pContext->pulse.pa_context_get_sink_info_list)((ma_pa_context*)(pContext->pulse.pPulseContext), ma_context_enumerate_devices_sink_callback__pulse, &callbackData); if (pOP == NULL) { result = MA_ERROR; goto done; } result = ma_wait_for_operation__pulse(pContext, pContext->pulse.pMainLoop, pOP); ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); if (result != MA_SUCCESS) { goto done; } } /* Capture. */ if (!callbackData.isTerminated) { pOP = ((ma_pa_context_get_source_info_list_proc)pContext->pulse.pa_context_get_source_info_list)((ma_pa_context*)(pContext->pulse.pPulseContext), ma_context_enumerate_devices_source_callback__pulse, &callbackData); if (pOP == NULL) { result = MA_ERROR; goto done; } result = ma_wait_for_operation__pulse(pContext, pContext->pulse.pMainLoop, pOP); ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); if (result != MA_SUCCESS) { goto done; } } done: return result; } typedef struct { ma_device_info* pDeviceInfo; ma_uint32 defaultDeviceIndex; ma_bool32 foundDevice; } ma_context_get_device_info_callback_data__pulse; static void ma_context_get_device_info_sink_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData) { ma_context_get_device_info_callback_data__pulse* pData = (ma_context_get_device_info_callback_data__pulse*)pUserData; if (endOfList > 0) { return; } MA_ASSERT(pData != NULL); pData->foundDevice = MA_TRUE; if (pInfo->name != NULL) { ma_strncpy_s(pData->pDeviceInfo->id.pulse, sizeof(pData->pDeviceInfo->id.pulse), pInfo->name, (size_t)-1); } if (pInfo->description != NULL) { ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pInfo->description, (size_t)-1); } /* We're just reporting a single data format here. I think technically PulseAudio might support all formats, but I don't trust that PulseAudio will do *anything* right, so I'm just going to report the "native" device format. */ pData->pDeviceInfo->nativeDataFormats[0].format = ma_format_from_pulse(pInfo->sample_spec.format); pData->pDeviceInfo->nativeDataFormats[0].channels = pInfo->sample_spec.channels; pData->pDeviceInfo->nativeDataFormats[0].sampleRate = pInfo->sample_spec.rate; pData->pDeviceInfo->nativeDataFormats[0].flags = 0; pData->pDeviceInfo->nativeDataFormatCount = 1; if (pData->defaultDeviceIndex == pInfo->index) { pData->pDeviceInfo->isDefault = MA_TRUE; } (void)pPulseContext; /* Unused. */ } static void ma_context_get_device_info_source_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData) { ma_context_get_device_info_callback_data__pulse* pData = (ma_context_get_device_info_callback_data__pulse*)pUserData; if (endOfList > 0) { return; } MA_ASSERT(pData != NULL); pData->foundDevice = MA_TRUE; if (pInfo->name != NULL) { ma_strncpy_s(pData->pDeviceInfo->id.pulse, sizeof(pData->pDeviceInfo->id.pulse), pInfo->name, (size_t)-1); } if (pInfo->description != NULL) { ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pInfo->description, (size_t)-1); } /* We're just reporting a single data format here. I think technically PulseAudio might support all formats, but I don't trust that PulseAudio will do *anything* right, so I'm just going to report the "native" device format. */ pData->pDeviceInfo->nativeDataFormats[0].format = ma_format_from_pulse(pInfo->sample_spec.format); pData->pDeviceInfo->nativeDataFormats[0].channels = pInfo->sample_spec.channels; pData->pDeviceInfo->nativeDataFormats[0].sampleRate = pInfo->sample_spec.rate; pData->pDeviceInfo->nativeDataFormats[0].flags = 0; pData->pDeviceInfo->nativeDataFormatCount = 1; if (pData->defaultDeviceIndex == pInfo->index) { pData->pDeviceInfo->isDefault = MA_TRUE; } (void)pPulseContext; /* Unused. */ } static ma_result ma_context_get_device_info__pulse(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { ma_result result = MA_SUCCESS; ma_context_get_device_info_callback_data__pulse callbackData; ma_pa_operation* pOP = NULL; const char* pDeviceName = NULL; MA_ASSERT(pContext != NULL); callbackData.pDeviceInfo = pDeviceInfo; callbackData.foundDevice = MA_FALSE; if (pDeviceID != NULL) { pDeviceName = pDeviceID->pulse; } else { pDeviceName = NULL; } result = ma_context_get_default_device_index__pulse(pContext, deviceType, &callbackData.defaultDeviceIndex); if (deviceType == ma_device_type_playback) { pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)(pContext->pulse.pPulseContext), pDeviceName, ma_context_get_device_info_sink_callback__pulse, &callbackData); } else { pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)(pContext->pulse.pPulseContext), pDeviceName, ma_context_get_device_info_source_callback__pulse, &callbackData); } if (pOP != NULL) { ma_wait_for_operation_and_unref__pulse(pContext, pContext->pulse.pMainLoop, pOP); } else { result = MA_ERROR; goto done; } if (!callbackData.foundDevice) { result = MA_NO_DEVICE; goto done; } done: return result; } static ma_result ma_device_uninit__pulse(ma_device* pDevice) { ma_context* pContext; MA_ASSERT(pDevice != NULL); pContext = pDevice->pContext; MA_ASSERT(pContext != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture); ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); } if (pDevice->type == ma_device_type_duplex) { ma_duplex_rb_uninit(&pDevice->duplexRB); } ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)((ma_pa_context*)pDevice->pulse.pPulseContext); ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)pDevice->pulse.pPulseContext); ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)pDevice->pulse.pMainLoop); return MA_SUCCESS; } static ma_pa_buffer_attr ma_device__pa_buffer_attr_new(ma_uint32 periodSizeInFrames, ma_uint32 periods, const ma_pa_sample_spec* ss) { ma_pa_buffer_attr attr; attr.maxlength = periodSizeInFrames * periods * ma_get_bytes_per_frame(ma_format_from_pulse(ss->format), ss->channels); attr.tlength = attr.maxlength / periods; attr.prebuf = (ma_uint32)-1; attr.minreq = (ma_uint32)-1; attr.fragsize = attr.maxlength / periods; return attr; } static ma_pa_stream* ma_device__pa_stream_new__pulse(ma_device* pDevice, const char* pStreamName, const ma_pa_sample_spec* ss, const ma_pa_channel_map* cmap) { static ma_atomic_uint32 g_StreamCounter = { 0 }; char actualStreamName[256]; if (pStreamName != NULL) { ma_strncpy_s(actualStreamName, sizeof(actualStreamName), pStreamName, (size_t)-1); } else { const char* pBaseName = "miniaudio:"; size_t baseNameLen = strlen(pBaseName); ma_strcpy_s(actualStreamName, sizeof(actualStreamName), pBaseName); ma_itoa_s((int)ma_atomic_uint32_get(&g_StreamCounter), actualStreamName + baseNameLen, sizeof(actualStreamName)-baseNameLen, 10); } ma_atomic_uint32_fetch_add(&g_StreamCounter, 1); return ((ma_pa_stream_new_proc)pDevice->pContext->pulse.pa_stream_new)((ma_pa_context*)pDevice->pulse.pPulseContext, actualStreamName, ss, cmap); } static void ma_device_on_read__pulse(ma_pa_stream* pStream, size_t byteCount, void* pUserData) { ma_device* pDevice = (ma_device*)pUserData; ma_uint32 bpf; ma_uint32 deviceState; ma_uint64 frameCount; ma_uint64 framesProcessed; MA_ASSERT(pDevice != NULL); /* Don't do anything if the device isn't initialized yet. Yes, this can happen because PulseAudio can fire this callback before the stream has even started. Ridiculous. */ deviceState = ma_device_get_state(pDevice); if (deviceState != ma_device_state_starting && deviceState != ma_device_state_started) { return; } bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); MA_ASSERT(bpf > 0); frameCount = byteCount / bpf; framesProcessed = 0; while (ma_device_get_state(pDevice) == ma_device_state_started && framesProcessed < frameCount) { const void* pMappedPCMFrames; size_t bytesMapped; ma_uint64 framesMapped; int pulseResult = ((ma_pa_stream_peek_proc)pDevice->pContext->pulse.pa_stream_peek)(pStream, &pMappedPCMFrames, &bytesMapped); if (pulseResult < 0) { break; /* Failed to map. Abort. */ } framesMapped = bytesMapped / bpf; if (framesMapped > 0) { if (pMappedPCMFrames != NULL) { ma_device_handle_backend_data_callback(pDevice, NULL, pMappedPCMFrames, framesMapped); } else { /* It's a hole. */ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] ma_device_on_read__pulse: Hole.\n"); } pulseResult = ((ma_pa_stream_drop_proc)pDevice->pContext->pulse.pa_stream_drop)(pStream); if (pulseResult < 0) { break; /* Failed to drop the buffer. */ } framesProcessed += framesMapped; } else { /* Nothing was mapped. Just abort. */ break; } } } static ma_result ma_device_write_to_stream__pulse(ma_device* pDevice, ma_pa_stream* pStream, ma_uint64* pFramesProcessed) { ma_result result = MA_SUCCESS; ma_uint64 framesProcessed = 0; size_t bytesMapped; ma_uint32 bpf; ma_uint32 deviceState; MA_ASSERT(pDevice != NULL); MA_ASSERT(pStream != NULL); bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); MA_ASSERT(bpf > 0); deviceState = ma_device_get_state(pDevice); bytesMapped = ((ma_pa_stream_writable_size_proc)pDevice->pContext->pulse.pa_stream_writable_size)(pStream); if (bytesMapped != (size_t)-1) { if (bytesMapped > 0) { ma_uint64 framesMapped; void* pMappedPCMFrames; int pulseResult = ((ma_pa_stream_begin_write_proc)pDevice->pContext->pulse.pa_stream_begin_write)(pStream, &pMappedPCMFrames, &bytesMapped); if (pulseResult < 0) { result = ma_result_from_pulse(pulseResult); goto done; } framesMapped = bytesMapped / bpf; if (deviceState == ma_device_state_started || deviceState == ma_device_state_starting) { /* Check for starting state just in case this is being used to do the initial fill. */ ma_device_handle_backend_data_callback(pDevice, pMappedPCMFrames, NULL, framesMapped); } else { /* Device is not started. Write silence. */ ma_silence_pcm_frames(pMappedPCMFrames, framesMapped, pDevice->playback.format, pDevice->playback.channels); } pulseResult = ((ma_pa_stream_write_proc)pDevice->pContext->pulse.pa_stream_write)(pStream, pMappedPCMFrames, bytesMapped, NULL, 0, MA_PA_SEEK_RELATIVE); if (pulseResult < 0) { result = ma_result_from_pulse(pulseResult); goto done; /* Failed to write data to stream. */ } framesProcessed += framesMapped; } else { result = MA_SUCCESS; /* No data available for writing. */ goto done; } } else { result = MA_ERROR; /* Failed to retrieve the writable size. Abort. */ goto done; } done: if (pFramesProcessed != NULL) { *pFramesProcessed = framesProcessed; } return result; } static void ma_device_on_write__pulse(ma_pa_stream* pStream, size_t byteCount, void* pUserData) { ma_device* pDevice = (ma_device*)pUserData; ma_uint32 bpf; ma_uint64 frameCount; ma_uint64 framesProcessed; ma_uint32 deviceState; ma_result result; MA_ASSERT(pDevice != NULL); /* Don't do anything if the device isn't initialized yet. Yes, this can happen because PulseAudio can fire this callback before the stream has even started. Ridiculous. */ deviceState = ma_device_get_state(pDevice); if (deviceState != ma_device_state_starting && deviceState != ma_device_state_started) { return; } bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); MA_ASSERT(bpf > 0); frameCount = byteCount / bpf; framesProcessed = 0; while (framesProcessed < frameCount) { ma_uint64 framesProcessedThisIteration; /* Don't keep trying to process frames if the device isn't started. */ deviceState = ma_device_get_state(pDevice); if (deviceState != ma_device_state_starting && deviceState != ma_device_state_started) { break; } result = ma_device_write_to_stream__pulse(pDevice, pStream, &framesProcessedThisIteration); if (result != MA_SUCCESS) { break; } framesProcessed += framesProcessedThisIteration; } } static void ma_device_on_suspended__pulse(ma_pa_stream* pStream, void* pUserData) { ma_device* pDevice = (ma_device*)pUserData; int suspended; (void)pStream; suspended = ((ma_pa_stream_is_suspended_proc)pDevice->pContext->pulse.pa_stream_is_suspended)(pStream); ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. pa_stream_is_suspended() returned %d.\n", suspended); if (suspended < 0) { return; } if (suspended == 1) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. Suspended.\n"); ma_device__on_notification_stopped(pDevice); } else { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. Resumed.\n"); ma_device__on_notification_started(pDevice); } } static void ma_device_on_rerouted__pulse(ma_pa_stream* pStream, void* pUserData) { ma_device* pDevice = (ma_device*)pUserData; (void)pStream; (void)pUserData; ma_device__on_notification_rerouted(pDevice); } static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__pulse(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile) { /* There have been reports from users where buffers of < ~20ms result glitches when running through PipeWire. To work around this we're going to have to use a different default buffer size. */ const ma_uint32 defaultPeriodSizeInMilliseconds_LowLatency = 25; const ma_uint32 defaultPeriodSizeInMilliseconds_Conservative = MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE; MA_ASSERT(nativeSampleRate != 0); if (pDescriptor->periodSizeInFrames == 0) { if (pDescriptor->periodSizeInMilliseconds == 0) { if (performanceProfile == ma_performance_profile_low_latency) { return ma_calculate_buffer_size_in_frames_from_milliseconds(defaultPeriodSizeInMilliseconds_LowLatency, nativeSampleRate); } else { return ma_calculate_buffer_size_in_frames_from_milliseconds(defaultPeriodSizeInMilliseconds_Conservative, nativeSampleRate); } } else { return ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptor->periodSizeInMilliseconds, nativeSampleRate); } } else { return pDescriptor->periodSizeInFrames; } } static ma_result ma_device_init__pulse(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { /* Notes for PulseAudio: - When both the period size in frames and milliseconds are 0, we default to miniaudio's default buffer sizes rather than leaving it up to PulseAudio because I don't trust PulseAudio to give us any kind of reasonable latency by default. - Do not ever, *ever* forget to use MA_PA_STREAM_ADJUST_LATENCY. If you don't specify this flag, capture mode will just not work properly until you open another PulseAudio app. */ ma_result result = MA_SUCCESS; int error = 0; const char* devPlayback = NULL; const char* devCapture = NULL; ma_format format = ma_format_unknown; ma_uint32 channels = 0; ma_uint32 sampleRate = 0; ma_pa_sink_info sinkInfo; ma_pa_source_info sourceInfo; ma_pa_sample_spec ss; ma_pa_channel_map cmap; ma_pa_buffer_attr attr; const ma_pa_sample_spec* pActualSS = NULL; const ma_pa_buffer_attr* pActualAttr = NULL; const ma_pa_channel_map* pActualChannelMap = NULL; ma_uint32 iChannel; int streamFlags; MA_ASSERT(pDevice != NULL); MA_ZERO_OBJECT(&pDevice->pulse); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } /* No exclusive mode with the PulseAudio backend. */ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { return MA_SHARE_MODE_NOT_SUPPORTED; } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { if (pDescriptorPlayback->pDeviceID != NULL) { devPlayback = pDescriptorPlayback->pDeviceID->pulse; } format = pDescriptorPlayback->format; channels = pDescriptorPlayback->channels; sampleRate = pDescriptorPlayback->sampleRate; } if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { if (pDescriptorCapture->pDeviceID != NULL) { devCapture = pDescriptorCapture->pDeviceID->pulse; } format = pDescriptorCapture->format; channels = pDescriptorCapture->channels; sampleRate = pDescriptorCapture->sampleRate; } result = ma_init_pa_mainloop_and_pa_context__pulse(pDevice->pContext, pDevice->pContext->pulse.pApplicationName, pDevice->pContext->pulse.pServerName, MA_FALSE, &pDevice->pulse.pMainLoop, &pDevice->pulse.pPulseContext); if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to initialize PA mainloop and context for device.\n"); return result; } if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { result = ma_context_get_source_info__pulse(pDevice->pContext, devCapture, &sourceInfo); if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve source info for capture device."); goto on_error0; } ss = sourceInfo.sample_spec; cmap = sourceInfo.channel_map; /* Use the requested channel count if we have one. */ if (pDescriptorCapture->channels != 0) { ss.channels = pDescriptorCapture->channels; } /* PulseAudio has a maximum channel count of 32. We'll get a crash if this is exceeded. */ if (ss.channels > 32) { ss.channels = 32; } /* Use a default channel map. */ ((ma_pa_channel_map_init_extend_proc)pDevice->pContext->pulse.pa_channel_map_init_extend)(&cmap, ss.channels, (ma_pa_channel_map_def_t)pConfig->pulse.channelMap); /* Use the requested sample rate if one was specified. */ if (pDescriptorCapture->sampleRate != 0) { ss.rate = pDescriptorCapture->sampleRate; } streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_ADJUST_LATENCY; if (ma_format_from_pulse(ss.format) == ma_format_unknown) { if (ma_is_little_endian()) { ss.format = MA_PA_SAMPLE_FLOAT32LE; } else { ss.format = MA_PA_SAMPLE_FLOAT32BE; } streamFlags |= MA_PA_STREAM_FIX_FORMAT; ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.format not supported by miniaudio. Defaulting to PA_SAMPLE_FLOAT32.\n"); } if (ss.rate == 0) { ss.rate = MA_DEFAULT_SAMPLE_RATE; streamFlags |= MA_PA_STREAM_FIX_RATE; ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.rate = 0. Defaulting to %d.\n", ss.rate); } if (ss.channels == 0) { ss.channels = MA_DEFAULT_CHANNELS; streamFlags |= MA_PA_STREAM_FIX_CHANNELS; ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.channels = 0. Defaulting to %d.\n", ss.channels); } /* We now have enough information to calculate our actual period size in frames. */ pDescriptorCapture->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__pulse(pDescriptorCapture, ss.rate, pConfig->performanceProfile); attr = ma_device__pa_buffer_attr_new(pDescriptorCapture->periodSizeInFrames, pDescriptorCapture->periodCount, &ss); ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Capture attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorCapture->periodSizeInFrames); pDevice->pulse.pStreamCapture = ma_device__pa_stream_new__pulse(pDevice, pConfig->pulse.pStreamNameCapture, &ss, &cmap); if (pDevice->pulse.pStreamCapture == NULL) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio capture stream.\n"); result = MA_ERROR; goto on_error0; } /* The callback needs to be set before connecting the stream. */ ((ma_pa_stream_set_read_callback_proc)pDevice->pContext->pulse.pa_stream_set_read_callback)((ma_pa_stream*)pDevice->pulse.pStreamCapture, ma_device_on_read__pulse, pDevice); /* State callback for checking when the device has been corked. */ ((ma_pa_stream_set_suspended_callback_proc)pDevice->pContext->pulse.pa_stream_set_suspended_callback)((ma_pa_stream*)pDevice->pulse.pStreamCapture, ma_device_on_suspended__pulse, pDevice); /* Rerouting notification. */ ((ma_pa_stream_set_moved_callback_proc)pDevice->pContext->pulse.pa_stream_set_moved_callback)((ma_pa_stream*)pDevice->pulse.pStreamCapture, ma_device_on_rerouted__pulse, pDevice); /* Connect after we've got all of our internal state set up. */ if (devCapture != NULL) { streamFlags |= MA_PA_STREAM_DONT_MOVE; } error = ((ma_pa_stream_connect_record_proc)pDevice->pContext->pulse.pa_stream_connect_record)((ma_pa_stream*)pDevice->pulse.pStreamCapture, devCapture, &attr, (ma_pa_stream_flags_t)streamFlags); if (error != MA_PA_OK) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio capture stream."); result = ma_result_from_pulse(error); goto on_error1; } result = ma_wait_for_pa_stream_to_connect__pulse(pDevice->pContext, pDevice->pulse.pMainLoop, (ma_pa_stream*)pDevice->pulse.pStreamCapture); if (result != MA_SUCCESS) { goto on_error2; } /* Internal format. */ pActualSS = ((ma_pa_stream_get_sample_spec_proc)pDevice->pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamCapture); if (pActualSS != NULL) { ss = *pActualSS; ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Capture sample spec: format=%s, channels=%d, rate=%d\n", ma_get_format_name(ma_format_from_pulse(ss.format)), ss.channels, ss.rate); } else { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Failed to retrieve capture sample spec.\n"); } pDescriptorCapture->format = ma_format_from_pulse(ss.format); pDescriptorCapture->channels = ss.channels; pDescriptorCapture->sampleRate = ss.rate; if (pDescriptorCapture->format == ma_format_unknown || pDescriptorCapture->channels == 0 || pDescriptorCapture->sampleRate == 0) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Capture sample spec is invalid. Device unusable by miniaudio. format=%s, channels=%d, sampleRate=%d.\n", ma_get_format_name(pDescriptorCapture->format), pDescriptorCapture->channels, pDescriptorCapture->sampleRate); result = MA_ERROR; goto on_error4; } /* Internal channel map. */ pActualChannelMap = ((ma_pa_stream_get_channel_map_proc)pDevice->pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamCapture); if (pActualChannelMap == NULL) { pActualChannelMap = &cmap; /* Fallback just in case. */ } /* Bug in PipeWire. There have been reports that PipeWire is returning AUX channels when reporting the channel map. To somewhat workaround this, I'm hacking in a hard coded channel map for mono and stereo. In this case it should be safe to assume mono = MONO and stereo = LEFT/RIGHT. For all other channel counts we need to just put up with whatever PipeWire reports and hope it gets fixed sooner than later. I might remove this hack later. */ if (pDescriptorCapture->channels > 2) { for (iChannel = 0; iChannel < pDescriptorCapture->channels; iChannel += 1) { pDescriptorCapture->channelMap[iChannel] = ma_channel_position_from_pulse(pActualChannelMap->map[iChannel]); } } else { /* Hack for mono and stereo. */ if (pDescriptorCapture->channels == 1) { pDescriptorCapture->channelMap[0] = MA_CHANNEL_MONO; } else if (pDescriptorCapture->channels == 2) { pDescriptorCapture->channelMap[0] = MA_CHANNEL_FRONT_LEFT; pDescriptorCapture->channelMap[1] = MA_CHANNEL_FRONT_RIGHT; } else { MA_ASSERT(MA_FALSE); /* Should never hit this. */ } } /* Buffer. */ pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pDevice->pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamCapture); if (pActualAttr != NULL) { attr = *pActualAttr; } if (attr.fragsize > 0) { pDescriptorCapture->periodCount = ma_max(attr.maxlength / attr.fragsize, 1); } else { pDescriptorCapture->periodCount = 1; } pDescriptorCapture->periodSizeInFrames = attr.maxlength / ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) / pDescriptorCapture->periodCount; ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Capture actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorCapture->periodSizeInFrames); } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { result = ma_context_get_sink_info__pulse(pDevice->pContext, devPlayback, &sinkInfo); if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve sink info for playback device.\n"); goto on_error2; } ss = sinkInfo.sample_spec; cmap = sinkInfo.channel_map; /* Use the requested channel count if we have one. */ if (pDescriptorPlayback->channels != 0) { ss.channels = pDescriptorPlayback->channels; } /* PulseAudio has a maximum channel count of 32. We'll get a crash if this is exceeded. */ if (ss.channels > 32) { ss.channels = 32; } /* Use a default channel map. */ ((ma_pa_channel_map_init_extend_proc)pDevice->pContext->pulse.pa_channel_map_init_extend)(&cmap, ss.channels, (ma_pa_channel_map_def_t)pConfig->pulse.channelMap); /* Use the requested sample rate if one was specified. */ if (pDescriptorPlayback->sampleRate != 0) { ss.rate = pDescriptorPlayback->sampleRate; } streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_ADJUST_LATENCY; if (ma_format_from_pulse(ss.format) == ma_format_unknown) { if (ma_is_little_endian()) { ss.format = MA_PA_SAMPLE_FLOAT32LE; } else { ss.format = MA_PA_SAMPLE_FLOAT32BE; } streamFlags |= MA_PA_STREAM_FIX_FORMAT; ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.format not supported by miniaudio. Defaulting to PA_SAMPLE_FLOAT32.\n"); } if (ss.rate == 0) { ss.rate = MA_DEFAULT_SAMPLE_RATE; streamFlags |= MA_PA_STREAM_FIX_RATE; ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.rate = 0. Defaulting to %d.\n", ss.rate); } if (ss.channels == 0) { ss.channels = MA_DEFAULT_CHANNELS; streamFlags |= MA_PA_STREAM_FIX_CHANNELS; ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.channels = 0. Defaulting to %d.\n", ss.channels); } /* We now have enough information to calculate the actual buffer size in frames. */ pDescriptorPlayback->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__pulse(pDescriptorPlayback, ss.rate, pConfig->performanceProfile); attr = ma_device__pa_buffer_attr_new(pDescriptorPlayback->periodSizeInFrames, pDescriptorPlayback->periodCount, &ss); ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Playback attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorPlayback->periodSizeInFrames); pDevice->pulse.pStreamPlayback = ma_device__pa_stream_new__pulse(pDevice, pConfig->pulse.pStreamNamePlayback, &ss, &cmap); if (pDevice->pulse.pStreamPlayback == NULL) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio playback stream.\n"); result = MA_ERROR; goto on_error2; } /* Note that this callback will be fired as soon as the stream is connected, even though it's started as corked. The callback needs to handle a device state of ma_device_state_uninitialized. */ ((ma_pa_stream_set_write_callback_proc)pDevice->pContext->pulse.pa_stream_set_write_callback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_device_on_write__pulse, pDevice); /* State callback for checking when the device has been corked. */ ((ma_pa_stream_set_suspended_callback_proc)pDevice->pContext->pulse.pa_stream_set_suspended_callback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_device_on_suspended__pulse, pDevice); /* Rerouting notification. */ ((ma_pa_stream_set_moved_callback_proc)pDevice->pContext->pulse.pa_stream_set_moved_callback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_device_on_rerouted__pulse, pDevice); /* Connect after we've got all of our internal state set up. */ if (devPlayback != NULL) { streamFlags |= MA_PA_STREAM_DONT_MOVE; } error = ((ma_pa_stream_connect_playback_proc)pDevice->pContext->pulse.pa_stream_connect_playback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, devPlayback, &attr, (ma_pa_stream_flags_t)streamFlags, NULL, NULL); if (error != MA_PA_OK) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio playback stream."); result = ma_result_from_pulse(error); goto on_error3; } result = ma_wait_for_pa_stream_to_connect__pulse(pDevice->pContext, pDevice->pulse.pMainLoop, (ma_pa_stream*)pDevice->pulse.pStreamPlayback); if (result != MA_SUCCESS) { goto on_error3; } /* Internal format. */ pActualSS = ((ma_pa_stream_get_sample_spec_proc)pDevice->pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); if (pActualSS != NULL) { ss = *pActualSS; ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Playback sample spec: format=%s, channels=%d, rate=%d\n", ma_get_format_name(ma_format_from_pulse(ss.format)), ss.channels, ss.rate); } else { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Failed to retrieve playback sample spec.\n"); } pDescriptorPlayback->format = ma_format_from_pulse(ss.format); pDescriptorPlayback->channels = ss.channels; pDescriptorPlayback->sampleRate = ss.rate; if (pDescriptorPlayback->format == ma_format_unknown || pDescriptorPlayback->channels == 0 || pDescriptorPlayback->sampleRate == 0) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Playback sample spec is invalid. Device unusable by miniaudio. format=%s, channels=%d, sampleRate=%d.\n", ma_get_format_name(pDescriptorPlayback->format), pDescriptorPlayback->channels, pDescriptorPlayback->sampleRate); result = MA_ERROR; goto on_error4; } /* Internal channel map. */ pActualChannelMap = ((ma_pa_stream_get_channel_map_proc)pDevice->pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); if (pActualChannelMap == NULL) { pActualChannelMap = &cmap; /* Fallback just in case. */ } /* Bug in PipeWire. There have been reports that PipeWire is returning AUX channels when reporting the channel map. To somewhat workaround this, I'm hacking in a hard coded channel map for mono and stereo. In this case it should be safe to assume mono = MONO and stereo = LEFT/RIGHT. For all other channel counts we need to just put up with whatever PipeWire reports and hope it gets fixed sooner than later. I might remove this hack later. */ if (pDescriptorPlayback->channels > 2) { for (iChannel = 0; iChannel < pDescriptorPlayback->channels; iChannel += 1) { pDescriptorPlayback->channelMap[iChannel] = ma_channel_position_from_pulse(pActualChannelMap->map[iChannel]); } } else { /* Hack for mono and stereo. */ if (pDescriptorPlayback->channels == 1) { pDescriptorPlayback->channelMap[0] = MA_CHANNEL_MONO; } else if (pDescriptorPlayback->channels == 2) { pDescriptorPlayback->channelMap[0] = MA_CHANNEL_FRONT_LEFT; pDescriptorPlayback->channelMap[1] = MA_CHANNEL_FRONT_RIGHT; } else { MA_ASSERT(MA_FALSE); /* Should never hit this. */ } } /* Buffer. */ pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pDevice->pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); if (pActualAttr != NULL) { attr = *pActualAttr; } if (attr.tlength > 0) { pDescriptorPlayback->periodCount = ma_max(attr.maxlength / attr.tlength, 1); } else { pDescriptorPlayback->periodCount = 1; } pDescriptorPlayback->periodSizeInFrames = attr.maxlength / ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels) / pDescriptorPlayback->periodCount; ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Playback actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalPeriodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorPlayback->periodSizeInFrames); } /* We need a ring buffer for handling duplex mode. We can use the main duplex ring buffer in the main part of the ma_device struct. We cannot, however, depend on ma_device_init() initializing this for us later on because that will only do it if it's a fully asynchronous backend - i.e. the onDeviceDataLoop callback is NULL, which is not the case for PulseAudio. */ if (pConfig->deviceType == ma_device_type_duplex) { ma_format rbFormat = (format != ma_format_unknown) ? format : pDescriptorCapture->format; ma_uint32 rbChannels = (channels > 0) ? channels : pDescriptorCapture->channels; ma_uint32 rbSampleRate = (sampleRate > 0) ? sampleRate : pDescriptorCapture->sampleRate; result = ma_duplex_rb_init(rbFormat, rbChannels, rbSampleRate, pDescriptorCapture->sampleRate, pDescriptorCapture->periodSizeInFrames, &pDevice->pContext->allocationCallbacks, &pDevice->duplexRB); if (result != MA_SUCCESS) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to initialize ring buffer. %s.\n", ma_result_description(result)); goto on_error4; } } return MA_SUCCESS; on_error4: if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { ((ma_pa_stream_disconnect_proc)pDevice->pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); } on_error3: if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { ((ma_pa_stream_unref_proc)pDevice->pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); } on_error2: if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { ((ma_pa_stream_disconnect_proc)pDevice->pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture); } on_error1: if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { ((ma_pa_stream_unref_proc)pDevice->pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture); } on_error0: return result; } static void ma_pulse_operation_complete_callback(ma_pa_stream* pStream, int success, void* pUserData) { ma_bool32* pIsSuccessful = (ma_bool32*)pUserData; MA_ASSERT(pIsSuccessful != NULL); *pIsSuccessful = (ma_bool32)success; (void)pStream; /* Unused. */ } static ma_result ma_device__cork_stream__pulse(ma_device* pDevice, ma_device_type deviceType, int cork) { ma_context* pContext = pDevice->pContext; ma_bool32 wasSuccessful; ma_pa_stream* pStream; ma_pa_operation* pOP; ma_result result; /* This should not be called with a duplex device type. */ if (deviceType == ma_device_type_duplex) { return MA_INVALID_ARGS; } wasSuccessful = MA_FALSE; pStream = (ma_pa_stream*)((deviceType == ma_device_type_capture) ? pDevice->pulse.pStreamCapture : pDevice->pulse.pStreamPlayback); MA_ASSERT(pStream != NULL); pOP = ((ma_pa_stream_cork_proc)pContext->pulse.pa_stream_cork)(pStream, cork, ma_pulse_operation_complete_callback, &wasSuccessful); if (pOP == NULL) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to cork PulseAudio stream."); return MA_ERROR; } result = ma_wait_for_operation_and_unref__pulse(pDevice->pContext, pDevice->pulse.pMainLoop, pOP); if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while waiting for the PulseAudio stream to cork."); return result; } if (!wasSuccessful) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to %s PulseAudio stream.", (cork) ? "stop" : "start"); return MA_ERROR; } return MA_SUCCESS; } static ma_result ma_device_start__pulse(ma_device* pDevice) { ma_result result; MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { result = ma_device__cork_stream__pulse(pDevice, ma_device_type_capture, 0); if (result != MA_SUCCESS) { return result; } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { /* We need to fill some data before uncorking. Not doing this will result in the write callback never getting fired. We're not going to abort if writing fails because I still want the device to get uncorked. */ ma_device_write_to_stream__pulse(pDevice, (ma_pa_stream*)(pDevice->pulse.pStreamPlayback), NULL); /* No need to check the result here. Always want to fall through an uncork.*/ result = ma_device__cork_stream__pulse(pDevice, ma_device_type_playback, 0); if (result != MA_SUCCESS) { return result; } } return MA_SUCCESS; } static ma_result ma_device_stop__pulse(ma_device* pDevice) { ma_result result; MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { result = ma_device__cork_stream__pulse(pDevice, ma_device_type_capture, 1); if (result != MA_SUCCESS) { return result; } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { /* Ideally we would drain the device here, but there's been cases where PulseAudio seems to be broken on some systems to the point where no audio processing seems to happen. When this happens, draining never completes and we get stuck here. For now I'm disabling draining of the device so we don't just freeze the application. */ #if 0 ma_pa_operation* pOP = ((ma_pa_stream_drain_proc)pDevice->pContext->pulse.pa_stream_drain)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_pulse_operation_complete_callback, &wasSuccessful); ma_wait_for_operation_and_unref__pulse(pDevice->pContext, pDevice->pulse.pMainLoop, pOP); #endif result = ma_device__cork_stream__pulse(pDevice, ma_device_type_playback, 1); if (result != MA_SUCCESS) { return result; } } return MA_SUCCESS; } static ma_result ma_device_data_loop__pulse(ma_device* pDevice) { int resultPA; MA_ASSERT(pDevice != NULL); /* NOTE: Don't start the device here. It'll be done at a higher level. */ /* All data is handled through callbacks. All we need to do is iterate over the main loop and let the callbacks deal with it. */ while (ma_device_get_state(pDevice) == ma_device_state_started) { resultPA = ((ma_pa_mainloop_iterate_proc)pDevice->pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 1, NULL); if (resultPA < 0) { break; } } /* NOTE: Don't stop the device here. It'll be done at a higher level. */ return MA_SUCCESS; } static ma_result ma_device_data_loop_wakeup__pulse(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); ((ma_pa_mainloop_wakeup_proc)pDevice->pContext->pulse.pa_mainloop_wakeup)((ma_pa_mainloop*)pDevice->pulse.pMainLoop); return MA_SUCCESS; } static ma_result ma_context_uninit__pulse(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_pulseaudio); ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)((ma_pa_context*)pContext->pulse.pPulseContext); ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)pContext->pulse.pPulseContext); ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)pContext->pulse.pMainLoop); ma_free(pContext->pulse.pServerName, &pContext->allocationCallbacks); ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks); #ifndef MA_NO_RUNTIME_LINKING ma_dlclose(ma_context_get_log(pContext), pContext->pulse.pulseSO); #endif return MA_SUCCESS; } static ma_result ma_context_init__pulse(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { ma_result result; #ifndef MA_NO_RUNTIME_LINKING const char* libpulseNames[] = { "libpulse.so", "libpulse.so.0" }; size_t i; for (i = 0; i < ma_countof(libpulseNames); ++i) { pContext->pulse.pulseSO = ma_dlopen(ma_context_get_log(pContext), libpulseNames[i]); if (pContext->pulse.pulseSO != NULL) { break; } } if (pContext->pulse.pulseSO == NULL) { return MA_NO_BACKEND; } pContext->pulse.pa_mainloop_new = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_mainloop_new"); pContext->pulse.pa_mainloop_free = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_mainloop_free"); pContext->pulse.pa_mainloop_quit = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_mainloop_quit"); pContext->pulse.pa_mainloop_get_api = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_mainloop_get_api"); pContext->pulse.pa_mainloop_iterate = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_mainloop_iterate"); pContext->pulse.pa_mainloop_wakeup = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_mainloop_wakeup"); pContext->pulse.pa_threaded_mainloop_new = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_new"); pContext->pulse.pa_threaded_mainloop_free = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_free"); pContext->pulse.pa_threaded_mainloop_start = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_start"); pContext->pulse.pa_threaded_mainloop_stop = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_stop"); pContext->pulse.pa_threaded_mainloop_lock = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_lock"); pContext->pulse.pa_threaded_mainloop_unlock = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_unlock"); pContext->pulse.pa_threaded_mainloop_wait = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_wait"); pContext->pulse.pa_threaded_mainloop_signal = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_signal"); pContext->pulse.pa_threaded_mainloop_accept = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_accept"); pContext->pulse.pa_threaded_mainloop_get_retval = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_get_retval"); pContext->pulse.pa_threaded_mainloop_get_api = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_get_api"); pContext->pulse.pa_threaded_mainloop_in_thread = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_in_thread"); pContext->pulse.pa_threaded_mainloop_set_name = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_threaded_mainloop_set_name"); pContext->pulse.pa_context_new = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_context_new"); pContext->pulse.pa_context_unref = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_context_unref"); pContext->pulse.pa_context_connect = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_context_connect"); pContext->pulse.pa_context_disconnect = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_context_disconnect"); pContext->pulse.pa_context_set_state_callback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_context_set_state_callback"); pContext->pulse.pa_context_get_state = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_context_get_state"); pContext->pulse.pa_context_get_sink_info_list = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_context_get_sink_info_list"); pContext->pulse.pa_context_get_source_info_list = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_context_get_source_info_list"); pContext->pulse.pa_context_get_sink_info_by_name = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_context_get_sink_info_by_name"); pContext->pulse.pa_context_get_source_info_by_name = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_context_get_source_info_by_name"); pContext->pulse.pa_operation_unref = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_operation_unref"); pContext->pulse.pa_operation_get_state = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_operation_get_state"); pContext->pulse.pa_channel_map_init_extend = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_channel_map_init_extend"); pContext->pulse.pa_channel_map_valid = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_channel_map_valid"); pContext->pulse.pa_channel_map_compatible = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_channel_map_compatible"); pContext->pulse.pa_stream_new = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_new"); pContext->pulse.pa_stream_unref = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_unref"); pContext->pulse.pa_stream_connect_playback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_connect_playback"); pContext->pulse.pa_stream_connect_record = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_connect_record"); pContext->pulse.pa_stream_disconnect = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_disconnect"); pContext->pulse.pa_stream_get_state = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_get_state"); pContext->pulse.pa_stream_get_sample_spec = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_get_sample_spec"); pContext->pulse.pa_stream_get_channel_map = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_get_channel_map"); pContext->pulse.pa_stream_get_buffer_attr = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_get_buffer_attr"); pContext->pulse.pa_stream_set_buffer_attr = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_set_buffer_attr"); pContext->pulse.pa_stream_get_device_name = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_get_device_name"); pContext->pulse.pa_stream_set_write_callback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_set_write_callback"); pContext->pulse.pa_stream_set_read_callback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_set_read_callback"); pContext->pulse.pa_stream_set_suspended_callback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_set_suspended_callback"); pContext->pulse.pa_stream_set_moved_callback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_set_moved_callback"); pContext->pulse.pa_stream_is_suspended = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_is_suspended"); pContext->pulse.pa_stream_flush = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_flush"); pContext->pulse.pa_stream_drain = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_drain"); pContext->pulse.pa_stream_is_corked = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_is_corked"); pContext->pulse.pa_stream_cork = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_cork"); pContext->pulse.pa_stream_trigger = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_trigger"); pContext->pulse.pa_stream_begin_write = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_begin_write"); pContext->pulse.pa_stream_write = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_write"); pContext->pulse.pa_stream_peek = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_peek"); pContext->pulse.pa_stream_drop = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_drop"); pContext->pulse.pa_stream_writable_size = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_writable_size"); pContext->pulse.pa_stream_readable_size = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->pulse.pulseSO, "pa_stream_readable_size"); #else /* This strange assignment system is just for type safety. */ ma_pa_mainloop_new_proc _pa_mainloop_new = pa_mainloop_new; ma_pa_mainloop_free_proc _pa_mainloop_free = pa_mainloop_free; ma_pa_mainloop_quit_proc _pa_mainloop_quit = pa_mainloop_quit; ma_pa_mainloop_get_api_proc _pa_mainloop_get_api = pa_mainloop_get_api; ma_pa_mainloop_iterate_proc _pa_mainloop_iterate = pa_mainloop_iterate; ma_pa_mainloop_wakeup_proc _pa_mainloop_wakeup = pa_mainloop_wakeup; ma_pa_threaded_mainloop_new_proc _pa_threaded_mainloop_new = pa_threaded_mainloop_new; ma_pa_threaded_mainloop_free_proc _pa_threaded_mainloop_free = pa_threaded_mainloop_free; ma_pa_threaded_mainloop_start_proc _pa_threaded_mainloop_start = pa_threaded_mainloop_start; ma_pa_threaded_mainloop_stop_proc _pa_threaded_mainloop_stop = pa_threaded_mainloop_stop; ma_pa_threaded_mainloop_lock_proc _pa_threaded_mainloop_lock = pa_threaded_mainloop_lock; ma_pa_threaded_mainloop_unlock_proc _pa_threaded_mainloop_unlock = pa_threaded_mainloop_unlock; ma_pa_threaded_mainloop_wait_proc _pa_threaded_mainloop_wait = pa_threaded_mainloop_wait; ma_pa_threaded_mainloop_signal_proc _pa_threaded_mainloop_signal = pa_threaded_mainloop_signal; ma_pa_threaded_mainloop_accept_proc _pa_threaded_mainloop_accept = pa_threaded_mainloop_accept; ma_pa_threaded_mainloop_get_retval_proc _pa_threaded_mainloop_get_retval = pa_threaded_mainloop_get_retval; ma_pa_threaded_mainloop_get_api_proc _pa_threaded_mainloop_get_api = pa_threaded_mainloop_get_api; ma_pa_threaded_mainloop_in_thread_proc _pa_threaded_mainloop_in_thread = pa_threaded_mainloop_in_thread; ma_pa_threaded_mainloop_set_name_proc _pa_threaded_mainloop_set_name = pa_threaded_mainloop_set_name; ma_pa_context_new_proc _pa_context_new = pa_context_new; ma_pa_context_unref_proc _pa_context_unref = pa_context_unref; ma_pa_context_connect_proc _pa_context_connect = pa_context_connect; ma_pa_context_disconnect_proc _pa_context_disconnect = pa_context_disconnect; ma_pa_context_set_state_callback_proc _pa_context_set_state_callback = pa_context_set_state_callback; ma_pa_context_get_state_proc _pa_context_get_state = pa_context_get_state; ma_pa_context_get_sink_info_list_proc _pa_context_get_sink_info_list = pa_context_get_sink_info_list; ma_pa_context_get_source_info_list_proc _pa_context_get_source_info_list = pa_context_get_source_info_list; ma_pa_context_get_sink_info_by_name_proc _pa_context_get_sink_info_by_name = pa_context_get_sink_info_by_name; ma_pa_context_get_source_info_by_name_proc _pa_context_get_source_info_by_name= pa_context_get_source_info_by_name; ma_pa_operation_unref_proc _pa_operation_unref = pa_operation_unref; ma_pa_operation_get_state_proc _pa_operation_get_state = pa_operation_get_state; ma_pa_channel_map_init_extend_proc _pa_channel_map_init_extend = pa_channel_map_init_extend; ma_pa_channel_map_valid_proc _pa_channel_map_valid = pa_channel_map_valid; ma_pa_channel_map_compatible_proc _pa_channel_map_compatible = pa_channel_map_compatible; ma_pa_stream_new_proc _pa_stream_new = pa_stream_new; ma_pa_stream_unref_proc _pa_stream_unref = pa_stream_unref; ma_pa_stream_connect_playback_proc _pa_stream_connect_playback = pa_stream_connect_playback; ma_pa_stream_connect_record_proc _pa_stream_connect_record = pa_stream_connect_record; ma_pa_stream_disconnect_proc _pa_stream_disconnect = pa_stream_disconnect; ma_pa_stream_get_state_proc _pa_stream_get_state = pa_stream_get_state; ma_pa_stream_get_sample_spec_proc _pa_stream_get_sample_spec = pa_stream_get_sample_spec; ma_pa_stream_get_channel_map_proc _pa_stream_get_channel_map = pa_stream_get_channel_map; ma_pa_stream_get_buffer_attr_proc _pa_stream_get_buffer_attr = pa_stream_get_buffer_attr; ma_pa_stream_set_buffer_attr_proc _pa_stream_set_buffer_attr = pa_stream_set_buffer_attr; ma_pa_stream_get_device_name_proc _pa_stream_get_device_name = pa_stream_get_device_name; ma_pa_stream_set_write_callback_proc _pa_stream_set_write_callback = pa_stream_set_write_callback; ma_pa_stream_set_read_callback_proc _pa_stream_set_read_callback = pa_stream_set_read_callback; ma_pa_stream_set_suspended_callback_proc _pa_stream_set_suspended_callback = pa_stream_set_suspended_callback; ma_pa_stream_set_moved_callback_proc _pa_stream_set_moved_callback = pa_stream_set_moved_callback; ma_pa_stream_is_suspended_proc _pa_stream_is_suspended = pa_stream_is_suspended; ma_pa_stream_flush_proc _pa_stream_flush = pa_stream_flush; ma_pa_stream_drain_proc _pa_stream_drain = pa_stream_drain; ma_pa_stream_is_corked_proc _pa_stream_is_corked = pa_stream_is_corked; ma_pa_stream_cork_proc _pa_stream_cork = pa_stream_cork; ma_pa_stream_trigger_proc _pa_stream_trigger = pa_stream_trigger; ma_pa_stream_begin_write_proc _pa_stream_begin_write = pa_stream_begin_write; ma_pa_stream_write_proc _pa_stream_write = pa_stream_write; ma_pa_stream_peek_proc _pa_stream_peek = pa_stream_peek; ma_pa_stream_drop_proc _pa_stream_drop = pa_stream_drop; ma_pa_stream_writable_size_proc _pa_stream_writable_size = pa_stream_writable_size; ma_pa_stream_readable_size_proc _pa_stream_readable_size = pa_stream_readable_size; pContext->pulse.pa_mainloop_new = (ma_proc)_pa_mainloop_new; pContext->pulse.pa_mainloop_free = (ma_proc)_pa_mainloop_free; pContext->pulse.pa_mainloop_quit = (ma_proc)_pa_mainloop_quit; pContext->pulse.pa_mainloop_get_api = (ma_proc)_pa_mainloop_get_api; pContext->pulse.pa_mainloop_iterate = (ma_proc)_pa_mainloop_iterate; pContext->pulse.pa_mainloop_wakeup = (ma_proc)_pa_mainloop_wakeup; pContext->pulse.pa_threaded_mainloop_new = (ma_proc)_pa_threaded_mainloop_new; pContext->pulse.pa_threaded_mainloop_free = (ma_proc)_pa_threaded_mainloop_free; pContext->pulse.pa_threaded_mainloop_start = (ma_proc)_pa_threaded_mainloop_start; pContext->pulse.pa_threaded_mainloop_stop = (ma_proc)_pa_threaded_mainloop_stop; pContext->pulse.pa_threaded_mainloop_lock = (ma_proc)_pa_threaded_mainloop_lock; pContext->pulse.pa_threaded_mainloop_unlock = (ma_proc)_pa_threaded_mainloop_unlock; pContext->pulse.pa_threaded_mainloop_wait = (ma_proc)_pa_threaded_mainloop_wait; pContext->pulse.pa_threaded_mainloop_signal = (ma_proc)_pa_threaded_mainloop_signal; pContext->pulse.pa_threaded_mainloop_accept = (ma_proc)_pa_threaded_mainloop_accept; pContext->pulse.pa_threaded_mainloop_get_retval = (ma_proc)_pa_threaded_mainloop_get_retval; pContext->pulse.pa_threaded_mainloop_get_api = (ma_proc)_pa_threaded_mainloop_get_api; pContext->pulse.pa_threaded_mainloop_in_thread = (ma_proc)_pa_threaded_mainloop_in_thread; pContext->pulse.pa_threaded_mainloop_set_name = (ma_proc)_pa_threaded_mainloop_set_name; pContext->pulse.pa_context_new = (ma_proc)_pa_context_new; pContext->pulse.pa_context_unref = (ma_proc)_pa_context_unref; pContext->pulse.pa_context_connect = (ma_proc)_pa_context_connect; pContext->pulse.pa_context_disconnect = (ma_proc)_pa_context_disconnect; pContext->pulse.pa_context_set_state_callback = (ma_proc)_pa_context_set_state_callback; pContext->pulse.pa_context_get_state = (ma_proc)_pa_context_get_state; pContext->pulse.pa_context_get_sink_info_list = (ma_proc)_pa_context_get_sink_info_list; pContext->pulse.pa_context_get_source_info_list = (ma_proc)_pa_context_get_source_info_list; pContext->pulse.pa_context_get_sink_info_by_name = (ma_proc)_pa_context_get_sink_info_by_name; pContext->pulse.pa_context_get_source_info_by_name = (ma_proc)_pa_context_get_source_info_by_name; pContext->pulse.pa_operation_unref = (ma_proc)_pa_operation_unref; pContext->pulse.pa_operation_get_state = (ma_proc)_pa_operation_get_state; pContext->pulse.pa_channel_map_init_extend = (ma_proc)_pa_channel_map_init_extend; pContext->pulse.pa_channel_map_valid = (ma_proc)_pa_channel_map_valid; pContext->pulse.pa_channel_map_compatible = (ma_proc)_pa_channel_map_compatible; pContext->pulse.pa_stream_new = (ma_proc)_pa_stream_new; pContext->pulse.pa_stream_unref = (ma_proc)_pa_stream_unref; pContext->pulse.pa_stream_connect_playback = (ma_proc)_pa_stream_connect_playback; pContext->pulse.pa_stream_connect_record = (ma_proc)_pa_stream_connect_record; pContext->pulse.pa_stream_disconnect = (ma_proc)_pa_stream_disconnect; pContext->pulse.pa_stream_get_state = (ma_proc)_pa_stream_get_state; pContext->pulse.pa_stream_get_sample_spec = (ma_proc)_pa_stream_get_sample_spec; pContext->pulse.pa_stream_get_channel_map = (ma_proc)_pa_stream_get_channel_map; pContext->pulse.pa_stream_get_buffer_attr = (ma_proc)_pa_stream_get_buffer_attr; pContext->pulse.pa_stream_set_buffer_attr = (ma_proc)_pa_stream_set_buffer_attr; pContext->pulse.pa_stream_get_device_name = (ma_proc)_pa_stream_get_device_name; pContext->pulse.pa_stream_set_write_callback = (ma_proc)_pa_stream_set_write_callback; pContext->pulse.pa_stream_set_read_callback = (ma_proc)_pa_stream_set_read_callback; pContext->pulse.pa_stream_set_suspended_callback = (ma_proc)_pa_stream_set_suspended_callback; pContext->pulse.pa_stream_set_moved_callback = (ma_proc)_pa_stream_set_moved_callback; pContext->pulse.pa_stream_is_suspended = (ma_proc)_pa_stream_is_suspended; pContext->pulse.pa_stream_flush = (ma_proc)_pa_stream_flush; pContext->pulse.pa_stream_drain = (ma_proc)_pa_stream_drain; pContext->pulse.pa_stream_is_corked = (ma_proc)_pa_stream_is_corked; pContext->pulse.pa_stream_cork = (ma_proc)_pa_stream_cork; pContext->pulse.pa_stream_trigger = (ma_proc)_pa_stream_trigger; pContext->pulse.pa_stream_begin_write = (ma_proc)_pa_stream_begin_write; pContext->pulse.pa_stream_write = (ma_proc)_pa_stream_write; pContext->pulse.pa_stream_peek = (ma_proc)_pa_stream_peek; pContext->pulse.pa_stream_drop = (ma_proc)_pa_stream_drop; pContext->pulse.pa_stream_writable_size = (ma_proc)_pa_stream_writable_size; pContext->pulse.pa_stream_readable_size = (ma_proc)_pa_stream_readable_size; #endif /* We need to make a copy of the application and server names so we can pass them to the pa_context of each device. */ pContext->pulse.pApplicationName = ma_copy_string(pConfig->pulse.pApplicationName, &pContext->allocationCallbacks); if (pContext->pulse.pApplicationName == NULL && pConfig->pulse.pApplicationName != NULL) { return MA_OUT_OF_MEMORY; } pContext->pulse.pServerName = ma_copy_string(pConfig->pulse.pServerName, &pContext->allocationCallbacks); if (pContext->pulse.pServerName == NULL && pConfig->pulse.pServerName != NULL) { ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks); return MA_OUT_OF_MEMORY; } result = ma_init_pa_mainloop_and_pa_context__pulse(pContext, pConfig->pulse.pApplicationName, pConfig->pulse.pServerName, pConfig->pulse.tryAutoSpawn, &pContext->pulse.pMainLoop, &pContext->pulse.pPulseContext); if (result != MA_SUCCESS) { ma_free(pContext->pulse.pServerName, &pContext->allocationCallbacks); ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks); #ifndef MA_NO_RUNTIME_LINKING ma_dlclose(ma_context_get_log(pContext), pContext->pulse.pulseSO); #endif return result; } /* With pa_mainloop we run a synchronous backend, but we implement our own main loop. */ pCallbacks->onContextInit = ma_context_init__pulse; pCallbacks->onContextUninit = ma_context_uninit__pulse; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__pulse; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__pulse; pCallbacks->onDeviceInit = ma_device_init__pulse; pCallbacks->onDeviceUninit = ma_device_uninit__pulse; pCallbacks->onDeviceStart = ma_device_start__pulse; pCallbacks->onDeviceStop = ma_device_stop__pulse; pCallbacks->onDeviceRead = NULL; /* Not used because we're implementing onDeviceDataLoop. */ pCallbacks->onDeviceWrite = NULL; /* Not used because we're implementing onDeviceDataLoop. */ pCallbacks->onDeviceDataLoop = ma_device_data_loop__pulse; pCallbacks->onDeviceDataLoopWakeup = ma_device_data_loop_wakeup__pulse; return MA_SUCCESS; } #endif /****************************************************************************** JACK Backend ******************************************************************************/ #ifdef MA_HAS_JACK /* It is assumed jack.h is available when compile-time linking is being used. */ #ifdef MA_NO_RUNTIME_LINKING #include <jack/jack.h> typedef jack_nframes_t ma_jack_nframes_t; typedef jack_options_t ma_jack_options_t; typedef jack_status_t ma_jack_status_t; typedef jack_client_t ma_jack_client_t; typedef jack_port_t ma_jack_port_t; typedef JackProcessCallback ma_JackProcessCallback; typedef JackBufferSizeCallback ma_JackBufferSizeCallback; typedef JackShutdownCallback ma_JackShutdownCallback; #define MA_JACK_DEFAULT_AUDIO_TYPE JACK_DEFAULT_AUDIO_TYPE #define ma_JackNullOption JackNullOption #define ma_JackNoStartServer JackNoStartServer #define ma_JackPortIsInput JackPortIsInput #define ma_JackPortIsOutput JackPortIsOutput #define ma_JackPortIsPhysical JackPortIsPhysical #else typedef ma_uint32 ma_jack_nframes_t; typedef int ma_jack_options_t; typedef int ma_jack_status_t; typedef struct ma_jack_client_t ma_jack_client_t; typedef struct ma_jack_port_t ma_jack_port_t; typedef int (* ma_JackProcessCallback) (ma_jack_nframes_t nframes, void* arg); typedef int (* ma_JackBufferSizeCallback)(ma_jack_nframes_t nframes, void* arg); typedef void (* ma_JackShutdownCallback) (void* arg); #define MA_JACK_DEFAULT_AUDIO_TYPE "32 bit float mono audio" #define ma_JackNullOption 0 #define ma_JackNoStartServer 1 #define ma_JackPortIsInput 1 #define ma_JackPortIsOutput 2 #define ma_JackPortIsPhysical 4 #endif typedef ma_jack_client_t* (* ma_jack_client_open_proc) (const char* client_name, ma_jack_options_t options, ma_jack_status_t* status, ...); typedef int (* ma_jack_client_close_proc) (ma_jack_client_t* client); typedef int (* ma_jack_client_name_size_proc) (void); typedef int (* ma_jack_set_process_callback_proc) (ma_jack_client_t* client, ma_JackProcessCallback process_callback, void* arg); typedef int (* ma_jack_set_buffer_size_callback_proc)(ma_jack_client_t* client, ma_JackBufferSizeCallback bufsize_callback, void* arg); typedef void (* ma_jack_on_shutdown_proc) (ma_jack_client_t* client, ma_JackShutdownCallback function, void* arg); typedef ma_jack_nframes_t (* ma_jack_get_sample_rate_proc) (ma_jack_client_t* client); typedef ma_jack_nframes_t (* ma_jack_get_buffer_size_proc) (ma_jack_client_t* client); typedef const char** (* ma_jack_get_ports_proc) (ma_jack_client_t* client, const char* port_name_pattern, const char* type_name_pattern, unsigned long flags); typedef int (* ma_jack_activate_proc) (ma_jack_client_t* client); typedef int (* ma_jack_deactivate_proc) (ma_jack_client_t* client); typedef int (* ma_jack_connect_proc) (ma_jack_client_t* client, const char* source_port, const char* destination_port); typedef ma_jack_port_t* (* ma_jack_port_register_proc) (ma_jack_client_t* client, const char* port_name, const char* port_type, unsigned long flags, unsigned long buffer_size); typedef const char* (* ma_jack_port_name_proc) (const ma_jack_port_t* port); typedef void* (* ma_jack_port_get_buffer_proc) (ma_jack_port_t* port, ma_jack_nframes_t nframes); typedef void (* ma_jack_free_proc) (void* ptr); static ma_result ma_context_open_client__jack(ma_context* pContext, ma_jack_client_t** ppClient) { size_t maxClientNameSize; char clientName[256]; ma_jack_status_t status; ma_jack_client_t* pClient; MA_ASSERT(pContext != NULL); MA_ASSERT(ppClient != NULL); if (ppClient) { *ppClient = NULL; } maxClientNameSize = ((ma_jack_client_name_size_proc)pContext->jack.jack_client_name_size)(); /* Includes null terminator. */ ma_strncpy_s(clientName, ma_min(sizeof(clientName), maxClientNameSize), (pContext->jack.pClientName != NULL) ? pContext->jack.pClientName : "miniaudio", (size_t)-1); pClient = ((ma_jack_client_open_proc)pContext->jack.jack_client_open)(clientName, (pContext->jack.tryStartServer) ? ma_JackNullOption : ma_JackNoStartServer, &status, NULL); if (pClient == NULL) { return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } if (ppClient) { *ppClient = pClient; } return MA_SUCCESS; } static ma_result ma_context_enumerate_devices__jack(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { ma_bool32 cbResult = MA_TRUE; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); /* Playback. */ if (cbResult) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); deviceInfo.isDefault = MA_TRUE; /* JACK only uses default devices. */ cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } /* Capture. */ if (cbResult) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); deviceInfo.isDefault = MA_TRUE; /* JACK only uses default devices. */ cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); } (void)cbResult; /* For silencing a static analysis warning. */ return MA_SUCCESS; } static ma_result ma_context_get_device_info__jack(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { ma_jack_client_t* pClient; ma_result result; const char** ppPorts; MA_ASSERT(pContext != NULL); if (pDeviceID != NULL && pDeviceID->jack != 0) { return MA_NO_DEVICE; /* Don't know the device. */ } /* Name / Description */ if (deviceType == ma_device_type_playback) { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); } else { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); } /* Jack only uses default devices. */ pDeviceInfo->isDefault = MA_TRUE; /* Jack only supports f32 and has a specific channel count and sample rate. */ pDeviceInfo->nativeDataFormats[0].format = ma_format_f32; /* The channel count and sample rate can only be determined by opening the device. */ result = ma_context_open_client__jack(pContext, &pClient); if (result != MA_SUCCESS) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[JACK] Failed to open client."); return result; } pDeviceInfo->nativeDataFormats[0].sampleRate = ((ma_jack_get_sample_rate_proc)pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pClient); pDeviceInfo->nativeDataFormats[0].channels = 0; ppPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ((deviceType == ma_device_type_playback) ? ma_JackPortIsInput : ma_JackPortIsOutput)); if (ppPorts == NULL) { ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient); ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports."); return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } while (ppPorts[pDeviceInfo->nativeDataFormats[0].channels] != NULL) { pDeviceInfo->nativeDataFormats[0].channels += 1; } pDeviceInfo->nativeDataFormats[0].flags = 0; pDeviceInfo->nativeDataFormatCount = 1; ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts); ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient); (void)pContext; return MA_SUCCESS; } static ma_result ma_device_uninit__jack(ma_device* pDevice) { ma_context* pContext; MA_ASSERT(pDevice != NULL); pContext = pDevice->pContext; MA_ASSERT(pContext != NULL); if (pDevice->jack.pClient != NULL) { ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pDevice->jack.pClient); } if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ma_free(pDevice->jack.pIntermediaryBufferCapture, &pDevice->pContext->allocationCallbacks); ma_free(pDevice->jack.ppPortsCapture, &pDevice->pContext->allocationCallbacks); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ma_free(pDevice->jack.pIntermediaryBufferPlayback, &pDevice->pContext->allocationCallbacks); ma_free(pDevice->jack.ppPortsPlayback, &pDevice->pContext->allocationCallbacks); } return MA_SUCCESS; } static void ma_device__jack_shutdown_callback(void* pUserData) { /* JACK died. Stop the device. */ ma_device* pDevice = (ma_device*)pUserData; MA_ASSERT(pDevice != NULL); ma_device_stop(pDevice); } static int ma_device__jack_buffer_size_callback(ma_jack_nframes_t frameCount, void* pUserData) { ma_device* pDevice = (ma_device*)pUserData; MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { size_t newBufferSize = frameCount * (pDevice->capture.internalChannels * ma_get_bytes_per_sample(pDevice->capture.internalFormat)); float* pNewBuffer = (float*)ma_calloc(newBufferSize, &pDevice->pContext->allocationCallbacks); if (pNewBuffer == NULL) { return MA_OUT_OF_MEMORY; } ma_free(pDevice->jack.pIntermediaryBufferCapture, &pDevice->pContext->allocationCallbacks); pDevice->jack.pIntermediaryBufferCapture = pNewBuffer; pDevice->playback.internalPeriodSizeInFrames = frameCount; } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { size_t newBufferSize = frameCount * (pDevice->playback.internalChannels * ma_get_bytes_per_sample(pDevice->playback.internalFormat)); float* pNewBuffer = (float*)ma_calloc(newBufferSize, &pDevice->pContext->allocationCallbacks); if (pNewBuffer == NULL) { return MA_OUT_OF_MEMORY; } ma_free(pDevice->jack.pIntermediaryBufferPlayback, &pDevice->pContext->allocationCallbacks); pDevice->jack.pIntermediaryBufferPlayback = pNewBuffer; pDevice->playback.internalPeriodSizeInFrames = frameCount; } return 0; } static int ma_device__jack_process_callback(ma_jack_nframes_t frameCount, void* pUserData) { ma_device* pDevice; ma_context* pContext; ma_uint32 iChannel; pDevice = (ma_device*)pUserData; MA_ASSERT(pDevice != NULL); pContext = pDevice->pContext; MA_ASSERT(pContext != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { /* Channels need to be interleaved. */ for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) { const float* pSrc = (const float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.ppPortsCapture[iChannel], frameCount); if (pSrc != NULL) { float* pDst = pDevice->jack.pIntermediaryBufferCapture + iChannel; ma_jack_nframes_t iFrame; for (iFrame = 0; iFrame < frameCount; ++iFrame) { *pDst = *pSrc; pDst += pDevice->capture.internalChannels; pSrc += 1; } } } ma_device_handle_backend_data_callback(pDevice, NULL, pDevice->jack.pIntermediaryBufferCapture, frameCount); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ma_device_handle_backend_data_callback(pDevice, pDevice->jack.pIntermediaryBufferPlayback, NULL, frameCount); /* Channels need to be deinterleaved. */ for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) { float* pDst = (float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.ppPortsPlayback[iChannel], frameCount); if (pDst != NULL) { const float* pSrc = pDevice->jack.pIntermediaryBufferPlayback + iChannel; ma_jack_nframes_t iFrame; for (iFrame = 0; iFrame < frameCount; ++iFrame) { *pDst = *pSrc; pDst += 1; pSrc += pDevice->playback.internalChannels; } } } } return 0; } static ma_result ma_device_init__jack(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { ma_result result; ma_uint32 periodSizeInFrames; MA_ASSERT(pConfig != NULL); MA_ASSERT(pDevice != NULL); if (pConfig->deviceType == ma_device_type_loopback) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Loopback mode not supported."); return MA_DEVICE_TYPE_NOT_SUPPORTED; } /* Only supporting default devices with JACK. */ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->pDeviceID != NULL && pDescriptorPlayback->pDeviceID->jack != 0) || ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->pDeviceID != NULL && pDescriptorCapture->pDeviceID->jack != 0)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Only default devices are supported."); return MA_NO_DEVICE; } /* No exclusive mode with the JACK backend. */ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) || ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Exclusive mode not supported."); return MA_SHARE_MODE_NOT_SUPPORTED; } /* Open the client. */ result = ma_context_open_client__jack(pDevice->pContext, (ma_jack_client_t**)&pDevice->jack.pClient); if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to open client."); return result; } /* Callbacks. */ if (((ma_jack_set_process_callback_proc)pDevice->pContext->jack.jack_set_process_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_process_callback, pDevice) != 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to set process callback."); return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } if (((ma_jack_set_buffer_size_callback_proc)pDevice->pContext->jack.jack_set_buffer_size_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_buffer_size_callback, pDevice) != 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to set buffer size callback."); return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } ((ma_jack_on_shutdown_proc)pDevice->pContext->jack.jack_on_shutdown)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_shutdown_callback, pDevice); /* The buffer size in frames can change. */ periodSizeInFrames = ((ma_jack_get_buffer_size_proc)pDevice->pContext->jack.jack_get_buffer_size)((ma_jack_client_t*)pDevice->jack.pClient); if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { ma_uint32 iPort; const char** ppPorts; pDescriptorCapture->format = ma_format_f32; pDescriptorCapture->channels = 0; pDescriptorCapture->sampleRate = ((ma_jack_get_sample_rate_proc)pDevice->pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient); ma_channel_map_init_standard(ma_standard_channel_map_alsa, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorCapture->channels); ppPorts = ((ma_jack_get_ports_proc)pDevice->pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsOutput); if (ppPorts == NULL) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports."); return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } /* Need to count the number of ports first so we can allocate some memory. */ while (ppPorts[pDescriptorCapture->channels] != NULL) { pDescriptorCapture->channels += 1; } pDevice->jack.ppPortsCapture = (ma_ptr*)ma_malloc(sizeof(*pDevice->jack.ppPortsCapture) * pDescriptorCapture->channels, &pDevice->pContext->allocationCallbacks); if (pDevice->jack.ppPortsCapture == NULL) { return MA_OUT_OF_MEMORY; } for (iPort = 0; iPort < pDescriptorCapture->channels; iPort += 1) { char name[64]; ma_strcpy_s(name, sizeof(name), "capture"); ma_itoa_s((int)iPort, name+7, sizeof(name)-7, 10); /* 7 = length of "capture" */ pDevice->jack.ppPortsCapture[iPort] = ((ma_jack_port_register_proc)pDevice->pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsInput, 0); if (pDevice->jack.ppPortsCapture[iPort] == NULL) { ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts); ma_device_uninit__jack(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports."); return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } } ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts); pDescriptorCapture->periodSizeInFrames = periodSizeInFrames; pDescriptorCapture->periodCount = 1; /* There's no notion of a period in JACK. Just set to 1. */ pDevice->jack.pIntermediaryBufferCapture = (float*)ma_calloc(pDescriptorCapture->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels), &pDevice->pContext->allocationCallbacks); if (pDevice->jack.pIntermediaryBufferCapture == NULL) { ma_device_uninit__jack(pDevice); return MA_OUT_OF_MEMORY; } } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { ma_uint32 iPort; const char** ppPorts; pDescriptorPlayback->format = ma_format_f32; pDescriptorPlayback->channels = 0; pDescriptorPlayback->sampleRate = ((ma_jack_get_sample_rate_proc)pDevice->pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient); ma_channel_map_init_standard(ma_standard_channel_map_alsa, pDescriptorPlayback->channelMap, ma_countof(pDescriptorPlayback->channelMap), pDescriptorPlayback->channels); ppPorts = ((ma_jack_get_ports_proc)pDevice->pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsInput); if (ppPorts == NULL) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports."); return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } /* Need to count the number of ports first so we can allocate some memory. */ while (ppPorts[pDescriptorPlayback->channels] != NULL) { pDescriptorPlayback->channels += 1; } pDevice->jack.ppPortsPlayback = (ma_ptr*)ma_malloc(sizeof(*pDevice->jack.ppPortsPlayback) * pDescriptorPlayback->channels, &pDevice->pContext->allocationCallbacks); if (pDevice->jack.ppPortsPlayback == NULL) { ma_free(pDevice->jack.ppPortsCapture, &pDevice->pContext->allocationCallbacks); return MA_OUT_OF_MEMORY; } for (iPort = 0; iPort < pDescriptorPlayback->channels; iPort += 1) { char name[64]; ma_strcpy_s(name, sizeof(name), "playback"); ma_itoa_s((int)iPort, name+8, sizeof(name)-8, 10); /* 8 = length of "playback" */ pDevice->jack.ppPortsPlayback[iPort] = ((ma_jack_port_register_proc)pDevice->pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsOutput, 0); if (pDevice->jack.ppPortsPlayback[iPort] == NULL) { ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts); ma_device_uninit__jack(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports."); return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } } ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts); pDescriptorPlayback->periodSizeInFrames = periodSizeInFrames; pDescriptorPlayback->periodCount = 1; /* There's no notion of a period in JACK. Just set to 1. */ pDevice->jack.pIntermediaryBufferPlayback = (float*)ma_calloc(pDescriptorPlayback->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels), &pDevice->pContext->allocationCallbacks); if (pDevice->jack.pIntermediaryBufferPlayback == NULL) { ma_device_uninit__jack(pDevice); return MA_OUT_OF_MEMORY; } } return MA_SUCCESS; } static ma_result ma_device_start__jack(ma_device* pDevice) { ma_context* pContext = pDevice->pContext; int resultJACK; size_t i; resultJACK = ((ma_jack_activate_proc)pContext->jack.jack_activate)((ma_jack_client_t*)pDevice->jack.pClient); if (resultJACK != 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to activate the JACK client."); return MA_FAILED_TO_START_BACKEND_DEVICE; } if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsOutput); if (ppServerPorts == NULL) { ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports."); return MA_ERROR; } for (i = 0; ppServerPorts[i] != NULL; ++i) { const char* pServerPort = ppServerPorts[i]; const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.ppPortsCapture[i]); resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pServerPort, pClientPort); if (resultJACK != 0) { ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to connect ports."); return MA_ERROR; } } ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsInput); if (ppServerPorts == NULL) { ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports."); return MA_ERROR; } for (i = 0; ppServerPorts[i] != NULL; ++i) { const char* pServerPort = ppServerPorts[i]; const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.ppPortsPlayback[i]); resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pClientPort, pServerPort); if (resultJACK != 0) { ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to connect ports."); return MA_ERROR; } } ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); } return MA_SUCCESS; } static ma_result ma_device_stop__jack(ma_device* pDevice) { ma_context* pContext = pDevice->pContext; if (((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient) != 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] An error occurred when deactivating the JACK client."); return MA_ERROR; } ma_device__on_notification_stopped(pDevice); return MA_SUCCESS; } static ma_result ma_context_uninit__jack(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_jack); ma_free(pContext->jack.pClientName, &pContext->allocationCallbacks); pContext->jack.pClientName = NULL; #ifndef MA_NO_RUNTIME_LINKING ma_dlclose(ma_context_get_log(pContext), pContext->jack.jackSO); #endif return MA_SUCCESS; } static ma_result ma_context_init__jack(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { #ifndef MA_NO_RUNTIME_LINKING const char* libjackNames[] = { #if defined(MA_WIN32) "libjack.dll", "libjack64.dll" #endif #if defined(MA_UNIX) "libjack.so", "libjack.so.0" #endif }; size_t i; for (i = 0; i < ma_countof(libjackNames); ++i) { pContext->jack.jackSO = ma_dlopen(ma_context_get_log(pContext), libjackNames[i]); if (pContext->jack.jackSO != NULL) { break; } } if (pContext->jack.jackSO == NULL) { return MA_NO_BACKEND; } pContext->jack.jack_client_open = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_client_open"); pContext->jack.jack_client_close = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_client_close"); pContext->jack.jack_client_name_size = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_client_name_size"); pContext->jack.jack_set_process_callback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_set_process_callback"); pContext->jack.jack_set_buffer_size_callback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_set_buffer_size_callback"); pContext->jack.jack_on_shutdown = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_on_shutdown"); pContext->jack.jack_get_sample_rate = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_get_sample_rate"); pContext->jack.jack_get_buffer_size = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_get_buffer_size"); pContext->jack.jack_get_ports = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_get_ports"); pContext->jack.jack_activate = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_activate"); pContext->jack.jack_deactivate = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_deactivate"); pContext->jack.jack_connect = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_connect"); pContext->jack.jack_port_register = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_port_register"); pContext->jack.jack_port_name = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_port_name"); pContext->jack.jack_port_get_buffer = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_port_get_buffer"); pContext->jack.jack_free = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->jack.jackSO, "jack_free"); #else /* This strange assignment system is here just to ensure type safety of miniaudio's function pointer types. If anything differs slightly the compiler should throw a warning. */ ma_jack_client_open_proc _jack_client_open = jack_client_open; ma_jack_client_close_proc _jack_client_close = jack_client_close; ma_jack_client_name_size_proc _jack_client_name_size = jack_client_name_size; ma_jack_set_process_callback_proc _jack_set_process_callback = jack_set_process_callback; ma_jack_set_buffer_size_callback_proc _jack_set_buffer_size_callback = jack_set_buffer_size_callback; ma_jack_on_shutdown_proc _jack_on_shutdown = jack_on_shutdown; ma_jack_get_sample_rate_proc _jack_get_sample_rate = jack_get_sample_rate; ma_jack_get_buffer_size_proc _jack_get_buffer_size = jack_get_buffer_size; ma_jack_get_ports_proc _jack_get_ports = jack_get_ports; ma_jack_activate_proc _jack_activate = jack_activate; ma_jack_deactivate_proc _jack_deactivate = jack_deactivate; ma_jack_connect_proc _jack_connect = jack_connect; ma_jack_port_register_proc _jack_port_register = jack_port_register; ma_jack_port_name_proc _jack_port_name = jack_port_name; ma_jack_port_get_buffer_proc _jack_port_get_buffer = jack_port_get_buffer; ma_jack_free_proc _jack_free = jack_free; pContext->jack.jack_client_open = (ma_proc)_jack_client_open; pContext->jack.jack_client_close = (ma_proc)_jack_client_close; pContext->jack.jack_client_name_size = (ma_proc)_jack_client_name_size; pContext->jack.jack_set_process_callback = (ma_proc)_jack_set_process_callback; pContext->jack.jack_set_buffer_size_callback = (ma_proc)_jack_set_buffer_size_callback; pContext->jack.jack_on_shutdown = (ma_proc)_jack_on_shutdown; pContext->jack.jack_get_sample_rate = (ma_proc)_jack_get_sample_rate; pContext->jack.jack_get_buffer_size = (ma_proc)_jack_get_buffer_size; pContext->jack.jack_get_ports = (ma_proc)_jack_get_ports; pContext->jack.jack_activate = (ma_proc)_jack_activate; pContext->jack.jack_deactivate = (ma_proc)_jack_deactivate; pContext->jack.jack_connect = (ma_proc)_jack_connect; pContext->jack.jack_port_register = (ma_proc)_jack_port_register; pContext->jack.jack_port_name = (ma_proc)_jack_port_name; pContext->jack.jack_port_get_buffer = (ma_proc)_jack_port_get_buffer; pContext->jack.jack_free = (ma_proc)_jack_free; #endif if (pConfig->jack.pClientName != NULL) { pContext->jack.pClientName = ma_copy_string(pConfig->jack.pClientName, &pContext->allocationCallbacks); } pContext->jack.tryStartServer = pConfig->jack.tryStartServer; /* Getting here means the JACK library is installed, but it doesn't necessarily mean it's usable. We need to quickly test this by connecting a temporary client. */ { ma_jack_client_t* pDummyClient; ma_result result = ma_context_open_client__jack(pContext, &pDummyClient); if (result != MA_SUCCESS) { ma_free(pContext->jack.pClientName, &pContext->allocationCallbacks); #ifndef MA_NO_RUNTIME_LINKING ma_dlclose(ma_context_get_log(pContext), pContext->jack.jackSO); #endif return MA_NO_BACKEND; } ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pDummyClient); } pCallbacks->onContextInit = ma_context_init__jack; pCallbacks->onContextUninit = ma_context_uninit__jack; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__jack; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__jack; pCallbacks->onDeviceInit = ma_device_init__jack; pCallbacks->onDeviceUninit = ma_device_uninit__jack; pCallbacks->onDeviceStart = ma_device_start__jack; pCallbacks->onDeviceStop = ma_device_stop__jack; pCallbacks->onDeviceRead = NULL; /* Not used because JACK is asynchronous. */ pCallbacks->onDeviceWrite = NULL; /* Not used because JACK is asynchronous. */ pCallbacks->onDeviceDataLoop = NULL; /* Not used because JACK is asynchronous. */ return MA_SUCCESS; } #endif /* MA_HAS_JACK */ /****************************************************************************** Core Audio Backend References ========== - Technical Note TN2091: Device input using the HAL Output Audio Unit https://developer.apple.com/library/archive/technotes/tn2091/_index.html ******************************************************************************/ #ifdef MA_HAS_COREAUDIO #include <TargetConditionals.h> #if defined(TARGET_OS_IPHONE) && TARGET_OS_IPHONE == 1 #define MA_APPLE_MOBILE #if defined(TARGET_OS_TV) && TARGET_OS_TV == 1 #define MA_APPLE_TV #endif #if defined(TARGET_OS_WATCH) && TARGET_OS_WATCH == 1 #define MA_APPLE_WATCH #endif #if __has_feature(objc_arc) #define MA_BRIDGE_TRANSFER __bridge_transfer #define MA_BRIDGE_RETAINED __bridge_retained #else #define MA_BRIDGE_TRANSFER #define MA_BRIDGE_RETAINED #endif #else #define MA_APPLE_DESKTOP #endif #if defined(MA_APPLE_DESKTOP) #include <CoreAudio/CoreAudio.h> #else #include <AVFoundation/AVFoundation.h> #endif #include <AudioToolbox/AudioToolbox.h> /* CoreFoundation */ typedef Boolean (* ma_CFStringGetCString_proc)(CFStringRef theString, char* buffer, CFIndex bufferSize, CFStringEncoding encoding); typedef void (* ma_CFRelease_proc)(CFTypeRef cf); /* CoreAudio */ #if defined(MA_APPLE_DESKTOP) typedef OSStatus (* ma_AudioObjectGetPropertyData_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32* ioDataSize, void* outData); typedef OSStatus (* ma_AudioObjectGetPropertyDataSize_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32* outDataSize); typedef OSStatus (* ma_AudioObjectSetPropertyData_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32 inDataSize, const void* inData); typedef OSStatus (* ma_AudioObjectAddPropertyListener_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, AudioObjectPropertyListenerProc inListener, void* inClientData); typedef OSStatus (* ma_AudioObjectRemovePropertyListener_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, AudioObjectPropertyListenerProc inListener, void* inClientData); #endif /* AudioToolbox */ typedef AudioComponent (* ma_AudioComponentFindNext_proc)(AudioComponent inComponent, const AudioComponentDescription* inDesc); typedef OSStatus (* ma_AudioComponentInstanceDispose_proc)(AudioComponentInstance inInstance); typedef OSStatus (* ma_AudioComponentInstanceNew_proc)(AudioComponent inComponent, AudioComponentInstance* outInstance); typedef OSStatus (* ma_AudioOutputUnitStart_proc)(AudioUnit inUnit); typedef OSStatus (* ma_AudioOutputUnitStop_proc)(AudioUnit inUnit); typedef OSStatus (* ma_AudioUnitAddPropertyListener_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitPropertyListenerProc inProc, void* inProcUserData); typedef OSStatus (* ma_AudioUnitGetPropertyInfo_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, UInt32* outDataSize, Boolean* outWriteable); typedef OSStatus (* ma_AudioUnitGetProperty_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void* outData, UInt32* ioDataSize); typedef OSStatus (* ma_AudioUnitSetProperty_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, const void* inData, UInt32 inDataSize); typedef OSStatus (* ma_AudioUnitInitialize_proc)(AudioUnit inUnit); typedef OSStatus (* ma_AudioUnitRender_proc)(AudioUnit inUnit, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inOutputBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData); #define MA_COREAUDIO_OUTPUT_BUS 0 #define MA_COREAUDIO_INPUT_BUS 1 #if defined(MA_APPLE_DESKTOP) static ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_device_type deviceType, ma_bool32 disposePreviousAudioUnit); #endif /* Core Audio So far, Core Audio has been the worst backend to work with due to being both unintuitive and having almost no documentation apart from comments in the headers (which admittedly are quite good). For my own purposes, and for anybody out there whose needing to figure out how this darn thing works, I'm going to outline a few things here. Since miniaudio is a fairly low-level API, one of the things it needs is control over specific devices, and it needs to be able to identify whether or not it can be used as playback and/or capture. The AudioObject API is the only one I've seen that supports this level of detail. There was some public domain sample code I stumbled across that used the AudioComponent and AudioUnit APIs, but I couldn't see anything that gave low-level control over device selection and capabilities (the distinction between playback and capture in particular). Therefore, miniaudio is using the AudioObject API. Most (all?) functions in the AudioObject API take a AudioObjectID as its input. This is the device identifier. When retrieving global information, such as the device list, you use kAudioObjectSystemObject. When retrieving device-specific data, you pass in the ID for that device. In order to retrieve device-specific IDs you need to enumerate over each of the devices. This is done using the AudioObjectGetPropertyDataSize() and AudioObjectGetPropertyData() APIs which seem to be the central APIs for retrieving information about the system and specific devices. To use the AudioObjectGetPropertyData() API you need to use the notion of a property address. A property address is a structure with three variables and is used to identify which property you are getting or setting. The first is the "selector" which is basically the specific property that you're wanting to retrieve or set. The second is the "scope", which is typically set to kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyScopeInput for input-specific properties and kAudioObjectPropertyScopeOutput for output-specific properties. The last is the "element" which is always set to kAudioObjectPropertyElementMain in miniaudio's case. I don't know of any cases where this would be set to anything different. Back to the earlier issue of device retrieval, you first use the AudioObjectGetPropertyDataSize() API to retrieve the size of the raw data which is just a list of AudioDeviceID's. You use the kAudioObjectSystemObject AudioObjectID, and a property address with the kAudioHardwarePropertyDevices selector and the kAudioObjectPropertyScopeGlobal scope. Once you have the size, allocate a block of memory of that size and then call AudioObjectGetPropertyData(). The data is just a list of AudioDeviceID's so just do "dataSize/sizeof(AudioDeviceID)" to know the device count. */ #if defined(MA_APPLE_MOBILE) static void ma_device__on_notification_interruption_began(ma_device* pDevice) { ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_interruption_began)); } static void ma_device__on_notification_interruption_ended(ma_device* pDevice) { ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_interruption_ended)); } #endif static ma_result ma_result_from_OSStatus(OSStatus status) { switch (status) { case noErr: return MA_SUCCESS; #if defined(MA_APPLE_DESKTOP) case kAudioHardwareNotRunningError: return MA_DEVICE_NOT_STARTED; case kAudioHardwareUnspecifiedError: return MA_ERROR; case kAudioHardwareUnknownPropertyError: return MA_INVALID_ARGS; case kAudioHardwareBadPropertySizeError: return MA_INVALID_OPERATION; case kAudioHardwareIllegalOperationError: return MA_INVALID_OPERATION; case kAudioHardwareBadObjectError: return MA_INVALID_ARGS; case kAudioHardwareBadDeviceError: return MA_INVALID_ARGS; case kAudioHardwareBadStreamError: return MA_INVALID_ARGS; case kAudioHardwareUnsupportedOperationError: return MA_INVALID_OPERATION; case kAudioDeviceUnsupportedFormatError: return MA_FORMAT_NOT_SUPPORTED; case kAudioDevicePermissionsError: return MA_ACCESS_DENIED; #endif default: return MA_ERROR; } } #if 0 static ma_channel ma_channel_from_AudioChannelBitmap(AudioChannelBitmap bit) { switch (bit) { case kAudioChannelBit_Left: return MA_CHANNEL_LEFT; case kAudioChannelBit_Right: return MA_CHANNEL_RIGHT; case kAudioChannelBit_Center: return MA_CHANNEL_FRONT_CENTER; case kAudioChannelBit_LFEScreen: return MA_CHANNEL_LFE; case kAudioChannelBit_LeftSurround: return MA_CHANNEL_BACK_LEFT; case kAudioChannelBit_RightSurround: return MA_CHANNEL_BACK_RIGHT; case kAudioChannelBit_LeftCenter: return MA_CHANNEL_FRONT_LEFT_CENTER; case kAudioChannelBit_RightCenter: return MA_CHANNEL_FRONT_RIGHT_CENTER; case kAudioChannelBit_CenterSurround: return MA_CHANNEL_BACK_CENTER; case kAudioChannelBit_LeftSurroundDirect: return MA_CHANNEL_SIDE_LEFT; case kAudioChannelBit_RightSurroundDirect: return MA_CHANNEL_SIDE_RIGHT; case kAudioChannelBit_TopCenterSurround: return MA_CHANNEL_TOP_CENTER; case kAudioChannelBit_VerticalHeightLeft: return MA_CHANNEL_TOP_FRONT_LEFT; case kAudioChannelBit_VerticalHeightCenter: return MA_CHANNEL_TOP_FRONT_CENTER; case kAudioChannelBit_VerticalHeightRight: return MA_CHANNEL_TOP_FRONT_RIGHT; case kAudioChannelBit_TopBackLeft: return MA_CHANNEL_TOP_BACK_LEFT; case kAudioChannelBit_TopBackCenter: return MA_CHANNEL_TOP_BACK_CENTER; case kAudioChannelBit_TopBackRight: return MA_CHANNEL_TOP_BACK_RIGHT; default: return MA_CHANNEL_NONE; } } #endif static ma_result ma_format_from_AudioStreamBasicDescription(const AudioStreamBasicDescription* pDescription, ma_format* pFormatOut) { MA_ASSERT(pDescription != NULL); MA_ASSERT(pFormatOut != NULL); *pFormatOut = ma_format_unknown; /* Safety. */ /* There's a few things miniaudio doesn't support. */ if (pDescription->mFormatID != kAudioFormatLinearPCM) { return MA_FORMAT_NOT_SUPPORTED; } /* We don't support any non-packed formats that are aligned high. */ if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsAlignedHigh) != 0) { return MA_FORMAT_NOT_SUPPORTED; } /* Only supporting native-endian. */ if ((ma_is_little_endian() && (pDescription->mFormatFlags & kAudioFormatFlagIsBigEndian) != 0) || (ma_is_big_endian() && (pDescription->mFormatFlags & kAudioFormatFlagIsBigEndian) == 0)) { return MA_FORMAT_NOT_SUPPORTED; } /* We are not currently supporting non-interleaved formats (this will be added in a future version of miniaudio). */ /*if ((pDescription->mFormatFlags & kAudioFormatFlagIsNonInterleaved) != 0) { return MA_FORMAT_NOT_SUPPORTED; }*/ if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsFloat) != 0) { if (pDescription->mBitsPerChannel == 32) { *pFormatOut = ma_format_f32; return MA_SUCCESS; } } else { if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsSignedInteger) != 0) { if (pDescription->mBitsPerChannel == 16) { *pFormatOut = ma_format_s16; return MA_SUCCESS; } else if (pDescription->mBitsPerChannel == 24) { if (pDescription->mBytesPerFrame == (pDescription->mBitsPerChannel/8 * pDescription->mChannelsPerFrame)) { *pFormatOut = ma_format_s24; return MA_SUCCESS; } else { if (pDescription->mBytesPerFrame/pDescription->mChannelsPerFrame == sizeof(ma_int32)) { /* TODO: Implement ma_format_s24_32. */ /**pFormatOut = ma_format_s24_32;*/ /*return MA_SUCCESS;*/ return MA_FORMAT_NOT_SUPPORTED; } } } else if (pDescription->mBitsPerChannel == 32) { *pFormatOut = ma_format_s32; return MA_SUCCESS; } } else { if (pDescription->mBitsPerChannel == 8) { *pFormatOut = ma_format_u8; return MA_SUCCESS; } } } /* Getting here means the format is not supported. */ return MA_FORMAT_NOT_SUPPORTED; } #if defined(MA_APPLE_DESKTOP) static ma_channel ma_channel_from_AudioChannelLabel(AudioChannelLabel label) { switch (label) { case kAudioChannelLabel_Unknown: return MA_CHANNEL_NONE; case kAudioChannelLabel_Unused: return MA_CHANNEL_NONE; case kAudioChannelLabel_UseCoordinates: return MA_CHANNEL_NONE; case kAudioChannelLabel_Left: return MA_CHANNEL_LEFT; case kAudioChannelLabel_Right: return MA_CHANNEL_RIGHT; case kAudioChannelLabel_Center: return MA_CHANNEL_FRONT_CENTER; case kAudioChannelLabel_LFEScreen: return MA_CHANNEL_LFE; case kAudioChannelLabel_LeftSurround: return MA_CHANNEL_BACK_LEFT; case kAudioChannelLabel_RightSurround: return MA_CHANNEL_BACK_RIGHT; case kAudioChannelLabel_LeftCenter: return MA_CHANNEL_FRONT_LEFT_CENTER; case kAudioChannelLabel_RightCenter: return MA_CHANNEL_FRONT_RIGHT_CENTER; case kAudioChannelLabel_CenterSurround: return MA_CHANNEL_BACK_CENTER; case kAudioChannelLabel_LeftSurroundDirect: return MA_CHANNEL_SIDE_LEFT; case kAudioChannelLabel_RightSurroundDirect: return MA_CHANNEL_SIDE_RIGHT; case kAudioChannelLabel_TopCenterSurround: return MA_CHANNEL_TOP_CENTER; case kAudioChannelLabel_VerticalHeightLeft: return MA_CHANNEL_TOP_FRONT_LEFT; case kAudioChannelLabel_VerticalHeightCenter: return MA_CHANNEL_TOP_FRONT_CENTER; case kAudioChannelLabel_VerticalHeightRight: return MA_CHANNEL_TOP_FRONT_RIGHT; case kAudioChannelLabel_TopBackLeft: return MA_CHANNEL_TOP_BACK_LEFT; case kAudioChannelLabel_TopBackCenter: return MA_CHANNEL_TOP_BACK_CENTER; case kAudioChannelLabel_TopBackRight: return MA_CHANNEL_TOP_BACK_RIGHT; case kAudioChannelLabel_RearSurroundLeft: return MA_CHANNEL_BACK_LEFT; case kAudioChannelLabel_RearSurroundRight: return MA_CHANNEL_BACK_RIGHT; case kAudioChannelLabel_LeftWide: return MA_CHANNEL_SIDE_LEFT; case kAudioChannelLabel_RightWide: return MA_CHANNEL_SIDE_RIGHT; case kAudioChannelLabel_LFE2: return MA_CHANNEL_LFE; case kAudioChannelLabel_LeftTotal: return MA_CHANNEL_LEFT; case kAudioChannelLabel_RightTotal: return MA_CHANNEL_RIGHT; case kAudioChannelLabel_HearingImpaired: return MA_CHANNEL_NONE; case kAudioChannelLabel_Narration: return MA_CHANNEL_MONO; case kAudioChannelLabel_Mono: return MA_CHANNEL_MONO; case kAudioChannelLabel_DialogCentricMix: return MA_CHANNEL_MONO; case kAudioChannelLabel_CenterSurroundDirect: return MA_CHANNEL_BACK_CENTER; case kAudioChannelLabel_Haptic: return MA_CHANNEL_NONE; case kAudioChannelLabel_Ambisonic_W: return MA_CHANNEL_NONE; case kAudioChannelLabel_Ambisonic_X: return MA_CHANNEL_NONE; case kAudioChannelLabel_Ambisonic_Y: return MA_CHANNEL_NONE; case kAudioChannelLabel_Ambisonic_Z: return MA_CHANNEL_NONE; case kAudioChannelLabel_MS_Mid: return MA_CHANNEL_LEFT; case kAudioChannelLabel_MS_Side: return MA_CHANNEL_RIGHT; case kAudioChannelLabel_XY_X: return MA_CHANNEL_LEFT; case kAudioChannelLabel_XY_Y: return MA_CHANNEL_RIGHT; case kAudioChannelLabel_HeadphonesLeft: return MA_CHANNEL_LEFT; case kAudioChannelLabel_HeadphonesRight: return MA_CHANNEL_RIGHT; case kAudioChannelLabel_ClickTrack: return MA_CHANNEL_NONE; case kAudioChannelLabel_ForeignLanguage: return MA_CHANNEL_NONE; case kAudioChannelLabel_Discrete: return MA_CHANNEL_NONE; case kAudioChannelLabel_Discrete_0: return MA_CHANNEL_AUX_0; case kAudioChannelLabel_Discrete_1: return MA_CHANNEL_AUX_1; case kAudioChannelLabel_Discrete_2: return MA_CHANNEL_AUX_2; case kAudioChannelLabel_Discrete_3: return MA_CHANNEL_AUX_3; case kAudioChannelLabel_Discrete_4: return MA_CHANNEL_AUX_4; case kAudioChannelLabel_Discrete_5: return MA_CHANNEL_AUX_5; case kAudioChannelLabel_Discrete_6: return MA_CHANNEL_AUX_6; case kAudioChannelLabel_Discrete_7: return MA_CHANNEL_AUX_7; case kAudioChannelLabel_Discrete_8: return MA_CHANNEL_AUX_8; case kAudioChannelLabel_Discrete_9: return MA_CHANNEL_AUX_9; case kAudioChannelLabel_Discrete_10: return MA_CHANNEL_AUX_10; case kAudioChannelLabel_Discrete_11: return MA_CHANNEL_AUX_11; case kAudioChannelLabel_Discrete_12: return MA_CHANNEL_AUX_12; case kAudioChannelLabel_Discrete_13: return MA_CHANNEL_AUX_13; case kAudioChannelLabel_Discrete_14: return MA_CHANNEL_AUX_14; case kAudioChannelLabel_Discrete_15: return MA_CHANNEL_AUX_15; case kAudioChannelLabel_Discrete_65535: return MA_CHANNEL_NONE; #if 0 /* Introduced in a later version of macOS. */ case kAudioChannelLabel_HOA_ACN: return MA_CHANNEL_NONE; case kAudioChannelLabel_HOA_ACN_0: return MA_CHANNEL_AUX_0; case kAudioChannelLabel_HOA_ACN_1: return MA_CHANNEL_AUX_1; case kAudioChannelLabel_HOA_ACN_2: return MA_CHANNEL_AUX_2; case kAudioChannelLabel_HOA_ACN_3: return MA_CHANNEL_AUX_3; case kAudioChannelLabel_HOA_ACN_4: return MA_CHANNEL_AUX_4; case kAudioChannelLabel_HOA_ACN_5: return MA_CHANNEL_AUX_5; case kAudioChannelLabel_HOA_ACN_6: return MA_CHANNEL_AUX_6; case kAudioChannelLabel_HOA_ACN_7: return MA_CHANNEL_AUX_7; case kAudioChannelLabel_HOA_ACN_8: return MA_CHANNEL_AUX_8; case kAudioChannelLabel_HOA_ACN_9: return MA_CHANNEL_AUX_9; case kAudioChannelLabel_HOA_ACN_10: return MA_CHANNEL_AUX_10; case kAudioChannelLabel_HOA_ACN_11: return MA_CHANNEL_AUX_11; case kAudioChannelLabel_HOA_ACN_12: return MA_CHANNEL_AUX_12; case kAudioChannelLabel_HOA_ACN_13: return MA_CHANNEL_AUX_13; case kAudioChannelLabel_HOA_ACN_14: return MA_CHANNEL_AUX_14; case kAudioChannelLabel_HOA_ACN_15: return MA_CHANNEL_AUX_15; case kAudioChannelLabel_HOA_ACN_65024: return MA_CHANNEL_NONE; #endif default: return MA_CHANNEL_NONE; } } static ma_result ma_get_channel_map_from_AudioChannelLayout(AudioChannelLayout* pChannelLayout, ma_channel* pChannelMap, size_t channelMapCap) { MA_ASSERT(pChannelLayout != NULL); if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelDescriptions) { UInt32 iChannel; for (iChannel = 0; iChannel < pChannelLayout->mNumberChannelDescriptions && iChannel < channelMapCap; ++iChannel) { pChannelMap[iChannel] = ma_channel_from_AudioChannelLabel(pChannelLayout->mChannelDescriptions[iChannel].mChannelLabel); } } else #if 0 if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelBitmap) { /* This is the same kind of system that's used by Windows audio APIs. */ UInt32 iChannel = 0; UInt32 iBit; AudioChannelBitmap bitmap = pChannelLayout->mChannelBitmap; for (iBit = 0; iBit < 32 && iChannel < channelMapCap; ++iBit) { AudioChannelBitmap bit = bitmap & (1 << iBit); if (bit != 0) { pChannelMap[iChannel++] = ma_channel_from_AudioChannelBit(bit); } } } else #endif { /* Need to use the tag to determine the channel map. For now I'm just assuming a default channel map, but later on this should be updated to determine the mapping based on the tag. */ UInt32 channelCount; /* Our channel map retrieval APIs below take 32-bit integers, so we'll want to clamp the channel map capacity. */ if (channelMapCap > 0xFFFFFFFF) { channelMapCap = 0xFFFFFFFF; } channelCount = ma_min(AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag), (UInt32)channelMapCap); switch (pChannelLayout->mChannelLayoutTag) { case kAudioChannelLayoutTag_Mono: case kAudioChannelLayoutTag_Stereo: case kAudioChannelLayoutTag_StereoHeadphones: case kAudioChannelLayoutTag_MatrixStereo: case kAudioChannelLayoutTag_MidSide: case kAudioChannelLayoutTag_XY: case kAudioChannelLayoutTag_Binaural: case kAudioChannelLayoutTag_Ambisonic_B_Format: { ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, channelCount); } break; case kAudioChannelLayoutTag_Octagonal: { pChannelMap[7] = MA_CHANNEL_SIDE_RIGHT; pChannelMap[6] = MA_CHANNEL_SIDE_LEFT; } MA_FALLTHROUGH; /* Intentional fallthrough. */ case kAudioChannelLayoutTag_Hexagonal: { pChannelMap[5] = MA_CHANNEL_BACK_CENTER; } MA_FALLTHROUGH; /* Intentional fallthrough. */ case kAudioChannelLayoutTag_Pentagonal: { pChannelMap[4] = MA_CHANNEL_FRONT_CENTER; } MA_FALLTHROUGH; /* Intentional fallthrough. */ case kAudioChannelLayoutTag_Quadraphonic: { pChannelMap[3] = MA_CHANNEL_BACK_RIGHT; pChannelMap[2] = MA_CHANNEL_BACK_LEFT; pChannelMap[1] = MA_CHANNEL_RIGHT; pChannelMap[0] = MA_CHANNEL_LEFT; } break; /* TODO: Add support for more tags here. */ default: { ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, channelCount); } break; } } return MA_SUCCESS; } #if (defined(MAC_OS_VERSION_12_0) && MAC_OS_X_VERSION_MAX_ALLOWED >= MAC_OS_VERSION_12_0) || \ (defined(__IPHONE_15_0) && __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_15_0) #define AUDIO_OBJECT_PROPERTY_ELEMENT kAudioObjectPropertyElementMain #else /* kAudioObjectPropertyElementMaster is deprecated. */ #define AUDIO_OBJECT_PROPERTY_ELEMENT kAudioObjectPropertyElementMaster #endif /* kAudioDevicePropertyScope* were renamed to kAudioObjectPropertyScope* in 10.8. */ #if !defined(MAC_OS_X_VERSION_10_8) || (MAC_OS_X_VERSION_MIN_REQUIRED < MAC_OS_X_VERSION_10_8) #define kAudioObjectPropertyScopeInput kAudioDevicePropertyScopeInput #define kAudioObjectPropertyScopeOutput kAudioDevicePropertyScopeOutput #endif static ma_result ma_get_device_object_ids__coreaudio(ma_context* pContext, UInt32* pDeviceCount, AudioObjectID** ppDeviceObjectIDs) /* NOTE: Free the returned buffer with ma_free(). */ { AudioObjectPropertyAddress propAddressDevices; UInt32 deviceObjectsDataSize; OSStatus status; AudioObjectID* pDeviceObjectIDs; MA_ASSERT(pContext != NULL); MA_ASSERT(pDeviceCount != NULL); MA_ASSERT(ppDeviceObjectIDs != NULL); /* Safety. */ *pDeviceCount = 0; *ppDeviceObjectIDs = NULL; propAddressDevices.mSelector = kAudioHardwarePropertyDevices; propAddressDevices.mScope = kAudioObjectPropertyScopeGlobal; propAddressDevices.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(kAudioObjectSystemObject, &propAddressDevices, 0, NULL, &deviceObjectsDataSize); if (status != noErr) { return ma_result_from_OSStatus(status); } pDeviceObjectIDs = (AudioObjectID*)ma_malloc(deviceObjectsDataSize, &pContext->allocationCallbacks); if (pDeviceObjectIDs == NULL) { return MA_OUT_OF_MEMORY; } status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(kAudioObjectSystemObject, &propAddressDevices, 0, NULL, &deviceObjectsDataSize, pDeviceObjectIDs); if (status != noErr) { ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks); return ma_result_from_OSStatus(status); } *pDeviceCount = deviceObjectsDataSize / sizeof(AudioObjectID); *ppDeviceObjectIDs = pDeviceObjectIDs; return MA_SUCCESS; } static ma_result ma_get_AudioObject_uid_as_CFStringRef(ma_context* pContext, AudioObjectID objectID, CFStringRef* pUID) { AudioObjectPropertyAddress propAddress; UInt32 dataSize; OSStatus status; MA_ASSERT(pContext != NULL); propAddress.mSelector = kAudioDevicePropertyDeviceUID; propAddress.mScope = kAudioObjectPropertyScopeGlobal; propAddress.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; dataSize = sizeof(*pUID); status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(objectID, &propAddress, 0, NULL, &dataSize, pUID); if (status != noErr) { return ma_result_from_OSStatus(status); } return MA_SUCCESS; } static ma_result ma_get_AudioObject_uid(ma_context* pContext, AudioObjectID objectID, size_t bufferSize, char* bufferOut) { CFStringRef uid; ma_result result; MA_ASSERT(pContext != NULL); result = ma_get_AudioObject_uid_as_CFStringRef(pContext, objectID, &uid); if (result != MA_SUCCESS) { return result; } if (!((ma_CFStringGetCString_proc)pContext->coreaudio.CFStringGetCString)(uid, bufferOut, bufferSize, kCFStringEncodingUTF8)) { return MA_ERROR; } ((ma_CFRelease_proc)pContext->coreaudio.CFRelease)(uid); return MA_SUCCESS; } static ma_result ma_get_AudioObject_name(ma_context* pContext, AudioObjectID objectID, size_t bufferSize, char* bufferOut) { AudioObjectPropertyAddress propAddress; CFStringRef deviceName = NULL; UInt32 dataSize; OSStatus status; MA_ASSERT(pContext != NULL); propAddress.mSelector = kAudioDevicePropertyDeviceNameCFString; propAddress.mScope = kAudioObjectPropertyScopeGlobal; propAddress.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; dataSize = sizeof(deviceName); status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(objectID, &propAddress, 0, NULL, &dataSize, &deviceName); if (status != noErr) { return ma_result_from_OSStatus(status); } if (!((ma_CFStringGetCString_proc)pContext->coreaudio.CFStringGetCString)(deviceName, bufferOut, bufferSize, kCFStringEncodingUTF8)) { return MA_ERROR; } ((ma_CFRelease_proc)pContext->coreaudio.CFRelease)(deviceName); return MA_SUCCESS; } static ma_bool32 ma_does_AudioObject_support_scope(ma_context* pContext, AudioObjectID deviceObjectID, AudioObjectPropertyScope scope) { AudioObjectPropertyAddress propAddress; UInt32 dataSize; OSStatus status; AudioBufferList* pBufferList; ma_bool32 isSupported; MA_ASSERT(pContext != NULL); /* To know whether or not a device is an input device we need ot look at the stream configuration. If it has an output channel it's a playback device. */ propAddress.mSelector = kAudioDevicePropertyStreamConfiguration; propAddress.mScope = scope; propAddress.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); if (status != noErr) { return MA_FALSE; } pBufferList = (AudioBufferList*)ma_malloc(dataSize, &pContext->allocationCallbacks); if (pBufferList == NULL) { return MA_FALSE; /* Out of memory. */ } status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pBufferList); if (status != noErr) { ma_free(pBufferList, &pContext->allocationCallbacks); return MA_FALSE; } isSupported = MA_FALSE; if (pBufferList->mNumberBuffers > 0) { isSupported = MA_TRUE; } ma_free(pBufferList, &pContext->allocationCallbacks); return isSupported; } static ma_bool32 ma_does_AudioObject_support_playback(ma_context* pContext, AudioObjectID deviceObjectID) { return ma_does_AudioObject_support_scope(pContext, deviceObjectID, kAudioObjectPropertyScopeOutput); } static ma_bool32 ma_does_AudioObject_support_capture(ma_context* pContext, AudioObjectID deviceObjectID) { return ma_does_AudioObject_support_scope(pContext, deviceObjectID, kAudioObjectPropertyScopeInput); } static ma_result ma_get_AudioObject_stream_descriptions(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, UInt32* pDescriptionCount, AudioStreamRangedDescription** ppDescriptions) /* NOTE: Free the returned pointer with ma_free(). */ { AudioObjectPropertyAddress propAddress; UInt32 dataSize; OSStatus status; AudioStreamRangedDescription* pDescriptions; MA_ASSERT(pContext != NULL); MA_ASSERT(pDescriptionCount != NULL); MA_ASSERT(ppDescriptions != NULL); /* TODO: Experiment with kAudioStreamPropertyAvailablePhysicalFormats instead of (or in addition to) kAudioStreamPropertyAvailableVirtualFormats. My MacBook Pro uses s24/32 format, however, which miniaudio does not currently support. */ propAddress.mSelector = kAudioStreamPropertyAvailableVirtualFormats; /*kAudioStreamPropertyAvailablePhysicalFormats;*/ propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; propAddress.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); if (status != noErr) { return ma_result_from_OSStatus(status); } pDescriptions = (AudioStreamRangedDescription*)ma_malloc(dataSize, &pContext->allocationCallbacks); if (pDescriptions == NULL) { return MA_OUT_OF_MEMORY; } status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pDescriptions); if (status != noErr) { ma_free(pDescriptions, &pContext->allocationCallbacks); return ma_result_from_OSStatus(status); } *pDescriptionCount = dataSize / sizeof(*pDescriptions); *ppDescriptions = pDescriptions; return MA_SUCCESS; } static ma_result ma_get_AudioObject_channel_layout(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, AudioChannelLayout** ppChannelLayout) /* NOTE: Free the returned pointer with ma_free(). */ { AudioObjectPropertyAddress propAddress; UInt32 dataSize; OSStatus status; AudioChannelLayout* pChannelLayout; MA_ASSERT(pContext != NULL); MA_ASSERT(ppChannelLayout != NULL); *ppChannelLayout = NULL; /* Safety. */ propAddress.mSelector = kAudioDevicePropertyPreferredChannelLayout; propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; propAddress.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); if (status != noErr) { return ma_result_from_OSStatus(status); } pChannelLayout = (AudioChannelLayout*)ma_malloc(dataSize, &pContext->allocationCallbacks); if (pChannelLayout == NULL) { return MA_OUT_OF_MEMORY; } status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pChannelLayout); if (status != noErr) { ma_free(pChannelLayout, &pContext->allocationCallbacks); return ma_result_from_OSStatus(status); } *ppChannelLayout = pChannelLayout; return MA_SUCCESS; } static ma_result ma_get_AudioObject_channel_count(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32* pChannelCount) { AudioChannelLayout* pChannelLayout; ma_result result; MA_ASSERT(pContext != NULL); MA_ASSERT(pChannelCount != NULL); *pChannelCount = 0; /* Safety. */ result = ma_get_AudioObject_channel_layout(pContext, deviceObjectID, deviceType, &pChannelLayout); if (result != MA_SUCCESS) { return result; } if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelDescriptions) { *pChannelCount = pChannelLayout->mNumberChannelDescriptions; } else if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelBitmap) { *pChannelCount = ma_count_set_bits(pChannelLayout->mChannelBitmap); } else { *pChannelCount = AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag); } ma_free(pChannelLayout, &pContext->allocationCallbacks); return MA_SUCCESS; } #if 0 static ma_result ma_get_AudioObject_channel_map(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_channel* pChannelMap, size_t channelMapCap) { AudioChannelLayout* pChannelLayout; ma_result result; MA_ASSERT(pContext != NULL); result = ma_get_AudioObject_channel_layout(pContext, deviceObjectID, deviceType, &pChannelLayout); if (result != MA_SUCCESS) { return result; /* Rather than always failing here, would it be more robust to simply assume a default? */ } result = ma_get_channel_map_from_AudioChannelLayout(pChannelLayout, pChannelMap, channelMapCap); if (result != MA_SUCCESS) { ma_free(pChannelLayout, &pContext->allocationCallbacks); return result; } ma_free(pChannelLayout, &pContext->allocationCallbacks); return result; } #endif static ma_result ma_get_AudioObject_sample_rates(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, UInt32* pSampleRateRangesCount, AudioValueRange** ppSampleRateRanges) /* NOTE: Free the returned pointer with ma_free(). */ { AudioObjectPropertyAddress propAddress; UInt32 dataSize; OSStatus status; AudioValueRange* pSampleRateRanges; MA_ASSERT(pContext != NULL); MA_ASSERT(pSampleRateRangesCount != NULL); MA_ASSERT(ppSampleRateRanges != NULL); /* Safety. */ *pSampleRateRangesCount = 0; *ppSampleRateRanges = NULL; propAddress.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; propAddress.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); if (status != noErr) { return ma_result_from_OSStatus(status); } pSampleRateRanges = (AudioValueRange*)ma_malloc(dataSize, &pContext->allocationCallbacks); if (pSampleRateRanges == NULL) { return MA_OUT_OF_MEMORY; } status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pSampleRateRanges); if (status != noErr) { ma_free(pSampleRateRanges, &pContext->allocationCallbacks); return ma_result_from_OSStatus(status); } *pSampleRateRangesCount = dataSize / sizeof(*pSampleRateRanges); *ppSampleRateRanges = pSampleRateRanges; return MA_SUCCESS; } #if 0 static ma_result ma_get_AudioObject_get_closest_sample_rate(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32 sampleRateIn, ma_uint32* pSampleRateOut) { UInt32 sampleRateRangeCount; AudioValueRange* pSampleRateRanges; ma_result result; MA_ASSERT(pContext != NULL); MA_ASSERT(pSampleRateOut != NULL); *pSampleRateOut = 0; /* Safety. */ result = ma_get_AudioObject_sample_rates(pContext, deviceObjectID, deviceType, &sampleRateRangeCount, &pSampleRateRanges); if (result != MA_SUCCESS) { return result; } if (sampleRateRangeCount == 0) { ma_free(pSampleRateRanges, &pContext->allocationCallbacks); return MA_ERROR; /* Should never hit this case should we? */ } if (sampleRateIn == 0) { /* Search in order of miniaudio's preferred priority. */ UInt32 iMALSampleRate; for (iMALSampleRate = 0; iMALSampleRate < ma_countof(g_maStandardSampleRatePriorities); ++iMALSampleRate) { ma_uint32 malSampleRate = g_maStandardSampleRatePriorities[iMALSampleRate]; UInt32 iCASampleRate; for (iCASampleRate = 0; iCASampleRate < sampleRateRangeCount; ++iCASampleRate) { AudioValueRange caSampleRate = pSampleRateRanges[iCASampleRate]; if (caSampleRate.mMinimum <= malSampleRate && caSampleRate.mMaximum >= malSampleRate) { *pSampleRateOut = malSampleRate; ma_free(pSampleRateRanges, &pContext->allocationCallbacks); return MA_SUCCESS; } } } /* If we get here it means none of miniaudio's standard sample rates matched any of the supported sample rates from the device. In this case we just fall back to the first one reported by Core Audio. */ MA_ASSERT(sampleRateRangeCount > 0); *pSampleRateOut = pSampleRateRanges[0].mMinimum; ma_free(pSampleRateRanges, &pContext->allocationCallbacks); return MA_SUCCESS; } else { /* Find the closest match to this sample rate. */ UInt32 currentAbsoluteDifference = INT32_MAX; UInt32 iCurrentClosestRange = (UInt32)-1; UInt32 iRange; for (iRange = 0; iRange < sampleRateRangeCount; ++iRange) { if (pSampleRateRanges[iRange].mMinimum <= sampleRateIn && pSampleRateRanges[iRange].mMaximum >= sampleRateIn) { *pSampleRateOut = sampleRateIn; ma_free(pSampleRateRanges, &pContext->allocationCallbacks); return MA_SUCCESS; } else { UInt32 absoluteDifference; if (pSampleRateRanges[iRange].mMinimum > sampleRateIn) { absoluteDifference = pSampleRateRanges[iRange].mMinimum - sampleRateIn; } else { absoluteDifference = sampleRateIn - pSampleRateRanges[iRange].mMaximum; } if (currentAbsoluteDifference > absoluteDifference) { currentAbsoluteDifference = absoluteDifference; iCurrentClosestRange = iRange; } } } MA_ASSERT(iCurrentClosestRange != (UInt32)-1); *pSampleRateOut = pSampleRateRanges[iCurrentClosestRange].mMinimum; ma_free(pSampleRateRanges, &pContext->allocationCallbacks); return MA_SUCCESS; } /* Should never get here, but it would mean we weren't able to find any suitable sample rates. */ /*ma_free(pSampleRateRanges, &pContext->allocationCallbacks);*/ /*return MA_ERROR;*/ } #endif static ma_result ma_get_AudioObject_closest_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32 bufferSizeInFramesIn, ma_uint32* pBufferSizeInFramesOut) { AudioObjectPropertyAddress propAddress; AudioValueRange bufferSizeRange; UInt32 dataSize; OSStatus status; MA_ASSERT(pContext != NULL); MA_ASSERT(pBufferSizeInFramesOut != NULL); *pBufferSizeInFramesOut = 0; /* Safety. */ propAddress.mSelector = kAudioDevicePropertyBufferFrameSizeRange; propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; propAddress.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; dataSize = sizeof(bufferSizeRange); status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &bufferSizeRange); if (status != noErr) { return ma_result_from_OSStatus(status); } /* This is just a clamp. */ if (bufferSizeInFramesIn < bufferSizeRange.mMinimum) { *pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMinimum; } else if (bufferSizeInFramesIn > bufferSizeRange.mMaximum) { *pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMaximum; } else { *pBufferSizeInFramesOut = bufferSizeInFramesIn; } return MA_SUCCESS; } static ma_result ma_set_AudioObject_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32* pPeriodSizeInOut) { ma_result result; ma_uint32 chosenBufferSizeInFrames; AudioObjectPropertyAddress propAddress; UInt32 dataSize; OSStatus status; MA_ASSERT(pContext != NULL); result = ma_get_AudioObject_closest_buffer_size_in_frames(pContext, deviceObjectID, deviceType, *pPeriodSizeInOut, &chosenBufferSizeInFrames); if (result != MA_SUCCESS) { return result; } /* Try setting the size of the buffer... If this fails we just use whatever is currently set. */ propAddress.mSelector = kAudioDevicePropertyBufferFrameSize; propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; propAddress.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; ((ma_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(chosenBufferSizeInFrames), &chosenBufferSizeInFrames); /* Get the actual size of the buffer. */ dataSize = sizeof(*pPeriodSizeInOut); status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &chosenBufferSizeInFrames); if (status != noErr) { return ma_result_from_OSStatus(status); } *pPeriodSizeInOut = chosenBufferSizeInFrames; return MA_SUCCESS; } static ma_result ma_find_default_AudioObjectID(ma_context* pContext, ma_device_type deviceType, AudioObjectID* pDeviceObjectID) { AudioObjectPropertyAddress propAddressDefaultDevice; UInt32 defaultDeviceObjectIDSize = sizeof(AudioObjectID); AudioObjectID defaultDeviceObjectID; OSStatus status; MA_ASSERT(pContext != NULL); MA_ASSERT(pDeviceObjectID != NULL); /* Safety. */ *pDeviceObjectID = 0; propAddressDefaultDevice.mScope = kAudioObjectPropertyScopeGlobal; propAddressDefaultDevice.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; if (deviceType == ma_device_type_playback) { propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultOutputDevice; } else { propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultInputDevice; } defaultDeviceObjectIDSize = sizeof(AudioObjectID); status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(kAudioObjectSystemObject, &propAddressDefaultDevice, 0, NULL, &defaultDeviceObjectIDSize, &defaultDeviceObjectID); if (status == noErr) { *pDeviceObjectID = defaultDeviceObjectID; return MA_SUCCESS; } /* If we get here it means we couldn't find the device. */ return MA_NO_DEVICE; } static ma_result ma_find_AudioObjectID(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, AudioObjectID* pDeviceObjectID) { MA_ASSERT(pContext != NULL); MA_ASSERT(pDeviceObjectID != NULL); /* Safety. */ *pDeviceObjectID = 0; if (pDeviceID == NULL) { /* Default device. */ return ma_find_default_AudioObjectID(pContext, deviceType, pDeviceObjectID); } else { /* Explicit device. */ UInt32 deviceCount; AudioObjectID* pDeviceObjectIDs; ma_result result; UInt32 iDevice; result = ma_get_device_object_ids__coreaudio(pContext, &deviceCount, &pDeviceObjectIDs); if (result != MA_SUCCESS) { return result; } for (iDevice = 0; iDevice < deviceCount; ++iDevice) { AudioObjectID deviceObjectID = pDeviceObjectIDs[iDevice]; char uid[256]; if (ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(uid), uid) != MA_SUCCESS) { continue; } if (deviceType == ma_device_type_playback) { if (ma_does_AudioObject_support_playback(pContext, deviceObjectID)) { if (strcmp(uid, pDeviceID->coreaudio) == 0) { *pDeviceObjectID = deviceObjectID; ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks); return MA_SUCCESS; } } } else { if (ma_does_AudioObject_support_capture(pContext, deviceObjectID)) { if (strcmp(uid, pDeviceID->coreaudio) == 0) { *pDeviceObjectID = deviceObjectID; ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks); return MA_SUCCESS; } } } } ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks); } /* If we get here it means we couldn't find the device. */ return MA_NO_DEVICE; } static ma_result ma_find_best_format__coreaudio(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const AudioStreamBasicDescription* pOrigFormat, AudioStreamBasicDescription* pFormat) { UInt32 deviceFormatDescriptionCount; AudioStreamRangedDescription* pDeviceFormatDescriptions; ma_result result; ma_uint32 desiredSampleRate; ma_uint32 desiredChannelCount; ma_format desiredFormat; AudioStreamBasicDescription bestDeviceFormatSoFar; ma_bool32 hasSupportedFormat; UInt32 iFormat; result = ma_get_AudioObject_stream_descriptions(pContext, deviceObjectID, deviceType, &deviceFormatDescriptionCount, &pDeviceFormatDescriptions); if (result != MA_SUCCESS) { return result; } desiredSampleRate = sampleRate; if (desiredSampleRate == 0) { desiredSampleRate = (ma_uint32)pOrigFormat->mSampleRate; } desiredChannelCount = channels; if (desiredChannelCount == 0) { desiredChannelCount = pOrigFormat->mChannelsPerFrame; } desiredFormat = format; if (desiredFormat == ma_format_unknown) { result = ma_format_from_AudioStreamBasicDescription(pOrigFormat, &desiredFormat); if (result != MA_SUCCESS || desiredFormat == ma_format_unknown) { desiredFormat = g_maFormatPriorities[0]; } } /* If we get here it means we don't have an exact match to what the client is asking for. We'll need to find the closest one. The next loop will check for formats that have the same sample rate to what we're asking for. If there is, we prefer that one in all cases. */ MA_ZERO_OBJECT(&bestDeviceFormatSoFar); hasSupportedFormat = MA_FALSE; for (iFormat = 0; iFormat < deviceFormatDescriptionCount; ++iFormat) { ma_format formatFromDescription; ma_result formatResult = ma_format_from_AudioStreamBasicDescription(&pDeviceFormatDescriptions[iFormat].mFormat, &formatFromDescription); if (formatResult == MA_SUCCESS && formatFromDescription != ma_format_unknown) { hasSupportedFormat = MA_TRUE; bestDeviceFormatSoFar = pDeviceFormatDescriptions[iFormat].mFormat; break; } } if (!hasSupportedFormat) { ma_free(pDeviceFormatDescriptions, &pContext->allocationCallbacks); return MA_FORMAT_NOT_SUPPORTED; } for (iFormat = 0; iFormat < deviceFormatDescriptionCount; ++iFormat) { AudioStreamBasicDescription thisDeviceFormat = pDeviceFormatDescriptions[iFormat].mFormat; ma_format thisSampleFormat; ma_result formatResult; ma_format bestSampleFormatSoFar; /* If the format is not supported by miniaudio we need to skip this one entirely. */ formatResult = ma_format_from_AudioStreamBasicDescription(&pDeviceFormatDescriptions[iFormat].mFormat, &thisSampleFormat); if (formatResult != MA_SUCCESS || thisSampleFormat == ma_format_unknown) { continue; /* The format is not supported by miniaudio. Skip. */ } ma_format_from_AudioStreamBasicDescription(&bestDeviceFormatSoFar, &bestSampleFormatSoFar); /* Getting here means the format is supported by miniaudio which makes this format a candidate. */ if (thisDeviceFormat.mSampleRate != desiredSampleRate) { /* The sample rate does not match, but this format could still be usable, although it's a very low priority. If the best format so far has an equal sample rate we can just ignore this one. */ if (bestDeviceFormatSoFar.mSampleRate == desiredSampleRate) { continue; /* The best sample rate so far has the same sample rate as what we requested which means it's still the best so far. Skip this format. */ } else { /* In this case, neither the best format so far nor this one have the same sample rate. Check the channel count next. */ if (thisDeviceFormat.mChannelsPerFrame != desiredChannelCount) { /* This format has a different sample rate _and_ a different channel count. */ if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) { continue; /* No change to the best format. */ } else { /* Both this format and the best so far have different sample rates and different channel counts. Whichever has the best format is the new best. */ if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { bestDeviceFormatSoFar = thisDeviceFormat; continue; } else { continue; /* No change to the best format. */ } } } else { /* This format has a different sample rate but the desired channel count. */ if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) { /* Both this format and the best so far have the desired channel count. Whichever has the best format is the new best. */ if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { bestDeviceFormatSoFar = thisDeviceFormat; continue; } else { continue; /* No change to the best format for now. */ } } else { /* This format has the desired channel count, but the best so far does not. We have a new best. */ bestDeviceFormatSoFar = thisDeviceFormat; continue; } } } } else { /* The sample rates match which makes this format a very high priority contender. If the best format so far has a different sample rate it needs to be replaced with this one. */ if (bestDeviceFormatSoFar.mSampleRate != desiredSampleRate) { bestDeviceFormatSoFar = thisDeviceFormat; continue; } else { /* In this case both this format and the best format so far have the same sample rate. Check the channel count next. */ if (thisDeviceFormat.mChannelsPerFrame == desiredChannelCount) { /* In this case this format has the same channel count as what the client is requesting. If the best format so far has a different count, this one becomes the new best. */ if (bestDeviceFormatSoFar.mChannelsPerFrame != desiredChannelCount) { bestDeviceFormatSoFar = thisDeviceFormat; continue; } else { /* In this case both this format and the best so far have the ideal sample rate and channel count. Check the format. */ if (thisSampleFormat == desiredFormat) { bestDeviceFormatSoFar = thisDeviceFormat; break; /* Found the exact match. */ } else { /* The formats are different. The new best format is the one with the highest priority format according to miniaudio. */ if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { bestDeviceFormatSoFar = thisDeviceFormat; continue; } else { continue; /* No change to the best format for now. */ } } } } else { /* In this case the channel count is different to what the client has requested. If the best so far has the same channel count as the requested count then it remains the best. */ if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) { continue; } else { /* This is the case where both have the same sample rate (good) but different channel counts. Right now both have about the same priority, but we need to compare the format now. */ if (thisSampleFormat == bestSampleFormatSoFar) { if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { bestDeviceFormatSoFar = thisDeviceFormat; continue; } else { continue; /* No change to the best format for now. */ } } } } } } } *pFormat = bestDeviceFormatSoFar; ma_free(pDeviceFormatDescriptions, &pContext->allocationCallbacks); return MA_SUCCESS; } static ma_result ma_get_AudioUnit_channel_map(ma_context* pContext, AudioUnit audioUnit, ma_device_type deviceType, ma_channel* pChannelMap, size_t channelMapCap) { AudioUnitScope deviceScope; AudioUnitElement deviceBus; UInt32 channelLayoutSize; OSStatus status; AudioChannelLayout* pChannelLayout; ma_result result; MA_ASSERT(pContext != NULL); if (deviceType == ma_device_type_playback) { deviceScope = kAudioUnitScope_Input; deviceBus = MA_COREAUDIO_OUTPUT_BUS; } else { deviceScope = kAudioUnitScope_Output; deviceBus = MA_COREAUDIO_INPUT_BUS; } status = ((ma_AudioUnitGetPropertyInfo_proc)pContext->coreaudio.AudioUnitGetPropertyInfo)(audioUnit, kAudioUnitProperty_AudioChannelLayout, deviceScope, deviceBus, &channelLayoutSize, NULL); if (status != noErr) { return ma_result_from_OSStatus(status); } pChannelLayout = (AudioChannelLayout*)ma_malloc(channelLayoutSize, &pContext->allocationCallbacks); if (pChannelLayout == NULL) { return MA_OUT_OF_MEMORY; } status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioUnitProperty_AudioChannelLayout, deviceScope, deviceBus, pChannelLayout, &channelLayoutSize); if (status != noErr) { ma_free(pChannelLayout, &pContext->allocationCallbacks); return ma_result_from_OSStatus(status); } result = ma_get_channel_map_from_AudioChannelLayout(pChannelLayout, pChannelMap, channelMapCap); if (result != MA_SUCCESS) { ma_free(pChannelLayout, &pContext->allocationCallbacks); return result; } ma_free(pChannelLayout, &pContext->allocationCallbacks); return MA_SUCCESS; } #endif /* MA_APPLE_DESKTOP */ #if !defined(MA_APPLE_DESKTOP) static void ma_AVAudioSessionPortDescription_to_device_info(AVAudioSessionPortDescription* pPortDesc, ma_device_info* pInfo) { MA_ZERO_OBJECT(pInfo); ma_strncpy_s(pInfo->name, sizeof(pInfo->name), [pPortDesc.portName UTF8String], (size_t)-1); ma_strncpy_s(pInfo->id.coreaudio, sizeof(pInfo->id.coreaudio), [pPortDesc.UID UTF8String], (size_t)-1); } #endif static ma_result ma_context_enumerate_devices__coreaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { #if defined(MA_APPLE_DESKTOP) UInt32 deviceCount; AudioObjectID* pDeviceObjectIDs; AudioObjectID defaultDeviceObjectIDPlayback; AudioObjectID defaultDeviceObjectIDCapture; ma_result result; UInt32 iDevice; ma_find_default_AudioObjectID(pContext, ma_device_type_playback, &defaultDeviceObjectIDPlayback); /* OK if this fails. */ ma_find_default_AudioObjectID(pContext, ma_device_type_capture, &defaultDeviceObjectIDCapture); /* OK if this fails. */ result = ma_get_device_object_ids__coreaudio(pContext, &deviceCount, &pDeviceObjectIDs); if (result != MA_SUCCESS) { return result; } for (iDevice = 0; iDevice < deviceCount; ++iDevice) { AudioObjectID deviceObjectID = pDeviceObjectIDs[iDevice]; ma_device_info info; MA_ZERO_OBJECT(&info); if (ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(info.id.coreaudio), info.id.coreaudio) != MA_SUCCESS) { continue; } if (ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(info.name), info.name) != MA_SUCCESS) { continue; } if (ma_does_AudioObject_support_playback(pContext, deviceObjectID)) { if (deviceObjectID == defaultDeviceObjectIDPlayback) { info.isDefault = MA_TRUE; } if (!callback(pContext, ma_device_type_playback, &info, pUserData)) { break; } } if (ma_does_AudioObject_support_capture(pContext, deviceObjectID)) { if (deviceObjectID == defaultDeviceObjectIDCapture) { info.isDefault = MA_TRUE; } if (!callback(pContext, ma_device_type_capture, &info, pUserData)) { break; } } } ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks); #else ma_device_info info; NSArray *pInputs = [[[AVAudioSession sharedInstance] currentRoute] inputs]; NSArray *pOutputs = [[[AVAudioSession sharedInstance] currentRoute] outputs]; for (AVAudioSessionPortDescription* pPortDesc in pOutputs) { ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, &info); if (!callback(pContext, ma_device_type_playback, &info, pUserData)) { return MA_SUCCESS; } } for (AVAudioSessionPortDescription* pPortDesc in pInputs) { ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, &info); if (!callback(pContext, ma_device_type_capture, &info, pUserData)) { return MA_SUCCESS; } } #endif return MA_SUCCESS; } static ma_result ma_context_get_device_info__coreaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { ma_result result; MA_ASSERT(pContext != NULL); #if defined(MA_APPLE_DESKTOP) /* Desktop */ { AudioObjectID deviceObjectID; AudioObjectID defaultDeviceObjectID; UInt32 streamDescriptionCount; AudioStreamRangedDescription* pStreamDescriptions; UInt32 iStreamDescription; UInt32 sampleRateRangeCount; AudioValueRange* pSampleRateRanges; ma_find_default_AudioObjectID(pContext, deviceType, &defaultDeviceObjectID); /* OK if this fails. */ result = ma_find_AudioObjectID(pContext, deviceType, pDeviceID, &deviceObjectID); if (result != MA_SUCCESS) { return result; } result = ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(pDeviceInfo->id.coreaudio), pDeviceInfo->id.coreaudio); if (result != MA_SUCCESS) { return result; } result = ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(pDeviceInfo->name), pDeviceInfo->name); if (result != MA_SUCCESS) { return result; } if (deviceObjectID == defaultDeviceObjectID) { pDeviceInfo->isDefault = MA_TRUE; } /* There could be a large number of permutations here. Fortunately there is only a single channel count being reported which reduces this quite a bit. For sample rates we're only reporting those that are one of miniaudio's recognized "standard" rates. If there are still more formats than can fit into our fixed sized array we'll just need to truncate them. This is unlikely and will probably only happen if some driver performs software data conversion and therefore reports every possible format and sample rate. */ pDeviceInfo->nativeDataFormatCount = 0; /* Formats. */ { ma_format uniqueFormats[ma_format_count]; ma_uint32 uniqueFormatCount = 0; ma_uint32 channels; /* Channels. */ result = ma_get_AudioObject_channel_count(pContext, deviceObjectID, deviceType, &channels); if (result != MA_SUCCESS) { return result; } /* Formats. */ result = ma_get_AudioObject_stream_descriptions(pContext, deviceObjectID, deviceType, &streamDescriptionCount, &pStreamDescriptions); if (result != MA_SUCCESS) { return result; } for (iStreamDescription = 0; iStreamDescription < streamDescriptionCount; ++iStreamDescription) { ma_format format; ma_bool32 hasFormatBeenHandled = MA_FALSE; ma_uint32 iOutputFormat; ma_uint32 iSampleRate; result = ma_format_from_AudioStreamBasicDescription(&pStreamDescriptions[iStreamDescription].mFormat, &format); if (result != MA_SUCCESS) { continue; } MA_ASSERT(format != ma_format_unknown); /* Make sure the format isn't already in the output list. */ for (iOutputFormat = 0; iOutputFormat < uniqueFormatCount; ++iOutputFormat) { if (uniqueFormats[iOutputFormat] == format) { hasFormatBeenHandled = MA_TRUE; break; } } /* If we've already handled this format just skip it. */ if (hasFormatBeenHandled) { continue; } uniqueFormats[uniqueFormatCount] = format; uniqueFormatCount += 1; /* Sample Rates */ result = ma_get_AudioObject_sample_rates(pContext, deviceObjectID, deviceType, &sampleRateRangeCount, &pSampleRateRanges); if (result != MA_SUCCESS) { return result; } /* Annoyingly Core Audio reports a sample rate range. We just get all the standard rates that are between this range. */ for (iSampleRate = 0; iSampleRate < sampleRateRangeCount; ++iSampleRate) { ma_uint32 iStandardSampleRate; for (iStandardSampleRate = 0; iStandardSampleRate < ma_countof(g_maStandardSampleRatePriorities); iStandardSampleRate += 1) { ma_uint32 standardSampleRate = g_maStandardSampleRatePriorities[iStandardSampleRate]; if (standardSampleRate >= pSampleRateRanges[iSampleRate].mMinimum && standardSampleRate <= pSampleRateRanges[iSampleRate].mMaximum) { /* We have a new data format. Add it to the list. */ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = standardSampleRate; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0; pDeviceInfo->nativeDataFormatCount += 1; if (pDeviceInfo->nativeDataFormatCount >= ma_countof(pDeviceInfo->nativeDataFormats)) { break; /* No more room for any more formats. */ } } } } ma_free(pSampleRateRanges, &pContext->allocationCallbacks); if (pDeviceInfo->nativeDataFormatCount >= ma_countof(pDeviceInfo->nativeDataFormats)) { break; /* No more room for any more formats. */ } } ma_free(pStreamDescriptions, &pContext->allocationCallbacks); } } #else /* Mobile */ { AudioComponentDescription desc; AudioComponent component; AudioUnit audioUnit; OSStatus status; AudioUnitScope formatScope; AudioUnitElement formatElement; AudioStreamBasicDescription bestFormat; UInt32 propSize; /* We want to ensure we use a consistent device name to device enumeration. */ if (pDeviceID != NULL && pDeviceID->coreaudio[0] != '\0') { ma_bool32 found = MA_FALSE; if (deviceType == ma_device_type_playback) { NSArray *pOutputs = [[[AVAudioSession sharedInstance] currentRoute] outputs]; for (AVAudioSessionPortDescription* pPortDesc in pOutputs) { if (strcmp(pDeviceID->coreaudio, [pPortDesc.UID UTF8String]) == 0) { ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, pDeviceInfo); found = MA_TRUE; break; } } } else { NSArray *pInputs = [[[AVAudioSession sharedInstance] currentRoute] inputs]; for (AVAudioSessionPortDescription* pPortDesc in pInputs) { if (strcmp(pDeviceID->coreaudio, [pPortDesc.UID UTF8String]) == 0) { ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, pDeviceInfo); found = MA_TRUE; break; } } } if (!found) { return MA_DOES_NOT_EXIST; } } else { if (deviceType == ma_device_type_playback) { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); } else { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); } } /* Retrieving device information is more annoying on mobile than desktop. For simplicity I'm locking this down to whatever format is reported on a temporary I/O unit. The problem, however, is that this doesn't return a value for the sample rate which we need to retrieve from the AVAudioSession shared instance. */ desc.componentType = kAudioUnitType_Output; desc.componentSubType = kAudioUnitSubType_RemoteIO; desc.componentManufacturer = kAudioUnitManufacturer_Apple; desc.componentFlags = 0; desc.componentFlagsMask = 0; component = ((ma_AudioComponentFindNext_proc)pContext->coreaudio.AudioComponentFindNext)(NULL, &desc); if (component == NULL) { return MA_FAILED_TO_INIT_BACKEND; } status = ((ma_AudioComponentInstanceNew_proc)pContext->coreaudio.AudioComponentInstanceNew)(component, &audioUnit); if (status != noErr) { return ma_result_from_OSStatus(status); } formatScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output; formatElement = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS; propSize = sizeof(bestFormat); status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, &propSize); if (status != noErr) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit); return ma_result_from_OSStatus(status); } ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit); audioUnit = NULL; /* Only a single format is being reported for iOS. */ pDeviceInfo->nativeDataFormatCount = 1; result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pDeviceInfo->nativeDataFormats[0].format); if (result != MA_SUCCESS) { return result; } pDeviceInfo->nativeDataFormats[0].channels = bestFormat.mChannelsPerFrame; /* It looks like Apple are wanting to push the whole AVAudioSession thing. Thus, we need to use that to determine device settings. To do this we just get the shared instance and inspect. */ @autoreleasepool { AVAudioSession* pAudioSession = [AVAudioSession sharedInstance]; MA_ASSERT(pAudioSession != NULL); pDeviceInfo->nativeDataFormats[0].sampleRate = (ma_uint32)pAudioSession.sampleRate; } } #endif (void)pDeviceInfo; /* Unused. */ return MA_SUCCESS; } static AudioBufferList* ma_allocate_AudioBufferList__coreaudio(ma_uint32 sizeInFrames, ma_format format, ma_uint32 channels, ma_stream_layout layout, const ma_allocation_callbacks* pAllocationCallbacks) { AudioBufferList* pBufferList; UInt32 audioBufferSizeInBytes; size_t allocationSize; MA_ASSERT(sizeInFrames > 0); MA_ASSERT(format != ma_format_unknown); MA_ASSERT(channels > 0); allocationSize = sizeof(AudioBufferList) - sizeof(AudioBuffer); /* Subtract sizeof(AudioBuffer) because that part is dynamically sized. */ if (layout == ma_stream_layout_interleaved) { /* Interleaved case. This is the simple case because we just have one buffer. */ allocationSize += sizeof(AudioBuffer) * 1; } else { /* Non-interleaved case. This is the more complex case because there's more than one buffer. */ allocationSize += sizeof(AudioBuffer) * channels; } allocationSize += sizeInFrames * ma_get_bytes_per_frame(format, channels); pBufferList = (AudioBufferList*)ma_malloc(allocationSize, pAllocationCallbacks); if (pBufferList == NULL) { return NULL; } audioBufferSizeInBytes = (UInt32)(sizeInFrames * ma_get_bytes_per_sample(format)); if (layout == ma_stream_layout_interleaved) { pBufferList->mNumberBuffers = 1; pBufferList->mBuffers[0].mNumberChannels = channels; pBufferList->mBuffers[0].mDataByteSize = audioBufferSizeInBytes * channels; pBufferList->mBuffers[0].mData = (ma_uint8*)pBufferList + sizeof(AudioBufferList); } else { ma_uint32 iBuffer; pBufferList->mNumberBuffers = channels; for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) { pBufferList->mBuffers[iBuffer].mNumberChannels = 1; pBufferList->mBuffers[iBuffer].mDataByteSize = audioBufferSizeInBytes; pBufferList->mBuffers[iBuffer].mData = (ma_uint8*)pBufferList + ((sizeof(AudioBufferList) - sizeof(AudioBuffer)) + (sizeof(AudioBuffer) * channels)) + (audioBufferSizeInBytes * iBuffer); } } return pBufferList; } static ma_result ma_device_realloc_AudioBufferList__coreaudio(ma_device* pDevice, ma_uint32 sizeInFrames, ma_format format, ma_uint32 channels, ma_stream_layout layout) { MA_ASSERT(pDevice != NULL); MA_ASSERT(format != ma_format_unknown); MA_ASSERT(channels > 0); /* Only resize the buffer if necessary. */ if (pDevice->coreaudio.audioBufferCapInFrames < sizeInFrames) { AudioBufferList* pNewAudioBufferList; pNewAudioBufferList = ma_allocate_AudioBufferList__coreaudio(sizeInFrames, format, channels, layout, &pDevice->pContext->allocationCallbacks); if (pNewAudioBufferList == NULL) { return MA_OUT_OF_MEMORY; } /* At this point we'll have a new AudioBufferList and we can free the old one. */ ma_free(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks); pDevice->coreaudio.pAudioBufferList = pNewAudioBufferList; pDevice->coreaudio.audioBufferCapInFrames = sizeInFrames; } /* Getting here means the capacity of the audio is fine. */ return MA_SUCCESS; } static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pBufferList) { ma_device* pDevice = (ma_device*)pUserData; ma_stream_layout layout; MA_ASSERT(pDevice != NULL); /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Output Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", (int)busNumber, (int)frameCount, (int)pBufferList->mNumberBuffers);*/ /* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */ layout = ma_stream_layout_interleaved; if (pBufferList->mBuffers[0].mNumberChannels != pDevice->playback.internalChannels) { layout = ma_stream_layout_deinterleaved; } if (layout == ma_stream_layout_interleaved) { /* For now we can assume everything is interleaved. */ UInt32 iBuffer; for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) { if (pBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->playback.internalChannels) { ma_uint32 frameCountForThisBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); if (frameCountForThisBuffer > 0) { ma_device_handle_backend_data_callback(pDevice, pBufferList->mBuffers[iBuffer].mData, NULL, frameCountForThisBuffer); } /*a_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pBufferList->mBuffers[iBuffer].mNumberChannels, (int)pBufferList->mBuffers[iBuffer].mDataByteSize);*/ } else { /* This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams. We just output silence here. */ MA_ZERO_MEMORY(pBufferList->mBuffers[iBuffer].mData, pBufferList->mBuffers[iBuffer].mDataByteSize); /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pBufferList->mBuffers[iBuffer].mNumberChannels, (int)pBufferList->mBuffers[iBuffer].mDataByteSize);*/ } } } else { /* This is the deinterleaved case. We need to update each buffer in groups of internalChannels. This assumes each buffer is the same size. */ MA_ASSERT(pDevice->playback.internalChannels <= MA_MAX_CHANNELS); /* This should have been validated at initialization time. */ /* For safety we'll check that the internal channels is a multiple of the buffer count. If it's not it means something very strange has happened and we're not going to support it. */ if ((pBufferList->mNumberBuffers % pDevice->playback.internalChannels) == 0) { ma_uint8 tempBuffer[4096]; UInt32 iBuffer; for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; iBuffer += pDevice->playback.internalChannels) { ma_uint32 frameCountPerBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_sample(pDevice->playback.internalFormat); ma_uint32 framesRemaining = frameCountPerBuffer; while (framesRemaining > 0) { void* ppDeinterleavedBuffers[MA_MAX_CHANNELS]; ma_uint32 iChannel; ma_uint32 framesToRead = sizeof(tempBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); if (framesToRead > framesRemaining) { framesToRead = framesRemaining; } ma_device_handle_backend_data_callback(pDevice, tempBuffer, NULL, framesToRead); for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) { ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pBufferList->mBuffers[iBuffer+iChannel].mData, (frameCountPerBuffer - framesRemaining) * ma_get_bytes_per_sample(pDevice->playback.internalFormat)); } ma_deinterleave_pcm_frames(pDevice->playback.internalFormat, pDevice->playback.internalChannels, framesToRead, tempBuffer, ppDeinterleavedBuffers); framesRemaining -= framesToRead; } } } } (void)pActionFlags; (void)pTimeStamp; (void)busNumber; (void)frameCount; return noErr; } static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pUnusedBufferList) { ma_device* pDevice = (ma_device*)pUserData; AudioBufferList* pRenderedBufferList; ma_result result; ma_stream_layout layout; ma_uint32 iBuffer; OSStatus status; MA_ASSERT(pDevice != NULL); pRenderedBufferList = (AudioBufferList*)pDevice->coreaudio.pAudioBufferList; MA_ASSERT(pRenderedBufferList); /* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */ layout = ma_stream_layout_interleaved; if (pRenderedBufferList->mBuffers[0].mNumberChannels != pDevice->capture.internalChannels) { layout = ma_stream_layout_deinterleaved; } /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Input Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", (int)busNumber, (int)frameCount, (int)pRenderedBufferList->mNumberBuffers);*/ /* There has been a situation reported where frame count passed into this function is greater than the capacity of our capture buffer. There doesn't seem to be a reliable way to determine what the maximum frame count will be, so we need to instead resort to dynamically reallocating our buffer to ensure it's large enough to capture the number of frames requested by this callback. */ result = ma_device_realloc_AudioBufferList__coreaudio(pDevice, frameCount, pDevice->capture.internalFormat, pDevice->capture.internalChannels, layout); if (result != MA_SUCCESS) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Failed to allocate AudioBufferList for capture.\n"); return noErr; } pRenderedBufferList = (AudioBufferList*)pDevice->coreaudio.pAudioBufferList; MA_ASSERT(pRenderedBufferList); /* When you call AudioUnitRender(), Core Audio tries to be helpful by setting the mDataByteSize to the number of bytes that were actually rendered. The problem with this is that the next call can fail with -50 due to the size no longer being set to the capacity of the buffer, but instead the size in bytes of the previous render. This will cause a problem when a future call to this callback specifies a larger number of frames. To work around this we need to explicitly set the size of each buffer to their respective size in bytes. */ for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) { pRenderedBufferList->mBuffers[iBuffer].mDataByteSize = pDevice->coreaudio.audioBufferCapInFrames * ma_get_bytes_per_sample(pDevice->capture.internalFormat) * pRenderedBufferList->mBuffers[iBuffer].mNumberChannels; /*printf("DEBUG: nDataByteSize = %d\n", (int)pRenderedBufferList->mBuffers[iBuffer].mDataByteSize);*/ } status = ((ma_AudioUnitRender_proc)pDevice->pContext->coreaudio.AudioUnitRender)((AudioUnit)pDevice->coreaudio.audioUnitCapture, pActionFlags, pTimeStamp, busNumber, frameCount, pRenderedBufferList); if (status != noErr) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "ERROR: AudioUnitRender() failed with %d.\n", (int)status); return status; } if (layout == ma_stream_layout_interleaved) { for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) { if (pRenderedBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->capture.internalChannels) { ma_device_handle_backend_data_callback(pDevice, NULL, pRenderedBufferList->mBuffers[iBuffer].mData, frameCount); /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " mDataByteSize=%d.\n", (int)pRenderedBufferList->mBuffers[iBuffer].mDataByteSize);*/ } else { /* This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams. */ ma_uint8 silentBuffer[4096]; ma_uint32 framesRemaining; MA_ZERO_MEMORY(silentBuffer, sizeof(silentBuffer)); framesRemaining = frameCount; while (framesRemaining > 0) { ma_uint32 framesToSend = sizeof(silentBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); if (framesToSend > framesRemaining) { framesToSend = framesRemaining; } ma_device_handle_backend_data_callback(pDevice, NULL, silentBuffer, framesToSend); framesRemaining -= framesToSend; } /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pRenderedBufferList->mBuffers[iBuffer].mNumberChannels, (int)pRenderedBufferList->mBuffers[iBuffer].mDataByteSize);*/ } } } else { /* This is the deinterleaved case. We need to interleave the audio data before sending it to the client. This assumes each buffer is the same size. */ MA_ASSERT(pDevice->capture.internalChannels <= MA_MAX_CHANNELS); /* This should have been validated at initialization time. */ /* For safety we'll check that the internal channels is a multiple of the buffer count. If it's not it means something very strange has happened and we're not going to support it. */ if ((pRenderedBufferList->mNumberBuffers % pDevice->capture.internalChannels) == 0) { ma_uint8 tempBuffer[4096]; for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; iBuffer += pDevice->capture.internalChannels) { ma_uint32 framesRemaining = frameCount; while (framesRemaining > 0) { void* ppDeinterleavedBuffers[MA_MAX_CHANNELS]; ma_uint32 iChannel; ma_uint32 framesToSend = sizeof(tempBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); if (framesToSend > framesRemaining) { framesToSend = framesRemaining; } for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) { ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pRenderedBufferList->mBuffers[iBuffer+iChannel].mData, (frameCount - framesRemaining) * ma_get_bytes_per_sample(pDevice->capture.internalFormat)); } ma_interleave_pcm_frames(pDevice->capture.internalFormat, pDevice->capture.internalChannels, framesToSend, (const void**)ppDeinterleavedBuffers, tempBuffer); ma_device_handle_backend_data_callback(pDevice, NULL, tempBuffer, framesToSend); framesRemaining -= framesToSend; } } } } (void)pActionFlags; (void)pTimeStamp; (void)busNumber; (void)frameCount; (void)pUnusedBufferList; return noErr; } static void on_start_stop__coreaudio(void* pUserData, AudioUnit audioUnit, AudioUnitPropertyID propertyID, AudioUnitScope scope, AudioUnitElement element) { ma_device* pDevice = (ma_device*)pUserData; MA_ASSERT(pDevice != NULL); /* Don't do anything if it looks like we're just reinitializing due to a device switch. */ if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isSwitchingPlaybackDevice) || ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isSwitchingCaptureDevice)) { return; } /* There's been a report of a deadlock here when triggered by ma_device_uninit(). It looks like AudioUnitGetProprty (called below) and AudioComponentInstanceDispose (called in ma_device_uninit) can try waiting on the same lock. I'm going to try working around this by not calling any Core Audio APIs in the callback when the device has been stopped or uninitialized. */ if (ma_device_get_state(pDevice) == ma_device_state_uninitialized || ma_device_get_state(pDevice) == ma_device_state_stopping || ma_device_get_state(pDevice) == ma_device_state_stopped) { ma_device__on_notification_stopped(pDevice); } else { UInt32 isRunning; UInt32 isRunningSize = sizeof(isRunning); OSStatus status = ((ma_AudioUnitGetProperty_proc)pDevice->pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioOutputUnitProperty_IsRunning, scope, element, &isRunning, &isRunningSize); if (status != noErr) { goto done; /* Don't really know what to do in this case... just ignore it, I suppose... */ } if (!isRunning) { /* The stop event is a bit annoying in Core Audio because it will be called when we automatically switch the default device. Some scenarios to consider: 1) When the device is unplugged, this will be called _before_ the default device change notification. 2) When the device is changed via the default device change notification, this will be called _after_ the switch. For case #1, we just check if there's a new default device available. If so, we just ignore the stop event. For case #2 we check a flag. */ if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isDefaultPlaybackDevice) || ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isDefaultCaptureDevice)) { /* It looks like the device is switching through an external event, such as the user unplugging the device or changing the default device via the operating system's sound settings. If we're re-initializing the device, we just terminate because we want the stopping of the device to be seamless to the client (we don't want them receiving the stopped event and thinking that the device has stopped when it hasn't!). */ if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isSwitchingPlaybackDevice) || ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isSwitchingCaptureDevice)) { goto done; } /* Getting here means the device is not reinitializing which means it may have been unplugged. From what I can see, it looks like Core Audio will try switching to the new default device seamlessly. We need to somehow find a way to determine whether or not Core Audio will most likely be successful in switching to the new device. TODO: Try to predict if Core Audio will switch devices. If not, the stopped callback needs to be posted. */ goto done; } /* Getting here means we need to stop the device. */ ma_device__on_notification_stopped(pDevice); } } (void)propertyID; /* Unused. */ done: /* Always signal the stop event. It's possible for the "else" case to get hit which can happen during an interruption. */ ma_event_signal(&pDevice->coreaudio.stopEvent); } #if defined(MA_APPLE_DESKTOP) static ma_spinlock g_DeviceTrackingInitLock_CoreAudio = 0; /* A spinlock for mutal exclusion of the init/uninit of the global tracking data. Initialization to 0 is what we need. */ static ma_uint32 g_DeviceTrackingInitCounter_CoreAudio = 0; static ma_mutex g_DeviceTrackingMutex_CoreAudio; static ma_device** g_ppTrackedDevices_CoreAudio = NULL; static ma_uint32 g_TrackedDeviceCap_CoreAudio = 0; static ma_uint32 g_TrackedDeviceCount_CoreAudio = 0; static OSStatus ma_default_device_changed__coreaudio(AudioObjectID objectID, UInt32 addressCount, const AudioObjectPropertyAddress* pAddresses, void* pUserData) { ma_device_type deviceType; /* Not sure if I really need to check this, but it makes me feel better. */ if (addressCount == 0) { return noErr; } if (pAddresses[0].mSelector == kAudioHardwarePropertyDefaultOutputDevice) { deviceType = ma_device_type_playback; } else if (pAddresses[0].mSelector == kAudioHardwarePropertyDefaultInputDevice) { deviceType = ma_device_type_capture; } else { return noErr; /* Should never hit this. */ } ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio); { ma_uint32 iDevice; for (iDevice = 0; iDevice < g_TrackedDeviceCount_CoreAudio; iDevice += 1) { ma_result reinitResult; ma_device* pDevice; pDevice = g_ppTrackedDevices_CoreAudio[iDevice]; if (pDevice->type == deviceType || pDevice->type == ma_device_type_duplex) { if (deviceType == ma_device_type_playback) { pDevice->coreaudio.isSwitchingPlaybackDevice = MA_TRUE; reinitResult = ma_device_reinit_internal__coreaudio(pDevice, deviceType, MA_TRUE); pDevice->coreaudio.isSwitchingPlaybackDevice = MA_FALSE; } else { pDevice->coreaudio.isSwitchingCaptureDevice = MA_TRUE; reinitResult = ma_device_reinit_internal__coreaudio(pDevice, deviceType, MA_TRUE); pDevice->coreaudio.isSwitchingCaptureDevice = MA_FALSE; } if (reinitResult == MA_SUCCESS) { ma_device__post_init_setup(pDevice, deviceType); /* Restart the device if required. If this fails we need to stop the device entirely. */ if (ma_device_get_state(pDevice) == ma_device_state_started) { OSStatus status; if (deviceType == ma_device_type_playback) { status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); if (status != noErr) { if (pDevice->type == ma_device_type_duplex) { ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); } ma_device__set_state(pDevice, ma_device_state_stopped); } } else if (deviceType == ma_device_type_capture) { status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitCapture); if (status != noErr) { if (pDevice->type == ma_device_type_duplex) { ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); } ma_device__set_state(pDevice, ma_device_state_stopped); } } } ma_device__on_notification_rerouted(pDevice); } } } } ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); /* Unused parameters. */ (void)objectID; (void)pUserData; return noErr; } static ma_result ma_context__init_device_tracking__coreaudio(ma_context* pContext) { MA_ASSERT(pContext != NULL); ma_spinlock_lock(&g_DeviceTrackingInitLock_CoreAudio); { /* Don't do anything if we've already initialized device tracking. */ if (g_DeviceTrackingInitCounter_CoreAudio == 0) { AudioObjectPropertyAddress propAddress; propAddress.mScope = kAudioObjectPropertyScopeGlobal; propAddress.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; ma_mutex_init(&g_DeviceTrackingMutex_CoreAudio); propAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice; ((ma_AudioObjectAddPropertyListener_proc)pContext->coreaudio.AudioObjectAddPropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); propAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; ((ma_AudioObjectAddPropertyListener_proc)pContext->coreaudio.AudioObjectAddPropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); } g_DeviceTrackingInitCounter_CoreAudio += 1; } ma_spinlock_unlock(&g_DeviceTrackingInitLock_CoreAudio); return MA_SUCCESS; } static ma_result ma_context__uninit_device_tracking__coreaudio(ma_context* pContext) { MA_ASSERT(pContext != NULL); ma_spinlock_lock(&g_DeviceTrackingInitLock_CoreAudio); { if (g_DeviceTrackingInitCounter_CoreAudio > 0) g_DeviceTrackingInitCounter_CoreAudio -= 1; if (g_DeviceTrackingInitCounter_CoreAudio == 0) { AudioObjectPropertyAddress propAddress; propAddress.mScope = kAudioObjectPropertyScopeGlobal; propAddress.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; propAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice; ((ma_AudioObjectRemovePropertyListener_proc)pContext->coreaudio.AudioObjectRemovePropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); propAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; ((ma_AudioObjectRemovePropertyListener_proc)pContext->coreaudio.AudioObjectRemovePropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); /* At this point there should be no tracked devices. If not there's an error somewhere. */ if (g_ppTrackedDevices_CoreAudio != NULL) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "You have uninitialized all contexts while an associated device is still active."); ma_spinlock_unlock(&g_DeviceTrackingInitLock_CoreAudio); return MA_INVALID_OPERATION; } ma_mutex_uninit(&g_DeviceTrackingMutex_CoreAudio); } } ma_spinlock_unlock(&g_DeviceTrackingInitLock_CoreAudio); return MA_SUCCESS; } static ma_result ma_device__track__coreaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio); { /* Allocate memory if required. */ if (g_TrackedDeviceCap_CoreAudio <= g_TrackedDeviceCount_CoreAudio) { ma_uint32 newCap; ma_device** ppNewDevices; newCap = g_TrackedDeviceCap_CoreAudio * 2; if (newCap == 0) { newCap = 1; } ppNewDevices = (ma_device**)ma_realloc(g_ppTrackedDevices_CoreAudio, sizeof(*g_ppTrackedDevices_CoreAudio)*newCap, &pDevice->pContext->allocationCallbacks); if (ppNewDevices == NULL) { ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); return MA_OUT_OF_MEMORY; } g_ppTrackedDevices_CoreAudio = ppNewDevices; g_TrackedDeviceCap_CoreAudio = newCap; } g_ppTrackedDevices_CoreAudio[g_TrackedDeviceCount_CoreAudio] = pDevice; g_TrackedDeviceCount_CoreAudio += 1; } ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); return MA_SUCCESS; } static ma_result ma_device__untrack__coreaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio); { ma_uint32 iDevice; for (iDevice = 0; iDevice < g_TrackedDeviceCount_CoreAudio; iDevice += 1) { if (g_ppTrackedDevices_CoreAudio[iDevice] == pDevice) { /* We've found the device. We now need to remove it from the list. */ ma_uint32 jDevice; for (jDevice = iDevice; jDevice < g_TrackedDeviceCount_CoreAudio-1; jDevice += 1) { g_ppTrackedDevices_CoreAudio[jDevice] = g_ppTrackedDevices_CoreAudio[jDevice+1]; } g_TrackedDeviceCount_CoreAudio -= 1; /* If there's nothing else in the list we need to free memory. */ if (g_TrackedDeviceCount_CoreAudio == 0) { ma_free(g_ppTrackedDevices_CoreAudio, &pDevice->pContext->allocationCallbacks); g_ppTrackedDevices_CoreAudio = NULL; g_TrackedDeviceCap_CoreAudio = 0; } break; } } } ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); return MA_SUCCESS; } #endif #if defined(MA_APPLE_MOBILE) @interface ma_ios_notification_handler:NSObject { ma_device* m_pDevice; } @end @implementation ma_ios_notification_handler -(id)init:(ma_device*)pDevice { self = [super init]; m_pDevice = pDevice; /* For route changes. */ [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(handle_route_change:) name:AVAudioSessionRouteChangeNotification object:[AVAudioSession sharedInstance]]; /* For interruptions. */ [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(handle_interruption:) name:AVAudioSessionInterruptionNotification object:[AVAudioSession sharedInstance]]; return self; } -(void)dealloc { [self remove_handler]; #if defined(__has_feature) #if !__has_feature(objc_arc) [super dealloc]; #endif #endif } -(void)remove_handler { [[NSNotificationCenter defaultCenter] removeObserver:self name:AVAudioSessionRouteChangeNotification object:nil]; [[NSNotificationCenter defaultCenter] removeObserver:self name:AVAudioSessionInterruptionNotification object:nil]; } -(void)handle_interruption:(NSNotification*)pNotification { NSInteger type = [[[pNotification userInfo] objectForKey:AVAudioSessionInterruptionTypeKey] integerValue]; switch (type) { case AVAudioSessionInterruptionTypeBegan: { ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Interruption: AVAudioSessionInterruptionTypeBegan\n"); /* Core Audio will have stopped the internal device automatically, but we need explicitly stop it at a higher level to ensure miniaudio-specific state is updated for consistency. */ ma_device_stop(m_pDevice); /* Fire the notification after the device has been stopped to ensure it's in the correct state when the notification handler is invoked. */ ma_device__on_notification_interruption_began(m_pDevice); } break; case AVAudioSessionInterruptionTypeEnded: { ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Interruption: AVAudioSessionInterruptionTypeEnded\n"); ma_device__on_notification_interruption_ended(m_pDevice); } break; } } -(void)handle_route_change:(NSNotification*)pNotification { AVAudioSession* pSession = [AVAudioSession sharedInstance]; NSInteger reason = [[[pNotification userInfo] objectForKey:AVAudioSessionRouteChangeReasonKey] integerValue]; switch (reason) { case AVAudioSessionRouteChangeReasonOldDeviceUnavailable: { ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonOldDeviceUnavailable\n"); } break; case AVAudioSessionRouteChangeReasonNewDeviceAvailable: { ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonNewDeviceAvailable\n"); } break; case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory: { ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory\n"); } break; case AVAudioSessionRouteChangeReasonWakeFromSleep: { ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonWakeFromSleep\n"); } break; case AVAudioSessionRouteChangeReasonOverride: { ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonOverride\n"); } break; case AVAudioSessionRouteChangeReasonCategoryChange: { ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonCategoryChange\n"); } break; case AVAudioSessionRouteChangeReasonUnknown: default: { ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonUnknown\n"); } break; } ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_DEBUG, "[Core Audio] Changing Route. inputNumberChannels=%d; outputNumberOfChannels=%d\n", (int)pSession.inputNumberOfChannels, (int)pSession.outputNumberOfChannels); /* Let the application know about the route change. */ ma_device__on_notification_rerouted(m_pDevice); } @end #endif static ma_result ma_device_uninit__coreaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); MA_ASSERT(ma_device_get_state(pDevice) == ma_device_state_uninitialized); #if defined(MA_APPLE_DESKTOP) /* Make sure we're no longer tracking the device. It doesn't matter if we call this for a non-default device because it'll just gracefully ignore it. */ ma_device__untrack__coreaudio(pDevice); #endif #if defined(MA_APPLE_MOBILE) if (pDevice->coreaudio.pNotificationHandler != NULL) { ma_ios_notification_handler* pNotificationHandler = (MA_BRIDGE_TRANSFER ma_ios_notification_handler*)pDevice->coreaudio.pNotificationHandler; [pNotificationHandler remove_handler]; } #endif if (pDevice->coreaudio.audioUnitCapture != NULL) { ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture); } if (pDevice->coreaudio.audioUnitPlayback != NULL) { ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); } if (pDevice->coreaudio.pAudioBufferList) { ma_free(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks); } return MA_SUCCESS; } typedef struct { ma_bool32 allowNominalSampleRateChange; /* Input. */ ma_format formatIn; ma_uint32 channelsIn; ma_uint32 sampleRateIn; ma_channel channelMapIn[MA_MAX_CHANNELS]; ma_uint32 periodSizeInFramesIn; ma_uint32 periodSizeInMillisecondsIn; ma_uint32 periodsIn; ma_share_mode shareMode; ma_performance_profile performanceProfile; ma_bool32 registerStopEvent; /* Output. */ #if defined(MA_APPLE_DESKTOP) AudioObjectID deviceObjectID; #endif AudioComponent component; AudioUnit audioUnit; AudioBufferList* pAudioBufferList; /* Only used for input devices. */ ma_format formatOut; ma_uint32 channelsOut; ma_uint32 sampleRateOut; ma_channel channelMapOut[MA_MAX_CHANNELS]; ma_uint32 periodSizeInFramesOut; ma_uint32 periodsOut; char deviceName[256]; } ma_device_init_internal_data__coreaudio; static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_init_internal_data__coreaudio* pData, void* pDevice_DoNotReference) /* <-- pDevice is typed as void* intentionally so as to avoid accidentally referencing it. */ { ma_result result = MA_SUCCESS; OSStatus status; UInt32 enableIOFlag; AudioStreamBasicDescription bestFormat; ma_uint32 actualPeriodSizeInFrames; AURenderCallbackStruct callbackInfo; #if defined(MA_APPLE_DESKTOP) AudioObjectID deviceObjectID; #endif /* This API should only be used for a single device type: playback or capture. No full-duplex mode. */ if (deviceType == ma_device_type_duplex) { return MA_INVALID_ARGS; } MA_ASSERT(pContext != NULL); MA_ASSERT(deviceType == ma_device_type_playback || deviceType == ma_device_type_capture); #if defined(MA_APPLE_DESKTOP) pData->deviceObjectID = 0; #endif pData->component = NULL; pData->audioUnit = NULL; pData->pAudioBufferList = NULL; #if defined(MA_APPLE_DESKTOP) result = ma_find_AudioObjectID(pContext, deviceType, pDeviceID, &deviceObjectID); if (result != MA_SUCCESS) { return result; } pData->deviceObjectID = deviceObjectID; #endif /* Core audio doesn't really use the notion of a period so we can leave this unmodified, but not too over the top. */ pData->periodsOut = pData->periodsIn; if (pData->periodsOut == 0) { pData->periodsOut = MA_DEFAULT_PERIODS; } if (pData->periodsOut > 16) { pData->periodsOut = 16; } /* Audio unit. */ status = ((ma_AudioComponentInstanceNew_proc)pContext->coreaudio.AudioComponentInstanceNew)((AudioComponent)pContext->coreaudio.component, (AudioUnit*)&pData->audioUnit); if (status != noErr) { return ma_result_from_OSStatus(status); } /* The input/output buses need to be explicitly enabled and disabled. We set the flag based on the output unit first, then we just swap it for input. */ enableIOFlag = 1; if (deviceType == ma_device_type_capture) { enableIOFlag = 0; } status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, MA_COREAUDIO_OUTPUT_BUS, &enableIOFlag, sizeof(enableIOFlag)); if (status != noErr) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return ma_result_from_OSStatus(status); } enableIOFlag = (enableIOFlag == 0) ? 1 : 0; status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, MA_COREAUDIO_INPUT_BUS, &enableIOFlag, sizeof(enableIOFlag)); if (status != noErr) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return ma_result_from_OSStatus(status); } /* Set the device to use with this audio unit. This is only used on desktop since we are using defaults on mobile. */ #if defined(MA_APPLE_DESKTOP) status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &deviceObjectID, sizeof(deviceObjectID)); if (status != noErr) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return ma_result_from_OSStatus(result); } #else /* For some reason it looks like Apple is only allowing selection of the input device. There does not appear to be any way to change the default output route. I have no idea why this is like this, but for now we'll only be able to configure capture devices. */ if (pDeviceID != NULL) { if (deviceType == ma_device_type_capture) { ma_bool32 found = MA_FALSE; NSArray *pInputs = [[[AVAudioSession sharedInstance] currentRoute] inputs]; for (AVAudioSessionPortDescription* pPortDesc in pInputs) { if (strcmp(pDeviceID->coreaudio, [pPortDesc.UID UTF8String]) == 0) { [[AVAudioSession sharedInstance] setPreferredInput:pPortDesc error:nil]; found = MA_TRUE; break; } } if (found == MA_FALSE) { return MA_DOES_NOT_EXIST; } } } #endif /* Format. This is the hardest part of initialization because there's a few variables to take into account. 1) The format must be supported by the device. 2) The format must be supported miniaudio. 3) There's a priority that miniaudio prefers. Ideally we would like to use a format that's as close to the hardware as possible so we can get as close to a passthrough as possible. The most important property is the sample rate. miniaudio can do format conversion for any sample rate and channel count, but cannot do the same for the sample data format. If the sample data format is not supported by miniaudio it must be ignored completely. On mobile platforms this is a bit different. We just force the use of whatever the audio unit's current format is set to. */ { AudioStreamBasicDescription origFormat; UInt32 origFormatSize = sizeof(origFormat); AudioUnitScope formatScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output; AudioUnitElement formatElement = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS; if (deviceType == ma_device_type_playback) { status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, MA_COREAUDIO_OUTPUT_BUS, &origFormat, &origFormatSize); } else { status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, MA_COREAUDIO_INPUT_BUS, &origFormat, &origFormatSize); } if (status != noErr) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return ma_result_from_OSStatus(status); } #if defined(MA_APPLE_DESKTOP) result = ma_find_best_format__coreaudio(pContext, deviceObjectID, deviceType, pData->formatIn, pData->channelsIn, pData->sampleRateIn, &origFormat, &bestFormat); if (result != MA_SUCCESS) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return result; } /* Technical Note TN2091: Device input using the HAL Output Audio Unit https://developer.apple.com/library/archive/technotes/tn2091/_index.html This documentation says the following: The internal AudioConverter can handle any *simple* conversion. Typically, this means that a client can specify ANY variant of the PCM formats. Consequently, the device's sample rate should match the desired sample rate. If sample rate conversion is needed, it can be accomplished by buffering the input and converting the data on a separate thread with another AudioConverter. The important part here is the mention that it can handle *simple* conversions, which does *not* include sample rate. We therefore want to ensure the sample rate stays consistent. This document is specifically for input, but I'm going to play it safe and apply the same rule to output as well. I have tried going against the documentation by setting the sample rate anyway, but this just results in AudioUnitRender() returning a result code of -10863. I have also tried changing the format directly on the input scope on the input bus, but this just results in `ca_require: IsStreamFormatWritable(inScope, inElement) NotWritable` when trying to set the format. Something that does seem to work, however, has been setting the nominal sample rate on the device object. The problem with this, however, is that it actually changes the sample rate at the operating system level and not just the application. This could be intrusive to the user, however, so I don't think it's wise to make this the default. Instead I'm making this a configuration option. When the `coreaudio.allowNominalSampleRateChange` config option is set to true, changing the sample rate will be allowed. Otherwise it'll be fixed to the current sample rate. To check the system-defined sample rate, run the Audio MIDI Setup program that comes installed on macOS and observe how the sample rate changes as the sample rate is changed by miniaudio. */ if (pData->allowNominalSampleRateChange) { AudioValueRange sampleRateRange; AudioObjectPropertyAddress propAddress; sampleRateRange.mMinimum = bestFormat.mSampleRate; sampleRateRange.mMaximum = bestFormat.mSampleRate; propAddress.mSelector = kAudioDevicePropertyNominalSampleRate; propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; propAddress.mElement = AUDIO_OBJECT_PROPERTY_ELEMENT; status = ((ma_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(sampleRateRange), &sampleRateRange); if (status != noErr) { bestFormat.mSampleRate = origFormat.mSampleRate; } } else { bestFormat.mSampleRate = origFormat.mSampleRate; } status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat)); if (status != noErr) { /* We failed to set the format, so fall back to the current format of the audio unit. */ bestFormat = origFormat; } #else bestFormat = origFormat; /* Sample rate is a little different here because for some reason kAudioUnitProperty_StreamFormat returns 0... Oh well. We need to instead try setting the sample rate to what the user has requested and then just see the results of it. Need to use some Objective-C here for this since it depends on Apple's AVAudioSession API. To do this we just get the shared AVAudioSession instance and then set it. Note that from what I can tell, it looks like the sample rate is shared between playback and capture for everything. */ @autoreleasepool { AVAudioSession* pAudioSession = [AVAudioSession sharedInstance]; MA_ASSERT(pAudioSession != NULL); [pAudioSession setPreferredSampleRate:(double)pData->sampleRateIn error:nil]; bestFormat.mSampleRate = pAudioSession.sampleRate; /* I've had a report that the channel count returned by AudioUnitGetProperty above is inconsistent with AVAudioSession outputNumberOfChannels. I'm going to try using the AVAudioSession values instead. UPDATE 20/02/2025: When testing on the simulator with an iPhone 15 and iOS 17 I get an error when initializing the audio unit if set the input channels to pAudioSession.inputNumberOfChannels. What is happening is the channel count returned from AudioUnitGetProperty() above is set to 2, but pAudioSession is reporting a channel count of 1. When this happens, the call to AudioUnitSetProprty() below just down below will succeed, but AudioUnitInitialize() further down will fail. The only solution I have come up with is to not set the channel count to pAudioSession.inputNumberOfChannels. */ if (deviceType == ma_device_type_playback) { bestFormat.mChannelsPerFrame = (UInt32)pAudioSession.outputNumberOfChannels; } #if 0 if (deviceType == ma_device_type_capture) { /*printf("DEBUG: bestFormat.mChannelsPerFrame = %d; pAudioSession.inputNumberOfChannels = %d\n", (int)bestFormat.mChannelsPerFrame, (int)pAudioSession.inputNumberOfChannels);*/ bestFormat.mChannelsPerFrame = (UInt32)pAudioSession.inputNumberOfChannels; } #endif } status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat)); if (status != noErr) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return ma_result_from_OSStatus(status); } #endif result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pData->formatOut); if (result != MA_SUCCESS) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return result; } if (pData->formatOut == ma_format_unknown) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return MA_FORMAT_NOT_SUPPORTED; } pData->channelsOut = bestFormat.mChannelsPerFrame; pData->sampleRateOut = (ma_uint32)bestFormat.mSampleRate; } /* Clamp the channel count for safety. */ if (pData->channelsOut > MA_MAX_CHANNELS) { pData->channelsOut = MA_MAX_CHANNELS; } /* Internal channel map. This is weird in my testing. If I use the AudioObject to get the channel map, the channel descriptions are set to "Unknown" for some reason. To work around this it looks like retrieving it from the AudioUnit will work. However, and this is where it gets weird, it doesn't seem to work with capture devices, nor at all on iOS... Therefore I'm going to fall back to a default assumption in these cases. */ #if defined(MA_APPLE_DESKTOP) result = ma_get_AudioUnit_channel_map(pContext, pData->audioUnit, deviceType, pData->channelMapOut, pData->channelsOut); if (result != MA_SUCCESS) { #if 0 /* Try falling back to the channel map from the AudioObject. */ result = ma_get_AudioObject_channel_map(pContext, deviceObjectID, deviceType, pData->channelMapOut, pData->channelsOut); if (result != MA_SUCCESS) { return result; } #else /* Fall back to default assumptions. */ ma_channel_map_init_standard(ma_standard_channel_map_default, pData->channelMapOut, ma_countof(pData->channelMapOut), pData->channelsOut); #endif } #else /* TODO: Figure out how to get the channel map using AVAudioSession. */ ma_channel_map_init_standard(ma_standard_channel_map_default, pData->channelMapOut, ma_countof(pData->channelMapOut), pData->channelsOut); #endif /* Buffer size. Not allowing this to be configurable on iOS. */ if (pData->periodSizeInFramesIn == 0) { if (pData->periodSizeInMillisecondsIn == 0) { if (pData->performanceProfile == ma_performance_profile_low_latency) { actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, pData->sampleRateOut); } else { actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, pData->sampleRateOut); } } else { actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->periodSizeInMillisecondsIn, pData->sampleRateOut); } } else { actualPeriodSizeInFrames = pData->periodSizeInFramesIn; } #if defined(MA_APPLE_DESKTOP) result = ma_set_AudioObject_buffer_size_in_frames(pContext, deviceObjectID, deviceType, &actualPeriodSizeInFrames); if (result != MA_SUCCESS) { return result; } #else /* On iOS, the size of the IO buffer needs to be specified in seconds and is a floating point number. I don't trust any potential truncation errors due to converting from float to integer so I'm going to explicitly set the actual period size to the next power of 2. */ @autoreleasepool { AVAudioSession* pAudioSession = [AVAudioSession sharedInstance]; MA_ASSERT(pAudioSession != NULL); [pAudioSession setPreferredIOBufferDuration:((float)actualPeriodSizeInFrames / pAudioSession.sampleRate) error:nil]; actualPeriodSizeInFrames = ma_next_power_of_2((ma_uint32)(pAudioSession.IOBufferDuration * pAudioSession.sampleRate)); } #endif /* During testing I discovered that the buffer size can be too big. You'll get an error like this: kAudioUnitErr_TooManyFramesToProcess : inFramesToProcess=4096, mMaxFramesPerSlice=512 Note how inFramesToProcess is smaller than mMaxFramesPerSlice. To fix, we need to set kAudioUnitProperty_MaximumFramesPerSlice to that of the size of our buffer, or do it the other way around and set our buffer size to the kAudioUnitProperty_MaximumFramesPerSlice. */ status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &actualPeriodSizeInFrames, sizeof(actualPeriodSizeInFrames)); if (status != noErr) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return ma_result_from_OSStatus(status); } pData->periodSizeInFramesOut = (ma_uint32)actualPeriodSizeInFrames; /* We need a buffer list if this is an input device. We render into this in the input callback. */ if (deviceType == ma_device_type_capture) { ma_bool32 isInterleaved = (bestFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) == 0; AudioBufferList* pBufferList; pBufferList = ma_allocate_AudioBufferList__coreaudio(pData->periodSizeInFramesOut, pData->formatOut, pData->channelsOut, (isInterleaved) ? ma_stream_layout_interleaved : ma_stream_layout_deinterleaved, &pContext->allocationCallbacks); if (pBufferList == NULL) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return MA_OUT_OF_MEMORY; } pData->pAudioBufferList = pBufferList; } /* Callbacks. */ callbackInfo.inputProcRefCon = pDevice_DoNotReference; if (deviceType == ma_device_type_playback) { callbackInfo.inputProc = ma_on_output__coreaudio; status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, 0, &callbackInfo, sizeof(callbackInfo)); if (status != noErr) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return ma_result_from_OSStatus(status); } } else { callbackInfo.inputProc = ma_on_input__coreaudio; status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callbackInfo, sizeof(callbackInfo)); if (status != noErr) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return ma_result_from_OSStatus(status); } } /* We need to listen for stop events. */ if (pData->registerStopEvent) { status = ((ma_AudioUnitAddPropertyListener_proc)pContext->coreaudio.AudioUnitAddPropertyListener)(pData->audioUnit, kAudioOutputUnitProperty_IsRunning, on_start_stop__coreaudio, pDevice_DoNotReference); if (status != noErr) { ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return ma_result_from_OSStatus(status); } } /* Initialize the audio unit. */ status = ((ma_AudioUnitInitialize_proc)pContext->coreaudio.AudioUnitInitialize)(pData->audioUnit); if (status != noErr) { ma_free(pData->pAudioBufferList, &pContext->allocationCallbacks); pData->pAudioBufferList = NULL; ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return ma_result_from_OSStatus(status); } /* Grab the name. */ #if defined(MA_APPLE_DESKTOP) ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(pData->deviceName), pData->deviceName); #else if (deviceType == ma_device_type_playback) { ma_strcpy_s(pData->deviceName, sizeof(pData->deviceName), MA_DEFAULT_PLAYBACK_DEVICE_NAME); } else { ma_strcpy_s(pData->deviceName, sizeof(pData->deviceName), MA_DEFAULT_CAPTURE_DEVICE_NAME); } #endif return result; } #if defined(MA_APPLE_DESKTOP) static ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_device_type deviceType, ma_bool32 disposePreviousAudioUnit) { ma_device_init_internal_data__coreaudio data; ma_result result; /* This should only be called for playback or capture, not duplex. */ if (deviceType == ma_device_type_duplex) { return MA_INVALID_ARGS; } data.allowNominalSampleRateChange = MA_FALSE; /* Don't change the nominal sample rate when switching devices. */ if (deviceType == ma_device_type_capture) { data.formatIn = pDevice->capture.format; data.channelsIn = pDevice->capture.channels; data.sampleRateIn = pDevice->sampleRate; MA_COPY_MEMORY(data.channelMapIn, pDevice->capture.channelMap, sizeof(pDevice->capture.channelMap)); data.shareMode = pDevice->capture.shareMode; data.performanceProfile = pDevice->coreaudio.originalPerformanceProfile; data.registerStopEvent = MA_TRUE; if (disposePreviousAudioUnit) { ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture); } if (pDevice->coreaudio.pAudioBufferList) { ma_free(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks); } } else if (deviceType == ma_device_type_playback) { data.formatIn = pDevice->playback.format; data.channelsIn = pDevice->playback.channels; data.sampleRateIn = pDevice->sampleRate; MA_COPY_MEMORY(data.channelMapIn, pDevice->playback.channelMap, sizeof(pDevice->playback.channelMap)); data.shareMode = pDevice->playback.shareMode; data.performanceProfile = pDevice->coreaudio.originalPerformanceProfile; data.registerStopEvent = (pDevice->type != ma_device_type_duplex); if (disposePreviousAudioUnit) { ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); } } data.periodSizeInFramesIn = pDevice->coreaudio.originalPeriodSizeInFrames; data.periodSizeInMillisecondsIn = pDevice->coreaudio.originalPeriodSizeInMilliseconds; data.periodsIn = pDevice->coreaudio.originalPeriods; /* Need at least 3 periods for duplex. */ if (data.periodsIn < 3 && pDevice->type == ma_device_type_duplex) { data.periodsIn = 3; } result = ma_device_init_internal__coreaudio(pDevice->pContext, deviceType, NULL, &data, (void*)pDevice); if (result != MA_SUCCESS) { return result; } if (deviceType == ma_device_type_capture) { #if defined(MA_APPLE_DESKTOP) pDevice->coreaudio.deviceObjectIDCapture = (ma_uint32)data.deviceObjectID; ma_get_AudioObject_uid(pDevice->pContext, pDevice->coreaudio.deviceObjectIDCapture, sizeof(pDevice->capture.id.coreaudio), pDevice->capture.id.coreaudio); #endif pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit; pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList; pDevice->coreaudio.audioBufferCapInFrames = data.periodSizeInFramesOut; pDevice->capture.internalFormat = data.formatOut; pDevice->capture.internalChannels = data.channelsOut; pDevice->capture.internalSampleRate = data.sampleRateOut; MA_COPY_MEMORY(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); pDevice->capture.internalPeriodSizeInFrames = data.periodSizeInFramesOut; pDevice->capture.internalPeriods = data.periodsOut; } else if (deviceType == ma_device_type_playback) { #if defined(MA_APPLE_DESKTOP) pDevice->coreaudio.deviceObjectIDPlayback = (ma_uint32)data.deviceObjectID; ma_get_AudioObject_uid(pDevice->pContext, pDevice->coreaudio.deviceObjectIDPlayback, sizeof(pDevice->playback.id.coreaudio), pDevice->playback.id.coreaudio); #endif pDevice->coreaudio.audioUnitPlayback = (ma_ptr)data.audioUnit; pDevice->playback.internalFormat = data.formatOut; pDevice->playback.internalChannels = data.channelsOut; pDevice->playback.internalSampleRate = data.sampleRateOut; MA_COPY_MEMORY(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); pDevice->playback.internalPeriodSizeInFrames = data.periodSizeInFramesOut; pDevice->playback.internalPeriods = data.periodsOut; } return MA_SUCCESS; } #endif /* MA_APPLE_DESKTOP */ static ma_result ma_device_init__coreaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { ma_result result; MA_ASSERT(pDevice != NULL); MA_ASSERT(pConfig != NULL); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } /* No exclusive mode with the Core Audio backend for now. */ if (((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive) || ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive)) { return MA_SHARE_MODE_NOT_SUPPORTED; } /* Capture needs to be initialized first. */ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { ma_device_init_internal_data__coreaudio data; data.allowNominalSampleRateChange = pConfig->coreaudio.allowNominalSampleRateChange; data.formatIn = pDescriptorCapture->format; data.channelsIn = pDescriptorCapture->channels; data.sampleRateIn = pDescriptorCapture->sampleRate; MA_COPY_MEMORY(data.channelMapIn, pDescriptorCapture->channelMap, sizeof(pDescriptorCapture->channelMap)); data.periodSizeInFramesIn = pDescriptorCapture->periodSizeInFrames; data.periodSizeInMillisecondsIn = pDescriptorCapture->periodSizeInMilliseconds; data.periodsIn = pDescriptorCapture->periodCount; data.shareMode = pDescriptorCapture->shareMode; data.performanceProfile = pConfig->performanceProfile; data.registerStopEvent = MA_TRUE; /* Need at least 3 periods for duplex. */ if (data.periodsIn < 3 && pConfig->deviceType == ma_device_type_duplex) { data.periodsIn = 3; } result = ma_device_init_internal__coreaudio(pDevice->pContext, ma_device_type_capture, pDescriptorCapture->pDeviceID, &data, (void*)pDevice); if (result != MA_SUCCESS) { return result; } pDevice->coreaudio.isDefaultCaptureDevice = (pConfig->capture.pDeviceID == NULL); #if defined(MA_APPLE_DESKTOP) pDevice->coreaudio.deviceObjectIDCapture = (ma_uint32)data.deviceObjectID; #endif pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit; pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList; pDevice->coreaudio.audioBufferCapInFrames = data.periodSizeInFramesOut; pDevice->coreaudio.originalPeriodSizeInFrames = pDescriptorCapture->periodSizeInFrames; pDevice->coreaudio.originalPeriodSizeInMilliseconds = pDescriptorCapture->periodSizeInMilliseconds; pDevice->coreaudio.originalPeriods = pDescriptorCapture->periodCount; pDevice->coreaudio.originalPerformanceProfile = pConfig->performanceProfile; pDescriptorCapture->format = data.formatOut; pDescriptorCapture->channels = data.channelsOut; pDescriptorCapture->sampleRate = data.sampleRateOut; MA_COPY_MEMORY(pDescriptorCapture->channelMap, data.channelMapOut, sizeof(data.channelMapOut)); pDescriptorCapture->periodSizeInFrames = data.periodSizeInFramesOut; pDescriptorCapture->periodCount = data.periodsOut; #if defined(MA_APPLE_DESKTOP) ma_get_AudioObject_uid(pDevice->pContext, pDevice->coreaudio.deviceObjectIDCapture, sizeof(pDevice->capture.id.coreaudio), pDevice->capture.id.coreaudio); /* If we are using the default device we'll need to listen for changes to the system's default device so we can seamlessly switch the device in the background. */ if (pConfig->capture.pDeviceID == NULL) { ma_device__track__coreaudio(pDevice); } #endif } /* Playback. */ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { ma_device_init_internal_data__coreaudio data; data.allowNominalSampleRateChange = pConfig->coreaudio.allowNominalSampleRateChange; data.formatIn = pDescriptorPlayback->format; data.channelsIn = pDescriptorPlayback->channels; data.sampleRateIn = pDescriptorPlayback->sampleRate; MA_COPY_MEMORY(data.channelMapIn, pDescriptorPlayback->channelMap, sizeof(pDescriptorPlayback->channelMap)); data.shareMode = pDescriptorPlayback->shareMode; data.performanceProfile = pConfig->performanceProfile; /* In full-duplex mode we want the playback buffer to be the same size as the capture buffer. */ if (pConfig->deviceType == ma_device_type_duplex) { data.periodSizeInFramesIn = pDescriptorCapture->periodSizeInFrames; data.periodsIn = pDescriptorCapture->periodCount; data.registerStopEvent = MA_FALSE; } else { data.periodSizeInFramesIn = pDescriptorPlayback->periodSizeInFrames; data.periodSizeInMillisecondsIn = pDescriptorPlayback->periodSizeInMilliseconds; data.periodsIn = pDescriptorPlayback->periodCount; data.registerStopEvent = MA_TRUE; } result = ma_device_init_internal__coreaudio(pDevice->pContext, ma_device_type_playback, pDescriptorPlayback->pDeviceID, &data, (void*)pDevice); if (result != MA_SUCCESS) { if (pConfig->deviceType == ma_device_type_duplex) { ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture); if (pDevice->coreaudio.pAudioBufferList) { ma_free(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks); } } return result; } pDevice->coreaudio.isDefaultPlaybackDevice = (pConfig->playback.pDeviceID == NULL); #if defined(MA_APPLE_DESKTOP) pDevice->coreaudio.deviceObjectIDPlayback = (ma_uint32)data.deviceObjectID; #endif pDevice->coreaudio.audioUnitPlayback = (ma_ptr)data.audioUnit; pDevice->coreaudio.originalPeriodSizeInFrames = pDescriptorPlayback->periodSizeInFrames; pDevice->coreaudio.originalPeriodSizeInMilliseconds = pDescriptorPlayback->periodSizeInMilliseconds; pDevice->coreaudio.originalPeriods = pDescriptorPlayback->periodCount; pDevice->coreaudio.originalPerformanceProfile = pConfig->performanceProfile; pDescriptorPlayback->format = data.formatOut; pDescriptorPlayback->channels = data.channelsOut; pDescriptorPlayback->sampleRate = data.sampleRateOut; MA_COPY_MEMORY(pDescriptorPlayback->channelMap, data.channelMapOut, sizeof(data.channelMapOut)); pDescriptorPlayback->periodSizeInFrames = data.periodSizeInFramesOut; pDescriptorPlayback->periodCount = data.periodsOut; #if defined(MA_APPLE_DESKTOP) ma_get_AudioObject_uid(pDevice->pContext, pDevice->coreaudio.deviceObjectIDPlayback, sizeof(pDevice->playback.id.coreaudio), pDevice->playback.id.coreaudio); /* If we are using the default device we'll need to listen for changes to the system's default device so we can seamlessly switch the device in the background. */ if (pDescriptorPlayback->pDeviceID == NULL && (pConfig->deviceType != ma_device_type_duplex || pDescriptorCapture->pDeviceID != NULL)) { ma_device__track__coreaudio(pDevice); } #endif } /* When stopping the device, a callback is called on another thread. We need to wait for this callback before returning from ma_device_stop(). This event is used for this. */ ma_event_init(&pDevice->coreaudio.stopEvent); /* We need to detect when a route has changed so we can update the data conversion pipeline accordingly. This is done differently on non-Desktop Apple platforms. */ #if defined(MA_APPLE_MOBILE) pDevice->coreaudio.pNotificationHandler = (MA_BRIDGE_RETAINED void*)[[ma_ios_notification_handler alloc] init:pDevice]; #endif return MA_SUCCESS; } static ma_result ma_device_start__coreaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { OSStatus status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitCapture); if (status != noErr) { return ma_result_from_OSStatus(status); } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { OSStatus status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); if (status != noErr) { if (pDevice->type == ma_device_type_duplex) { ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); } return ma_result_from_OSStatus(status); } } return MA_SUCCESS; } static ma_result ma_device_stop__coreaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); /* It's not clear from the documentation whether or not AudioOutputUnitStop() actually drains the device or not. */ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { OSStatus status = ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); if (status != noErr) { return ma_result_from_OSStatus(status); } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { OSStatus status = ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); if (status != noErr) { return ma_result_from_OSStatus(status); } } /* We need to wait for the callback to finish before returning. */ ma_event_wait(&pDevice->coreaudio.stopEvent); return MA_SUCCESS; } static ma_result ma_context_uninit__coreaudio(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_coreaudio); #if defined(MA_APPLE_MOBILE) if (!pContext->coreaudio.noAudioSessionDeactivate) { if (![[AVAudioSession sharedInstance] setActive:false error:nil]) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "Failed to deactivate audio session."); return MA_FAILED_TO_INIT_BACKEND; } } #endif #if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE) ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit); ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hCoreAudio); ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hCoreFoundation); #endif #if !defined(MA_APPLE_MOBILE) ma_context__uninit_device_tracking__coreaudio(pContext); #endif (void)pContext; return MA_SUCCESS; } #if defined(MA_APPLE_MOBILE) && defined(__IPHONE_12_0) static AVAudioSessionCategory ma_to_AVAudioSessionCategory(ma_ios_session_category category) { /* The "default" and "none" categories are treated different and should not be used as an input into this function. */ MA_ASSERT(category != ma_ios_session_category_default); MA_ASSERT(category != ma_ios_session_category_none); switch (category) { case ma_ios_session_category_ambient: return AVAudioSessionCategoryAmbient; case ma_ios_session_category_solo_ambient: return AVAudioSessionCategorySoloAmbient; case ma_ios_session_category_playback: return AVAudioSessionCategoryPlayback; case ma_ios_session_category_record: return AVAudioSessionCategoryRecord; case ma_ios_session_category_play_and_record: return AVAudioSessionCategoryPlayAndRecord; case ma_ios_session_category_multi_route: return AVAudioSessionCategoryMultiRoute; case ma_ios_session_category_none: return AVAudioSessionCategoryAmbient; case ma_ios_session_category_default: return AVAudioSessionCategoryAmbient; default: return AVAudioSessionCategoryAmbient; } } #endif static ma_result ma_context_init__coreaudio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { #if !defined(MA_APPLE_MOBILE) ma_result result; #endif MA_ASSERT(pConfig != NULL); MA_ASSERT(pContext != NULL); #if defined(MA_APPLE_MOBILE) @autoreleasepool { AVAudioSession* pAudioSession = [AVAudioSession sharedInstance]; AVAudioSessionCategoryOptions options = pConfig->coreaudio.sessionCategoryOptions; MA_ASSERT(pAudioSession != NULL); if (pConfig->coreaudio.sessionCategory == ma_ios_session_category_default) { /* I'm going to use trial and error to determine our default session category. First we'll try PlayAndRecord. If that fails we'll try Playback and if that fails we'll try record. If all of these fail we'll just not set the category. */ #if !defined(MA_APPLE_TV) && !defined(MA_APPLE_WATCH) options |= AVAudioSessionCategoryOptionDefaultToSpeaker; #endif if ([pAudioSession setCategory: AVAudioSessionCategoryPlayAndRecord withOptions:options error:nil]) { /* Using PlayAndRecord */ } else if ([pAudioSession setCategory: AVAudioSessionCategoryPlayback withOptions:options error:nil]) { /* Using Playback */ } else if ([pAudioSession setCategory: AVAudioSessionCategoryRecord withOptions:options error:nil]) { /* Using Record */ } else { /* Leave as default? */ } } else { if (pConfig->coreaudio.sessionCategory != ma_ios_session_category_none) { #if defined(__IPHONE_12_0) if (![pAudioSession setCategory: ma_to_AVAudioSessionCategory(pConfig->coreaudio.sessionCategory) withOptions:options error:nil]) { return MA_INVALID_OPERATION; /* Failed to set session category. */ } #else /* Ignore the session category on version 11 and older, but post a warning. */ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "Session category only supported in iOS 12 and newer."); #endif } } if (!pConfig->coreaudio.noAudioSessionActivate) { if (![pAudioSession setActive:true error:nil]) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "Failed to activate audio session."); return MA_FAILED_TO_INIT_BACKEND; } } } #endif #if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE) pContext->coreaudio.hCoreFoundation = ma_dlopen(ma_context_get_log(pContext), "/System/Library/Frameworks/CoreFoundation.framework/CoreFoundation"); if (pContext->coreaudio.hCoreFoundation == NULL) { return MA_API_NOT_FOUND; } pContext->coreaudio.CFStringGetCString = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hCoreFoundation, "CFStringGetCString"); pContext->coreaudio.CFRelease = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hCoreFoundation, "CFRelease"); pContext->coreaudio.hCoreAudio = ma_dlopen(ma_context_get_log(pContext), "/System/Library/Frameworks/CoreAudio.framework/CoreAudio"); if (pContext->coreaudio.hCoreAudio == NULL) { ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hCoreFoundation); return MA_API_NOT_FOUND; } pContext->coreaudio.AudioObjectGetPropertyData = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyData"); pContext->coreaudio.AudioObjectGetPropertyDataSize = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyDataSize"); pContext->coreaudio.AudioObjectSetPropertyData = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hCoreAudio, "AudioObjectSetPropertyData"); pContext->coreaudio.AudioObjectAddPropertyListener = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hCoreAudio, "AudioObjectAddPropertyListener"); pContext->coreaudio.AudioObjectRemovePropertyListener = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hCoreAudio, "AudioObjectRemovePropertyListener"); /* It looks like Apple has moved some APIs from AudioUnit into AudioToolbox on more recent versions of macOS. They are still defined in AudioUnit, but just in case they decide to remove them from there entirely I'm going to implement a fallback. The way it'll work is that it'll first try AudioUnit, and if the required symbols are not present there we'll fall back to AudioToolbox. */ pContext->coreaudio.hAudioUnit = ma_dlopen(ma_context_get_log(pContext), "/System/Library/Frameworks/AudioUnit.framework/AudioUnit"); if (pContext->coreaudio.hAudioUnit == NULL) { ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hCoreAudio); ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hCoreFoundation); return MA_API_NOT_FOUND; } if (ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioComponentFindNext") == NULL) { /* Couldn't find the required symbols in AudioUnit, so fall back to AudioToolbox. */ ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit); pContext->coreaudio.hAudioUnit = ma_dlopen(ma_context_get_log(pContext), "/System/Library/Frameworks/AudioToolbox.framework/AudioToolbox"); if (pContext->coreaudio.hAudioUnit == NULL) { ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hCoreAudio); ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hCoreFoundation); return MA_API_NOT_FOUND; } } pContext->coreaudio.AudioComponentFindNext = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioComponentFindNext"); pContext->coreaudio.AudioComponentInstanceDispose = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioComponentInstanceDispose"); pContext->coreaudio.AudioComponentInstanceNew = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioComponentInstanceNew"); pContext->coreaudio.AudioOutputUnitStart = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioOutputUnitStart"); pContext->coreaudio.AudioOutputUnitStop = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioOutputUnitStop"); pContext->coreaudio.AudioUnitAddPropertyListener = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioUnitAddPropertyListener"); pContext->coreaudio.AudioUnitGetPropertyInfo = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioUnitGetPropertyInfo"); pContext->coreaudio.AudioUnitGetProperty = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioUnitGetProperty"); pContext->coreaudio.AudioUnitSetProperty = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioUnitSetProperty"); pContext->coreaudio.AudioUnitInitialize = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioUnitInitialize"); pContext->coreaudio.AudioUnitRender = ma_dlsym(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit, "AudioUnitRender"); #else pContext->coreaudio.CFStringGetCString = (ma_proc)CFStringGetCString; pContext->coreaudio.CFRelease = (ma_proc)CFRelease; #if defined(MA_APPLE_DESKTOP) pContext->coreaudio.AudioObjectGetPropertyData = (ma_proc)AudioObjectGetPropertyData; pContext->coreaudio.AudioObjectGetPropertyDataSize = (ma_proc)AudioObjectGetPropertyDataSize; pContext->coreaudio.AudioObjectSetPropertyData = (ma_proc)AudioObjectSetPropertyData; pContext->coreaudio.AudioObjectAddPropertyListener = (ma_proc)AudioObjectAddPropertyListener; pContext->coreaudio.AudioObjectRemovePropertyListener = (ma_proc)AudioObjectRemovePropertyListener; #endif pContext->coreaudio.AudioComponentFindNext = (ma_proc)AudioComponentFindNext; pContext->coreaudio.AudioComponentInstanceDispose = (ma_proc)AudioComponentInstanceDispose; pContext->coreaudio.AudioComponentInstanceNew = (ma_proc)AudioComponentInstanceNew; pContext->coreaudio.AudioOutputUnitStart = (ma_proc)AudioOutputUnitStart; pContext->coreaudio.AudioOutputUnitStop = (ma_proc)AudioOutputUnitStop; pContext->coreaudio.AudioUnitAddPropertyListener = (ma_proc)AudioUnitAddPropertyListener; pContext->coreaudio.AudioUnitGetPropertyInfo = (ma_proc)AudioUnitGetPropertyInfo; pContext->coreaudio.AudioUnitGetProperty = (ma_proc)AudioUnitGetProperty; pContext->coreaudio.AudioUnitSetProperty = (ma_proc)AudioUnitSetProperty; pContext->coreaudio.AudioUnitInitialize = (ma_proc)AudioUnitInitialize; pContext->coreaudio.AudioUnitRender = (ma_proc)AudioUnitRender; #endif /* Audio component. */ { AudioComponentDescription desc; desc.componentType = kAudioUnitType_Output; #if defined(MA_APPLE_DESKTOP) desc.componentSubType = kAudioUnitSubType_HALOutput; #else desc.componentSubType = kAudioUnitSubType_RemoteIO; #endif desc.componentManufacturer = kAudioUnitManufacturer_Apple; desc.componentFlags = 0; desc.componentFlagsMask = 0; pContext->coreaudio.component = ((ma_AudioComponentFindNext_proc)pContext->coreaudio.AudioComponentFindNext)(NULL, &desc); if (pContext->coreaudio.component == NULL) { #if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE) ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit); ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hCoreAudio); ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hCoreFoundation); #endif return MA_FAILED_TO_INIT_BACKEND; } } #if !defined(MA_APPLE_MOBILE) result = ma_context__init_device_tracking__coreaudio(pContext); if (result != MA_SUCCESS) { #if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE) ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hAudioUnit); ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hCoreAudio); ma_dlclose(ma_context_get_log(pContext), pContext->coreaudio.hCoreFoundation); #endif return result; } #endif pContext->coreaudio.noAudioSessionDeactivate = pConfig->coreaudio.noAudioSessionDeactivate; pCallbacks->onContextInit = ma_context_init__coreaudio; pCallbacks->onContextUninit = ma_context_uninit__coreaudio; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__coreaudio; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__coreaudio; pCallbacks->onDeviceInit = ma_device_init__coreaudio; pCallbacks->onDeviceUninit = ma_device_uninit__coreaudio; pCallbacks->onDeviceStart = ma_device_start__coreaudio; pCallbacks->onDeviceStop = ma_device_stop__coreaudio; pCallbacks->onDeviceRead = NULL; pCallbacks->onDeviceWrite = NULL; pCallbacks->onDeviceDataLoop = NULL; return MA_SUCCESS; } #endif /* MA_HAS_COREAUDIO */ /****************************************************************************** sndio Backend ******************************************************************************/ #ifdef MA_HAS_SNDIO #include <fcntl.h> /* Only supporting OpenBSD. This did not work very well at all on FreeBSD when I tried it. Not sure if this is due to miniaudio's implementation or if it's some kind of system configuration issue, but basically the default device just doesn't emit any sound, or at times you'll hear tiny pieces. I will consider enabling this when there's demand for it or if I can get it tested and debugged more thoroughly. */ #if 0 #if defined(__NetBSD__) || defined(__OpenBSD__) #include <sys/audioio.h> #endif #if defined(__FreeBSD__) || defined(__DragonFly__) #include <sys/soundcard.h> #endif #endif #define MA_SIO_DEVANY "default" #define MA_SIO_PLAY 1 #define MA_SIO_REC 2 #define MA_SIO_NENC 8 #define MA_SIO_NCHAN 8 #define MA_SIO_NRATE 16 #define MA_SIO_NCONF 4 struct ma_sio_hdl; /* <-- Opaque */ struct ma_sio_par { unsigned int bits; unsigned int bps; unsigned int sig; unsigned int le; unsigned int msb; unsigned int rchan; unsigned int pchan; unsigned int rate; unsigned int bufsz; unsigned int xrun; unsigned int round; unsigned int appbufsz; int __pad[3]; unsigned int __magic; }; struct ma_sio_enc { unsigned int bits; unsigned int bps; unsigned int sig; unsigned int le; unsigned int msb; }; struct ma_sio_conf { unsigned int enc; unsigned int rchan; unsigned int pchan; unsigned int rate; }; struct ma_sio_cap { struct ma_sio_enc enc[MA_SIO_NENC]; unsigned int rchan[MA_SIO_NCHAN]; unsigned int pchan[MA_SIO_NCHAN]; unsigned int rate[MA_SIO_NRATE]; int __pad[7]; unsigned int nconf; struct ma_sio_conf confs[MA_SIO_NCONF]; }; typedef struct ma_sio_hdl* (* ma_sio_open_proc) (const char*, unsigned int, int); typedef void (* ma_sio_close_proc) (struct ma_sio_hdl*); typedef int (* ma_sio_setpar_proc) (struct ma_sio_hdl*, struct ma_sio_par*); typedef int (* ma_sio_getpar_proc) (struct ma_sio_hdl*, struct ma_sio_par*); typedef int (* ma_sio_getcap_proc) (struct ma_sio_hdl*, struct ma_sio_cap*); typedef size_t (* ma_sio_write_proc) (struct ma_sio_hdl*, const void*, size_t); typedef size_t (* ma_sio_read_proc) (struct ma_sio_hdl*, void*, size_t); typedef int (* ma_sio_start_proc) (struct ma_sio_hdl*); typedef int (* ma_sio_stop_proc) (struct ma_sio_hdl*); typedef int (* ma_sio_initpar_proc)(struct ma_sio_par*); static ma_uint32 ma_get_standard_sample_rate_priority_index__sndio(ma_uint32 sampleRate) /* Lower = higher priority */ { ma_uint32 i; for (i = 0; i < ma_countof(g_maStandardSampleRatePriorities); ++i) { if (g_maStandardSampleRatePriorities[i] == sampleRate) { return i; } } return (ma_uint32)-1; } static ma_format ma_format_from_sio_enc__sndio(unsigned int bits, unsigned int bps, unsigned int sig, unsigned int le, unsigned int msb) { /* We only support native-endian right now. */ if ((ma_is_little_endian() && le == 0) || (ma_is_big_endian() && le == 1)) { return ma_format_unknown; } if (bits == 8 && bps == 1 && sig == 0) { return ma_format_u8; } if (bits == 16 && bps == 2 && sig == 1) { return ma_format_s16; } if (bits == 24 && bps == 3 && sig == 1) { return ma_format_s24; } if (bits == 24 && bps == 4 && sig == 1 && msb == 0) { /*return ma_format_s24_32;*/ } if (bits == 32 && bps == 4 && sig == 1) { return ma_format_s32; } return ma_format_unknown; } static ma_format ma_find_best_format_from_sio_cap__sndio(struct ma_sio_cap* caps) { ma_format bestFormat; unsigned int iConfig; MA_ASSERT(caps != NULL); bestFormat = ma_format_unknown; for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) { unsigned int iEncoding; for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { unsigned int bits; unsigned int bps; unsigned int sig; unsigned int le; unsigned int msb; ma_format format; if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) { continue; } bits = caps->enc[iEncoding].bits; bps = caps->enc[iEncoding].bps; sig = caps->enc[iEncoding].sig; le = caps->enc[iEncoding].le; msb = caps->enc[iEncoding].msb; format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); if (format == ma_format_unknown) { continue; /* Format not supported. */ } if (bestFormat == ma_format_unknown) { bestFormat = format; } else { if (ma_get_format_priority_index(bestFormat) > ma_get_format_priority_index(format)) { /* <-- Lower = better. */ bestFormat = format; } } } } return bestFormat; } static ma_uint32 ma_find_best_channels_from_sio_cap__sndio(struct ma_sio_cap* caps, ma_device_type deviceType, ma_format requiredFormat) { ma_uint32 maxChannels; unsigned int iConfig; MA_ASSERT(caps != NULL); MA_ASSERT(requiredFormat != ma_format_unknown); /* Just pick whatever configuration has the most channels. */ maxChannels = 0; for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) { /* The encoding should be of requiredFormat. */ unsigned int iEncoding; for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { unsigned int iChannel; unsigned int bits; unsigned int bps; unsigned int sig; unsigned int le; unsigned int msb; ma_format format; if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) { continue; } bits = caps->enc[iEncoding].bits; bps = caps->enc[iEncoding].bps; sig = caps->enc[iEncoding].sig; le = caps->enc[iEncoding].le; msb = caps->enc[iEncoding].msb; format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); if (format != requiredFormat) { continue; } /* Getting here means the format is supported. Iterate over each channel count and grab the biggest one. */ for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) { unsigned int chan = 0; unsigned int channels; if (deviceType == ma_device_type_playback) { chan = caps->confs[iConfig].pchan; } else { chan = caps->confs[iConfig].rchan; } if ((chan & (1UL << iChannel)) == 0) { continue; } if (deviceType == ma_device_type_playback) { channels = caps->pchan[iChannel]; } else { channels = caps->rchan[iChannel]; } if (maxChannels < channels) { maxChannels = channels; } } } } return maxChannels; } static ma_uint32 ma_find_best_sample_rate_from_sio_cap__sndio(struct ma_sio_cap* caps, ma_device_type deviceType, ma_format requiredFormat, ma_uint32 requiredChannels) { ma_uint32 firstSampleRate; ma_uint32 bestSampleRate; unsigned int iConfig; MA_ASSERT(caps != NULL); MA_ASSERT(requiredFormat != ma_format_unknown); MA_ASSERT(requiredChannels > 0); MA_ASSERT(requiredChannels <= MA_MAX_CHANNELS); firstSampleRate = 0; /* <-- If the device does not support a standard rate we'll fall back to the first one that's found. */ bestSampleRate = 0; for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) { /* The encoding should be of requiredFormat. */ unsigned int iEncoding; for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { unsigned int iChannel; unsigned int bits; unsigned int bps; unsigned int sig; unsigned int le; unsigned int msb; ma_format format; if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) { continue; } bits = caps->enc[iEncoding].bits; bps = caps->enc[iEncoding].bps; sig = caps->enc[iEncoding].sig; le = caps->enc[iEncoding].le; msb = caps->enc[iEncoding].msb; format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); if (format != requiredFormat) { continue; } /* Getting here means the format is supported. Iterate over each channel count and grab the biggest one. */ for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) { unsigned int chan = 0; unsigned int channels; unsigned int iRate; if (deviceType == ma_device_type_playback) { chan = caps->confs[iConfig].pchan; } else { chan = caps->confs[iConfig].rchan; } if ((chan & (1UL << iChannel)) == 0) { continue; } if (deviceType == ma_device_type_playback) { channels = caps->pchan[iChannel]; } else { channels = caps->rchan[iChannel]; } if (channels != requiredChannels) { continue; } /* Getting here means we have found a compatible encoding/channel pair. */ for (iRate = 0; iRate < MA_SIO_NRATE; iRate += 1) { ma_uint32 rate = (ma_uint32)caps->rate[iRate]; ma_uint32 ratePriority; if (firstSampleRate == 0) { firstSampleRate = rate; } /* Disregard this rate if it's not a standard one. */ ratePriority = ma_get_standard_sample_rate_priority_index__sndio(rate); if (ratePriority == (ma_uint32)-1) { continue; } if (ma_get_standard_sample_rate_priority_index__sndio(bestSampleRate) > ratePriority) { /* Lower = better. */ bestSampleRate = rate; } } } } } /* If a standard sample rate was not found just fall back to the first one that was iterated. */ if (bestSampleRate == 0) { bestSampleRate = firstSampleRate; } return bestSampleRate; } static ma_result ma_context_enumerate_devices__sndio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { ma_bool32 isTerminating = MA_FALSE; struct ma_sio_hdl* handle; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); /* sndio doesn't seem to have a good device enumeration API, so I'm therefore only enumerating over default devices for now. */ /* Playback. */ if (!isTerminating) { handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(MA_SIO_DEVANY, MA_SIO_PLAY, 0); if (handle != NULL) { /* Supports playback. */ ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_strcpy_s(deviceInfo.id.sndio, sizeof(deviceInfo.id.sndio), MA_SIO_DEVANY); ma_strcpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME); isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); ((ma_sio_close_proc)pContext->sndio.sio_close)(handle); } } /* Capture. */ if (!isTerminating) { handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(MA_SIO_DEVANY, MA_SIO_REC, 0); if (handle != NULL) { /* Supports capture. */ ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_strcpy_s(deviceInfo.id.sndio, sizeof(deviceInfo.id.sndio), "default"); ma_strcpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME); isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); ((ma_sio_close_proc)pContext->sndio.sio_close)(handle); } } return MA_SUCCESS; } static ma_result ma_context_get_device_info__sndio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { char devid[256]; struct ma_sio_hdl* handle; struct ma_sio_cap caps; unsigned int iConfig; MA_ASSERT(pContext != NULL); /* We need to open the device before we can get information about it. */ if (pDeviceID == NULL) { ma_strcpy_s(devid, sizeof(devid), MA_SIO_DEVANY); ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (deviceType == ma_device_type_playback) ? MA_DEFAULT_PLAYBACK_DEVICE_NAME : MA_DEFAULT_CAPTURE_DEVICE_NAME); } else { ma_strcpy_s(devid, sizeof(devid), pDeviceID->sndio); ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), devid); } handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(devid, (deviceType == ma_device_type_playback) ? MA_SIO_PLAY : MA_SIO_REC, 0); if (handle == NULL) { return MA_NO_DEVICE; } if (((ma_sio_getcap_proc)pContext->sndio.sio_getcap)(handle, &caps) == 0) { return MA_ERROR; } pDeviceInfo->nativeDataFormatCount = 0; for (iConfig = 0; iConfig < caps.nconf; iConfig += 1) { /* The main thing we care about is that the encoding is supported by miniaudio. If it is, we want to give preference to some formats over others. */ unsigned int iEncoding; unsigned int iChannel; unsigned int iRate; for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { unsigned int bits; unsigned int bps; unsigned int sig; unsigned int le; unsigned int msb; ma_format format; if ((caps.confs[iConfig].enc & (1UL << iEncoding)) == 0) { continue; } bits = caps.enc[iEncoding].bits; bps = caps.enc[iEncoding].bps; sig = caps.enc[iEncoding].sig; le = caps.enc[iEncoding].le; msb = caps.enc[iEncoding].msb; format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); if (format == ma_format_unknown) { continue; /* Format not supported. */ } /* Channels. */ for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) { unsigned int chan = 0; unsigned int channels; if (deviceType == ma_device_type_playback) { chan = caps.confs[iConfig].pchan; } else { chan = caps.confs[iConfig].rchan; } if ((chan & (1UL << iChannel)) == 0) { continue; } if (deviceType == ma_device_type_playback) { channels = caps.pchan[iChannel]; } else { channels = caps.rchan[iChannel]; } /* Sample Rates. */ for (iRate = 0; iRate < MA_SIO_NRATE; iRate += 1) { if ((caps.confs[iConfig].rate & (1UL << iRate)) != 0) { ma_device_info_add_native_data_format(pDeviceInfo, format, channels, caps.rate[iRate], 0); } } } } } ((ma_sio_close_proc)pContext->sndio.sio_close)(handle); return MA_SUCCESS; } static ma_result ma_device_uninit__sndio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)pDevice->sndio.handleCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); } return MA_SUCCESS; } static ma_result ma_device_init_handle__sndio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType) { const char* pDeviceName; ma_ptr handle; int openFlags = 0; struct ma_sio_cap caps; struct ma_sio_par par; const ma_device_id* pDeviceID; ma_format format; ma_uint32 channels; ma_uint32 sampleRate; ma_format internalFormat; ma_uint32 internalChannels; ma_uint32 internalSampleRate; ma_uint32 internalPeriodSizeInFrames; ma_uint32 internalPeriods; MA_ASSERT(pConfig != NULL); MA_ASSERT(deviceType != ma_device_type_duplex); MA_ASSERT(pDevice != NULL); if (deviceType == ma_device_type_capture) { openFlags = MA_SIO_REC; } else { openFlags = MA_SIO_PLAY; } pDeviceID = pDescriptor->pDeviceID; format = pDescriptor->format; channels = pDescriptor->channels; sampleRate = pDescriptor->sampleRate; pDeviceName = MA_SIO_DEVANY; if (pDeviceID != NULL) { pDeviceName = pDeviceID->sndio; } handle = (ma_ptr)((ma_sio_open_proc)pDevice->pContext->sndio.sio_open)(pDeviceName, openFlags, 0); if (handle == NULL) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to open device."); return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } /* We need to retrieve the device caps to determine the most appropriate format to use. */ if (((ma_sio_getcap_proc)pDevice->pContext->sndio.sio_getcap)((struct ma_sio_hdl*)handle, &caps) == 0) { ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)handle); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to retrieve device caps."); return MA_ERROR; } /* Note: sndio reports a huge range of available channels. This is inconvenient for us because there's no real way, as far as I can tell, to get the _actual_ channel count of the device. I'm therefore restricting this to the requested channels, regardless of whether or not the default channel count is requested. For hardware devices, I'm suspecting only a single channel count will be reported and we can safely use the value returned by ma_find_best_channels_from_sio_cap__sndio(). */ if (deviceType == ma_device_type_capture) { if (format == ma_format_unknown) { format = ma_find_best_format_from_sio_cap__sndio(&caps); } if (channels == 0) { if (strlen(pDeviceName) > strlen("rsnd/") && strncmp(pDeviceName, "rsnd/", strlen("rsnd/")) == 0) { channels = ma_find_best_channels_from_sio_cap__sndio(&caps, deviceType, format); } else { channels = MA_DEFAULT_CHANNELS; } } } else { if (format == ma_format_unknown) { format = ma_find_best_format_from_sio_cap__sndio(&caps); } if (channels == 0) { if (strlen(pDeviceName) > strlen("rsnd/") && strncmp(pDeviceName, "rsnd/", strlen("rsnd/")) == 0) { channels = ma_find_best_channels_from_sio_cap__sndio(&caps, deviceType, format); } else { channels = MA_DEFAULT_CHANNELS; } } } if (sampleRate == 0) { sampleRate = ma_find_best_sample_rate_from_sio_cap__sndio(&caps, pConfig->deviceType, format, channels); } ((ma_sio_initpar_proc)pDevice->pContext->sndio.sio_initpar)(&par); par.msb = 0; par.le = ma_is_little_endian(); switch (format) { case ma_format_u8: { par.bits = 8; par.bps = 1; par.sig = 0; } break; case ma_format_s24: { par.bits = 24; par.bps = 3; par.sig = 1; } break; case ma_format_s32: { par.bits = 32; par.bps = 4; par.sig = 1; } break; case ma_format_s16: case ma_format_f32: case ma_format_unknown: default: { par.bits = 16; par.bps = 2; par.sig = 1; } break; } if (deviceType == ma_device_type_capture) { par.rchan = channels; } else { par.pchan = channels; } par.rate = sampleRate; internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, par.rate, pConfig->performanceProfile); par.round = internalPeriodSizeInFrames; par.appbufsz = par.round * pDescriptor->periodCount; if (((ma_sio_setpar_proc)pDevice->pContext->sndio.sio_setpar)((struct ma_sio_hdl*)handle, &par) == 0) { ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)handle); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to set buffer size."); return MA_ERROR; } if (((ma_sio_getpar_proc)pDevice->pContext->sndio.sio_getpar)((struct ma_sio_hdl*)handle, &par) == 0) { ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)handle); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to retrieve buffer size."); return MA_ERROR; } internalFormat = ma_format_from_sio_enc__sndio(par.bits, par.bps, par.sig, par.le, par.msb); internalChannels = (deviceType == ma_device_type_capture) ? par.rchan : par.pchan; internalSampleRate = par.rate; internalPeriods = par.appbufsz / par.round; internalPeriodSizeInFrames = par.round; if (deviceType == ma_device_type_capture) { pDevice->sndio.handleCapture = handle; } else { pDevice->sndio.handlePlayback = handle; } pDescriptor->format = internalFormat; pDescriptor->channels = internalChannels; pDescriptor->sampleRate = internalSampleRate; ma_channel_map_init_standard(ma_standard_channel_map_sndio, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), internalChannels); pDescriptor->periodSizeInFrames = internalPeriodSizeInFrames; pDescriptor->periodCount = internalPeriods; return MA_SUCCESS; } static ma_result ma_device_init__sndio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { MA_ASSERT(pDevice != NULL); MA_ZERO_OBJECT(&pDevice->sndio); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { ma_result result = ma_device_init_handle__sndio(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture); if (result != MA_SUCCESS) { return result; } } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { ma_result result = ma_device_init_handle__sndio(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback); if (result != MA_SUCCESS) { return result; } } return MA_SUCCESS; } static ma_result ma_device_start__sndio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handleCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); /* <-- Doesn't actually playback until data is written. */ } return MA_SUCCESS; } static ma_result ma_device_stop__sndio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); /* From the documentation: The sio_stop() function puts the audio subsystem in the same state as before sio_start() is called. It stops recording, drains the play buffer and then stops playback. If samples to play are queued but playback hasn't started yet then playback is forced immediately; playback will actually stop once the buffer is drained. In no case are samples in the play buffer discarded. Therefore, sio_stop() performs all of the necessary draining for us. */ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handleCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); } return MA_SUCCESS; } static ma_result ma_device_write__sndio(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { int result; if (pFramesWritten != NULL) { *pFramesWritten = 0; } result = ((ma_sio_write_proc)pDevice->pContext->sndio.sio_write)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); if (result == 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to send data from the client to the device."); return MA_IO_ERROR; } if (pFramesWritten != NULL) { *pFramesWritten = frameCount; } return MA_SUCCESS; } static ma_result ma_device_read__sndio(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) { int result; if (pFramesRead != NULL) { *pFramesRead = 0; } result = ((ma_sio_read_proc)pDevice->pContext->sndio.sio_read)((struct ma_sio_hdl*)pDevice->sndio.handleCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); if (result == 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to read data from the device to be sent to the device."); return MA_IO_ERROR; } if (pFramesRead != NULL) { *pFramesRead = frameCount; } return MA_SUCCESS; } static ma_result ma_context_uninit__sndio(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_sndio); (void)pContext; return MA_SUCCESS; } static ma_result ma_context_init__sndio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { #ifndef MA_NO_RUNTIME_LINKING const char* libsndioNames[] = { "libsndio.so" }; size_t i; for (i = 0; i < ma_countof(libsndioNames); ++i) { pContext->sndio.sndioSO = ma_dlopen(ma_context_get_log(pContext), libsndioNames[i]); if (pContext->sndio.sndioSO != NULL) { break; } } if (pContext->sndio.sndioSO == NULL) { return MA_NO_BACKEND; } pContext->sndio.sio_open = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->sndio.sndioSO, "sio_open"); pContext->sndio.sio_close = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->sndio.sndioSO, "sio_close"); pContext->sndio.sio_setpar = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->sndio.sndioSO, "sio_setpar"); pContext->sndio.sio_getpar = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->sndio.sndioSO, "sio_getpar"); pContext->sndio.sio_getcap = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->sndio.sndioSO, "sio_getcap"); pContext->sndio.sio_write = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->sndio.sndioSO, "sio_write"); pContext->sndio.sio_read = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->sndio.sndioSO, "sio_read"); pContext->sndio.sio_start = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->sndio.sndioSO, "sio_start"); pContext->sndio.sio_stop = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->sndio.sndioSO, "sio_stop"); pContext->sndio.sio_initpar = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->sndio.sndioSO, "sio_initpar"); #else pContext->sndio.sio_open = sio_open; pContext->sndio.sio_close = sio_close; pContext->sndio.sio_setpar = sio_setpar; pContext->sndio.sio_getpar = sio_getpar; pContext->sndio.sio_getcap = sio_getcap; pContext->sndio.sio_write = sio_write; pContext->sndio.sio_read = sio_read; pContext->sndio.sio_start = sio_start; pContext->sndio.sio_stop = sio_stop; pContext->sndio.sio_initpar = sio_initpar; #endif pCallbacks->onContextInit = ma_context_init__sndio; pCallbacks->onContextUninit = ma_context_uninit__sndio; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__sndio; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__sndio; pCallbacks->onDeviceInit = ma_device_init__sndio; pCallbacks->onDeviceUninit = ma_device_uninit__sndio; pCallbacks->onDeviceStart = ma_device_start__sndio; pCallbacks->onDeviceStop = ma_device_stop__sndio; pCallbacks->onDeviceRead = ma_device_read__sndio; pCallbacks->onDeviceWrite = ma_device_write__sndio; pCallbacks->onDeviceDataLoop = NULL; (void)pConfig; return MA_SUCCESS; } #endif /* MA_HAS_SNDIO */ /****************************************************************************** audio(4) Backend ******************************************************************************/ #ifdef MA_HAS_AUDIO4 #include <fcntl.h> #include <poll.h> #include <errno.h> #include <sys/stat.h> #include <sys/types.h> #include <sys/ioctl.h> #include <sys/audioio.h> #ifdef __NetBSD__ #include <sys/param.h> #endif #if defined(__OpenBSD__) #include <sys/param.h> #if defined(OpenBSD) && OpenBSD >= 201709 #define MA_AUDIO4_USE_NEW_API #endif #endif static void ma_construct_device_id__audio4(char* id, size_t idSize, const char* base, int deviceIndex) { size_t baseLen; MA_ASSERT(id != NULL); MA_ASSERT(idSize > 0); MA_ASSERT(deviceIndex >= 0); baseLen = strlen(base); MA_ASSERT(idSize > baseLen); ma_strcpy_s(id, idSize, base); ma_itoa_s(deviceIndex, id+baseLen, idSize-baseLen, 10); } static ma_result ma_extract_device_index_from_id__audio4(const char* id, const char* base, int* pIndexOut) { size_t idLen; size_t baseLen; const char* deviceIndexStr; MA_ASSERT(id != NULL); MA_ASSERT(base != NULL); MA_ASSERT(pIndexOut != NULL); idLen = strlen(id); baseLen = strlen(base); if (idLen <= baseLen) { return MA_ERROR; /* Doesn't look like the id starts with the base. */ } if (strncmp(id, base, baseLen) != 0) { return MA_ERROR; /* ID does not begin with base. */ } deviceIndexStr = id + baseLen; if (deviceIndexStr[0] == '\0') { return MA_ERROR; /* No index specified in the ID. */ } if (pIndexOut) { *pIndexOut = atoi(deviceIndexStr); } return MA_SUCCESS; } #if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */ static ma_format ma_format_from_encoding__audio4(unsigned int encoding, unsigned int precision) { if (precision == 8 && (encoding == AUDIO_ENCODING_ULINEAR || encoding == AUDIO_ENCODING_ULINEAR || encoding == AUDIO_ENCODING_ULINEAR_LE || encoding == AUDIO_ENCODING_ULINEAR_BE)) { return ma_format_u8; } else { if (ma_is_little_endian() && encoding == AUDIO_ENCODING_SLINEAR_LE) { if (precision == 16) { return ma_format_s16; } else if (precision == 24) { return ma_format_s24; } else if (precision == 32) { return ma_format_s32; } } else if (ma_is_big_endian() && encoding == AUDIO_ENCODING_SLINEAR_BE) { if (precision == 16) { return ma_format_s16; } else if (precision == 24) { return ma_format_s24; } else if (precision == 32) { return ma_format_s32; } } } return ma_format_unknown; /* Encoding not supported. */ } static void ma_encoding_from_format__audio4(ma_format format, unsigned int* pEncoding, unsigned int* pPrecision) { MA_ASSERT(pEncoding != NULL); MA_ASSERT(pPrecision != NULL); switch (format) { case ma_format_u8: { *pEncoding = AUDIO_ENCODING_ULINEAR; *pPrecision = 8; } break; case ma_format_s24: { *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE; *pPrecision = 24; } break; case ma_format_s32: { *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE; *pPrecision = 32; } break; case ma_format_s16: case ma_format_f32: case ma_format_unknown: default: { *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE; *pPrecision = 16; } break; } } static ma_format ma_format_from_prinfo__audio4(struct audio_prinfo* prinfo) { return ma_format_from_encoding__audio4(prinfo->encoding, prinfo->precision); } static ma_format ma_best_format_from_fd__audio4(int fd, ma_format preferredFormat) { audio_encoding_t encoding; ma_uint32 iFormat; int counter = 0; /* First check to see if the preferred format is supported. */ if (preferredFormat != ma_format_unknown) { counter = 0; for (;;) { MA_ZERO_OBJECT(&encoding); encoding.index = counter; if (ioctl(fd, AUDIO_GETENC, &encoding) < 0) { break; } if (preferredFormat == ma_format_from_encoding__audio4(encoding.encoding, encoding.precision)) { return preferredFormat; /* Found the preferred format. */ } /* Getting here means this encoding does not match our preferred format so we need to more on to the next encoding. */ counter += 1; } } /* Getting here means our preferred format is not supported, so fall back to our standard priorities. */ for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); iFormat += 1) { ma_format format = g_maFormatPriorities[iFormat]; counter = 0; for (;;) { MA_ZERO_OBJECT(&encoding); encoding.index = counter; if (ioctl(fd, AUDIO_GETENC, &encoding) < 0) { break; } if (format == ma_format_from_encoding__audio4(encoding.encoding, encoding.precision)) { return format; /* Found a workable format. */ } /* Getting here means this encoding does not match our preferred format so we need to more on to the next encoding. */ counter += 1; } } /* Getting here means not appropriate format was found. */ return ma_format_unknown; } #else static ma_format ma_format_from_swpar__audio4(struct audio_swpar* par) { if (par->bits == 8 && par->bps == 1 && par->sig == 0) { return ma_format_u8; } if (par->bits == 16 && par->bps == 2 && par->sig == 1 && par->le == ma_is_little_endian()) { return ma_format_s16; } if (par->bits == 24 && par->bps == 3 && par->sig == 1 && par->le == ma_is_little_endian()) { return ma_format_s24; } if (par->bits == 32 && par->bps == 4 && par->sig == 1 && par->le == ma_is_little_endian()) { return ma_format_f32; } /* Format not supported. */ return ma_format_unknown; } #endif static ma_result ma_context_get_device_info_from_fd__audio4(ma_context* pContext, ma_device_type deviceType, int fd, ma_device_info* pDeviceInfo) { audio_device_t fdDevice; MA_ASSERT(pContext != NULL); MA_ASSERT(fd >= 0); MA_ASSERT(pDeviceInfo != NULL); (void)pContext; (void)deviceType; if (ioctl(fd, AUDIO_GETDEV, &fdDevice) < 0) { return MA_ERROR; /* Failed to retrieve device info. */ } /* Name. */ ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), fdDevice.name); #if !defined(MA_AUDIO4_USE_NEW_API) { audio_info_t fdInfo; int counter = 0; ma_uint32 channels; ma_uint32 sampleRate; #if defined(__NetBSD__) && (__NetBSD_Version__ >= 900000000) if (ioctl(fd, AUDIO_GETFORMAT, &fdInfo) < 0) { return MA_ERROR; } #else if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) { return MA_ERROR; } #endif if (deviceType == ma_device_type_playback) { channels = fdInfo.play.channels; sampleRate = fdInfo.play.sample_rate; } else { channels = fdInfo.record.channels; sampleRate = fdInfo.record.sample_rate; } /* Supported formats. We get this by looking at the encodings. */ pDeviceInfo->nativeDataFormatCount = 0; for (;;) { audio_encoding_t encoding; ma_format format; MA_ZERO_OBJECT(&encoding); encoding.index = counter; if (ioctl(fd, AUDIO_GETENC, &encoding) < 0) { break; } format = ma_format_from_encoding__audio4(encoding.encoding, encoding.precision); if (format != ma_format_unknown) { ma_device_info_add_native_data_format(pDeviceInfo, format, channels, sampleRate, 0); } counter += 1; } } #else { struct audio_swpar fdPar; ma_format format; ma_uint32 channels; ma_uint32 sampleRate; if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) { return MA_ERROR; } format = ma_format_from_swpar__audio4(&fdPar); if (format == ma_format_unknown) { return MA_FORMAT_NOT_SUPPORTED; } if (deviceType == ma_device_type_playback) { channels = fdPar.pchan; } else { channels = fdPar.rchan; } sampleRate = fdPar.rate; pDeviceInfo->nativeDataFormatCount = 0; ma_device_info_add_native_data_format(pDeviceInfo, format, channels, sampleRate, 0); } #endif return MA_SUCCESS; } static ma_result ma_context_enumerate_devices__audio4(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { const int maxDevices = 64; char devpath[256]; int iDevice; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); /* Every device will be named "/dev/audioN", with a "/dev/audioctlN" equivalent. We use the "/dev/audioctlN" version here since we can open it even when another process has control of the "/dev/audioN" device. */ for (iDevice = 0; iDevice < maxDevices; ++iDevice) { struct stat st; int fd; ma_bool32 isTerminating = MA_FALSE; ma_strcpy_s(devpath, sizeof(devpath), "/dev/audioctl"); ma_itoa_s(iDevice, devpath+strlen(devpath), sizeof(devpath)-strlen(devpath), 10); if (stat(devpath, &st) < 0) { break; } /* The device exists, but we need to check if it's usable as playback and/or capture. */ /* Playback. */ if (!isTerminating) { fd = open(devpath, O_RDONLY, 0); if (fd >= 0) { /* Supports playback. */ ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_construct_device_id__audio4(deviceInfo.id.audio4, sizeof(deviceInfo.id.audio4), "/dev/audio", iDevice); if (ma_context_get_device_info_from_fd__audio4(pContext, ma_device_type_playback, fd, &deviceInfo) == MA_SUCCESS) { isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } close(fd); } } /* Capture. */ if (!isTerminating) { fd = open(devpath, O_WRONLY, 0); if (fd >= 0) { /* Supports capture. */ ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_construct_device_id__audio4(deviceInfo.id.audio4, sizeof(deviceInfo.id.audio4), "/dev/audio", iDevice); if (ma_context_get_device_info_from_fd__audio4(pContext, ma_device_type_capture, fd, &deviceInfo) == MA_SUCCESS) { isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); } close(fd); } } if (isTerminating) { break; } } return MA_SUCCESS; } static ma_result ma_context_get_device_info__audio4(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { int fd = -1; int deviceIndex = -1; char ctlid[256]; ma_result result; MA_ASSERT(pContext != NULL); /* We need to open the "/dev/audioctlN" device to get the info. To do this we need to extract the number from the device ID which will be in "/dev/audioN" format. */ if (pDeviceID == NULL) { /* Default device. */ ma_strcpy_s(ctlid, sizeof(ctlid), "/dev/audioctl"); } else { /* Specific device. We need to convert from "/dev/audioN" to "/dev/audioctlN". */ result = ma_extract_device_index_from_id__audio4(pDeviceID->audio4, "/dev/audio", &deviceIndex); if (result != MA_SUCCESS) { return result; } ma_construct_device_id__audio4(ctlid, sizeof(ctlid), "/dev/audioctl", deviceIndex); } fd = open(ctlid, (deviceType == ma_device_type_playback) ? O_WRONLY : O_RDONLY, 0); if (fd == -1) { return MA_NO_DEVICE; } if (deviceIndex == -1) { ma_strcpy_s(pDeviceInfo->id.audio4, sizeof(pDeviceInfo->id.audio4), "/dev/audio"); } else { ma_construct_device_id__audio4(pDeviceInfo->id.audio4, sizeof(pDeviceInfo->id.audio4), "/dev/audio", deviceIndex); } result = ma_context_get_device_info_from_fd__audio4(pContext, deviceType, fd, pDeviceInfo); close(fd); return result; } static ma_result ma_device_uninit__audio4(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { close(pDevice->audio4.fdCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { close(pDevice->audio4.fdPlayback); } return MA_SUCCESS; } static ma_result ma_device_init_fd__audio4(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType) { const char* pDefaultDeviceNames[] = { "/dev/audio", "/dev/audio0" }; const char* pDefaultDeviceCtlNames[] = { "/dev/audioctl", "/dev/audioctl0" }; int fd; int fdFlags = 0; size_t iDefaultDevice = (size_t)-1; ma_format internalFormat; ma_uint32 internalChannels; ma_uint32 internalSampleRate; ma_uint32 internalPeriodSizeInFrames; ma_uint32 internalPeriods; MA_ASSERT(pConfig != NULL); MA_ASSERT(deviceType != ma_device_type_duplex); MA_ASSERT(pDevice != NULL); /* The first thing to do is open the file. */ if (deviceType == ma_device_type_capture) { fdFlags = O_RDONLY; } else { fdFlags = O_WRONLY; } /*fdFlags |= O_NONBLOCK;*/ /* Find the index of the default device as a start. We'll use this index later. Set it to (size_t)-1 otherwise. */ if (pDescriptor->pDeviceID == NULL) { /* Default device. */ for (iDefaultDevice = 0; iDefaultDevice < ma_countof(pDefaultDeviceNames); ++iDefaultDevice) { fd = open(pDefaultDeviceNames[iDefaultDevice], fdFlags, 0); if (fd != -1) { break; } } } else { /* Specific device. */ fd = open(pDescriptor->pDeviceID->audio4, fdFlags, 0); for (iDefaultDevice = 0; iDefaultDevice < ma_countof(pDefaultDeviceNames); iDefaultDevice += 1) { if (ma_strcmp(pDefaultDeviceNames[iDefaultDevice], pDescriptor->pDeviceID->audio4) == 0) { break; } } if (iDefaultDevice == ma_countof(pDefaultDeviceNames)) { iDefaultDevice = (size_t)-1; } } if (fd == -1) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to open device."); return ma_result_from_errno(errno); } #if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */ { audio_info_t fdInfo; int fdInfoResult = -1; /* The documentation is a little bit unclear to me as to how it handles formats. It says the following: Regardless of formats supported by underlying driver, the audio driver accepts the following formats. By then the next sentence says this: `encoding` and `precision` are one of the values obtained by AUDIO_GETENC. It sounds like a direct contradiction to me. I'm going to play this safe any only use the best sample format returned by AUDIO_GETENC. If the requested format is supported we'll use that, but otherwise we'll just use our standard format priorities to pick an appropriate one. */ AUDIO_INITINFO(&fdInfo); /* Get the default format from the audioctl file if we're asking for a default device. If we retrieve it from /dev/audio it'll default to mono 8000Hz. */ if (iDefaultDevice != (size_t)-1) { /* We're using a default device. Get the info from the /dev/audioctl file instead of /dev/audio. */ int fdctl = open(pDefaultDeviceCtlNames[iDefaultDevice], fdFlags, 0); if (fdctl != -1) { #if defined(__NetBSD__) && (__NetBSD_Version__ >= 900000000) fdInfoResult = ioctl(fdctl, AUDIO_GETFORMAT, &fdInfo); #else fdInfoResult = ioctl(fdctl, AUDIO_GETINFO, &fdInfo); #endif close(fdctl); } } if (fdInfoResult == -1) { /* We still don't have the default device info so just retrieve it from the main audio device. */ if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] AUDIO_GETINFO failed."); return ma_result_from_errno(errno); } } /* We get the driver to do as much of the data conversion as possible. */ if (deviceType == ma_device_type_capture) { fdInfo.mode = AUMODE_RECORD; ma_encoding_from_format__audio4(ma_best_format_from_fd__audio4(fd, pDescriptor->format), &fdInfo.record.encoding, &fdInfo.record.precision); if (pDescriptor->channels != 0) { fdInfo.record.channels = ma_clamp(pDescriptor->channels, 1, 12); /* From the documentation: `channels` ranges from 1 to 12. */ } if (pDescriptor->sampleRate != 0) { fdInfo.record.sample_rate = ma_clamp(pDescriptor->sampleRate, 1000, 192000); /* From the documentation: `frequency` ranges from 1000Hz to 192000Hz. (They mean `sample_rate` instead of `frequency`.) */ } } else { fdInfo.mode = AUMODE_PLAY; ma_encoding_from_format__audio4(ma_best_format_from_fd__audio4(fd, pDescriptor->format), &fdInfo.play.encoding, &fdInfo.play.precision); if (pDescriptor->channels != 0) { fdInfo.play.channels = ma_clamp(pDescriptor->channels, 1, 12); /* From the documentation: `channels` ranges from 1 to 12. */ } if (pDescriptor->sampleRate != 0) { fdInfo.play.sample_rate = ma_clamp(pDescriptor->sampleRate, 1000, 192000); /* From the documentation: `frequency` ranges from 1000Hz to 192000Hz. (They mean `sample_rate` instead of `frequency`.) */ } } if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device format. AUDIO_SETINFO failed."); return ma_result_from_errno(errno); } if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] AUDIO_GETINFO failed."); return ma_result_from_errno(errno); } if (deviceType == ma_device_type_capture) { internalFormat = ma_format_from_prinfo__audio4(&fdInfo.record); internalChannels = fdInfo.record.channels; internalSampleRate = fdInfo.record.sample_rate; } else { internalFormat = ma_format_from_prinfo__audio4(&fdInfo.play); internalChannels = fdInfo.play.channels; internalSampleRate = fdInfo.play.sample_rate; } if (internalFormat == ma_format_unknown) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable."); return MA_FORMAT_NOT_SUPPORTED; } /* Buffer. */ { ma_uint32 internalPeriodSizeInBytes; internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile); internalPeriodSizeInBytes = internalPeriodSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels); if (internalPeriodSizeInBytes < 16) { internalPeriodSizeInBytes = 16; } internalPeriods = pDescriptor->periodCount; if (internalPeriods < 2) { internalPeriods = 2; } /* What miniaudio calls a period, audio4 calls a block. */ AUDIO_INITINFO(&fdInfo); fdInfo.hiwat = internalPeriods; fdInfo.lowat = internalPeriods-1; fdInfo.blocksize = internalPeriodSizeInBytes; if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to set internal buffer size. AUDIO_SETINFO failed."); return ma_result_from_errno(errno); } internalPeriods = fdInfo.hiwat; internalPeriodSizeInFrames = fdInfo.blocksize / ma_get_bytes_per_frame(internalFormat, internalChannels); } } #else { struct audio_swpar fdPar; /* We need to retrieve the format of the device so we can know the channel count and sample rate. Then we can calculate the buffer size. */ if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve initial device parameters."); return ma_result_from_errno(errno); } internalFormat = ma_format_from_swpar__audio4(&fdPar); internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan; internalSampleRate = fdPar.rate; if (internalFormat == ma_format_unknown) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable."); return MA_FORMAT_NOT_SUPPORTED; } /* Buffer. */ { ma_uint32 internalPeriodSizeInBytes; internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile); /* What miniaudio calls a period, audio4 calls a block. */ internalPeriodSizeInBytes = internalPeriodSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels); if (internalPeriodSizeInBytes < 16) { internalPeriodSizeInBytes = 16; } fdPar.nblks = pDescriptor->periodCount; fdPar.round = internalPeriodSizeInBytes; if (ioctl(fd, AUDIO_SETPAR, &fdPar) < 0) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device parameters."); return ma_result_from_errno(errno); } if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve actual device parameters."); return ma_result_from_errno(errno); } } internalFormat = ma_format_from_swpar__audio4(&fdPar); internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan; internalSampleRate = fdPar.rate; internalPeriods = fdPar.nblks; internalPeriodSizeInFrames = fdPar.round / ma_get_bytes_per_frame(internalFormat, internalChannels); } #endif if (internalFormat == ma_format_unknown) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable."); return MA_FORMAT_NOT_SUPPORTED; } if (deviceType == ma_device_type_capture) { pDevice->audio4.fdCapture = fd; } else { pDevice->audio4.fdPlayback = fd; } pDescriptor->format = internalFormat; pDescriptor->channels = internalChannels; pDescriptor->sampleRate = internalSampleRate; ma_channel_map_init_standard(ma_standard_channel_map_sound4, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), internalChannels); pDescriptor->periodSizeInFrames = internalPeriodSizeInFrames; pDescriptor->periodCount = internalPeriods; return MA_SUCCESS; } static ma_result ma_device_init__audio4(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { MA_ASSERT(pDevice != NULL); MA_ZERO_OBJECT(&pDevice->audio4); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } pDevice->audio4.fdCapture = -1; pDevice->audio4.fdPlayback = -1; /* The version of the operating system dictates whether or not the device is exclusive or shared. NetBSD introduced in-kernel mixing which means it's shared. All other BSD flavours are exclusive as far as I'm aware. */ #if defined(__NetBSD_Version__) && __NetBSD_Version__ >= 800000000 /* NetBSD 8.0+ */ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) || ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) { return MA_SHARE_MODE_NOT_SUPPORTED; } #else /* All other flavors. */ #endif if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { ma_result result = ma_device_init_fd__audio4(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture); if (result != MA_SUCCESS) { return result; } } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { ma_result result = ma_device_init_fd__audio4(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback); if (result != MA_SUCCESS) { if (pConfig->deviceType == ma_device_type_duplex) { close(pDevice->audio4.fdCapture); } return result; } } return MA_SUCCESS; } static ma_result ma_device_start__audio4(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { if (pDevice->audio4.fdCapture == -1) { return MA_INVALID_ARGS; } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { if (pDevice->audio4.fdPlayback == -1) { return MA_INVALID_ARGS; } } return MA_SUCCESS; } static ma_result ma_device_stop_fd__audio4(ma_device* pDevice, int fd) { if (fd == -1) { return MA_INVALID_ARGS; } #if !defined(MA_AUDIO4_USE_NEW_API) if (ioctl(fd, AUDIO_FLUSH, 0) < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_FLUSH failed."); return ma_result_from_errno(errno); } #else if (ioctl(fd, AUDIO_STOP, 0) < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_STOP failed."); return ma_result_from_errno(errno); } #endif return MA_SUCCESS; } static ma_result ma_device_stop__audio4(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ma_result result; result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdCapture); if (result != MA_SUCCESS) { return result; } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ma_result result; /* Drain the device first. If this fails we'll just need to flush without draining. Unfortunately draining isn't available on newer version of OpenBSD. */ #if !defined(MA_AUDIO4_USE_NEW_API) ioctl(pDevice->audio4.fdPlayback, AUDIO_DRAIN, 0); #endif /* Here is where the device is stopped immediately. */ result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdPlayback); if (result != MA_SUCCESS) { return result; } } return MA_SUCCESS; } static ma_result ma_device_write__audio4(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { int result; if (pFramesWritten != NULL) { *pFramesWritten = 0; } result = write(pDevice->audio4.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); if (result < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to write data to the device."); return ma_result_from_errno(errno); } if (pFramesWritten != NULL) { *pFramesWritten = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); } return MA_SUCCESS; } static ma_result ma_device_read__audio4(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) { int result; if (pFramesRead != NULL) { *pFramesRead = 0; } result = read(pDevice->audio4.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); if (result < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to read data from the device."); return ma_result_from_errno(errno); } if (pFramesRead != NULL) { *pFramesRead = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); } return MA_SUCCESS; } static ma_result ma_context_uninit__audio4(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_audio4); (void)pContext; return MA_SUCCESS; } static ma_result ma_context_init__audio4(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { MA_ASSERT(pContext != NULL); (void)pConfig; pCallbacks->onContextInit = ma_context_init__audio4; pCallbacks->onContextUninit = ma_context_uninit__audio4; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__audio4; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__audio4; pCallbacks->onDeviceInit = ma_device_init__audio4; pCallbacks->onDeviceUninit = ma_device_uninit__audio4; pCallbacks->onDeviceStart = ma_device_start__audio4; pCallbacks->onDeviceStop = ma_device_stop__audio4; pCallbacks->onDeviceRead = ma_device_read__audio4; pCallbacks->onDeviceWrite = ma_device_write__audio4; pCallbacks->onDeviceDataLoop = NULL; return MA_SUCCESS; } #endif /* MA_HAS_AUDIO4 */ /****************************************************************************** OSS Backend ******************************************************************************/ #ifdef MA_HAS_OSS #include <sys/ioctl.h> #include <unistd.h> #include <fcntl.h> #include <sys/soundcard.h> #ifndef SNDCTL_DSP_HALT #define SNDCTL_DSP_HALT SNDCTL_DSP_RESET #endif #define MA_OSS_DEFAULT_DEVICE_NAME "/dev/dsp" static int ma_open_temp_device__oss(void) { /* The OSS sample code uses "/dev/mixer" as the device for getting system properties so I'm going to do the same. */ int fd = open("/dev/mixer", O_RDONLY, 0); if (fd >= 0) { return fd; } return -1; } static ma_result ma_context_open_device__oss(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, int* pfd) { const char* deviceName; int flags; MA_ASSERT(pContext != NULL); MA_ASSERT(pfd != NULL); (void)pContext; *pfd = -1; /* This function should only be called for playback or capture, not duplex. */ if (deviceType == ma_device_type_duplex) { return MA_INVALID_ARGS; } deviceName = MA_OSS_DEFAULT_DEVICE_NAME; if (pDeviceID != NULL) { deviceName = pDeviceID->oss; } flags = (deviceType == ma_device_type_playback) ? O_WRONLY : O_RDONLY; if (shareMode == ma_share_mode_exclusive) { flags |= O_EXCL; } *pfd = open(deviceName, flags, 0); if (*pfd == -1) { return ma_result_from_errno(errno); } return MA_SUCCESS; } static ma_result ma_context_enumerate_devices__oss(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { int fd; oss_sysinfo si; int result; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); fd = ma_open_temp_device__oss(); if (fd == -1) { ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open a temporary device for retrieving system information used for device enumeration."); return MA_NO_BACKEND; } result = ioctl(fd, SNDCTL_SYSINFO, &si); if (result != -1) { int iAudioDevice; for (iAudioDevice = 0; iAudioDevice < si.numaudios; ++iAudioDevice) { oss_audioinfo ai; ai.dev = iAudioDevice; result = ioctl(fd, SNDCTL_AUDIOINFO, &ai); if (result != -1) { if (ai.devnode[0] != '\0') { /* <-- Can be blank, according to documentation. */ ma_device_info deviceInfo; ma_bool32 isTerminating = MA_FALSE; MA_ZERO_OBJECT(&deviceInfo); /* ID */ ma_strncpy_s(deviceInfo.id.oss, sizeof(deviceInfo.id.oss), ai.devnode, (size_t)-1); /* The human readable device name should be in the "ai.handle" variable, but it can sometimes be empty in which case we just fall back to "ai.name" which is less user friendly, but usually has a value. */ if (ai.handle[0] != '\0') { ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ai.handle, (size_t)-1); } else { ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ai.name, (size_t)-1); } /* The device can be both playback and capture. */ if (!isTerminating && (ai.caps & PCM_CAP_OUTPUT) != 0) { isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } if (!isTerminating && (ai.caps & PCM_CAP_INPUT) != 0) { isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); } if (isTerminating) { break; } } } } } else { close(fd); ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve system information for device enumeration."); return MA_NO_BACKEND; } close(fd); return MA_SUCCESS; } static void ma_context_add_native_data_format__oss(ma_context* pContext, oss_audioinfo* pAudioInfo, ma_format format, ma_device_info* pDeviceInfo) { unsigned int minChannels; unsigned int maxChannels; unsigned int iRate; MA_ASSERT(pContext != NULL); MA_ASSERT(pAudioInfo != NULL); MA_ASSERT(pDeviceInfo != NULL); /* If we support all channels we just report 0. */ minChannels = ma_clamp(pAudioInfo->min_channels, MA_MIN_CHANNELS, MA_MAX_CHANNELS); maxChannels = ma_clamp(pAudioInfo->max_channels, MA_MIN_CHANNELS, MA_MAX_CHANNELS); /* OSS has this annoying thing where sample rates can be reported in two ways. We prefer explicitness, which OSS has in the form of nrates/rates, however there are times where nrates can be 0, in which case we'll need to use min_rate and max_rate and report only standard rates. */ if (pAudioInfo->nrates > 0) { for (iRate = 0; iRate < pAudioInfo->nrates; iRate += 1) { unsigned int rate = pAudioInfo->rates[iRate]; if (minChannels == MA_MIN_CHANNELS && maxChannels == MA_MAX_CHANNELS) { ma_device_info_add_native_data_format(pDeviceInfo, format, 0, rate, 0); /* Set the channel count to 0 to indicate that all channel counts are supported. */ } else { unsigned int iChannel; for (iChannel = minChannels; iChannel <= maxChannels; iChannel += 1) { ma_device_info_add_native_data_format(pDeviceInfo, format, iChannel, rate, 0); } } } } else { for (iRate = 0; iRate < ma_countof(g_maStandardSampleRatePriorities); iRate += 1) { ma_uint32 standardRate = g_maStandardSampleRatePriorities[iRate]; if (standardRate >= (ma_uint32)pAudioInfo->min_rate && standardRate <= (ma_uint32)pAudioInfo->max_rate) { if (minChannels == MA_MIN_CHANNELS && maxChannels == MA_MAX_CHANNELS) { ma_device_info_add_native_data_format(pDeviceInfo, format, 0, standardRate, 0); /* Set the channel count to 0 to indicate that all channel counts are supported. */ } else { unsigned int iChannel; for (iChannel = minChannels; iChannel <= maxChannels; iChannel += 1) { ma_device_info_add_native_data_format(pDeviceInfo, format, iChannel, standardRate, 0); } } } } } } static ma_result ma_context_get_device_info__oss(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { ma_bool32 foundDevice; int fdTemp; oss_sysinfo si; int result; MA_ASSERT(pContext != NULL); /* Handle the default device a little differently. */ if (pDeviceID == NULL) { if (deviceType == ma_device_type_playback) { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); } else { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); } return MA_SUCCESS; } /* If we get here it means we are _not_ using the default device. */ foundDevice = MA_FALSE; fdTemp = ma_open_temp_device__oss(); if (fdTemp == -1) { ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open a temporary device for retrieving system information used for device enumeration."); return MA_NO_BACKEND; } result = ioctl(fdTemp, SNDCTL_SYSINFO, &si); if (result != -1) { int iAudioDevice; for (iAudioDevice = 0; iAudioDevice < si.numaudios; ++iAudioDevice) { oss_audioinfo ai; ai.dev = iAudioDevice; result = ioctl(fdTemp, SNDCTL_AUDIOINFO, &ai); if (result != -1) { if (ma_strcmp(ai.devnode, pDeviceID->oss) == 0) { /* It has the same name, so now just confirm the type. */ if ((deviceType == ma_device_type_playback && ((ai.caps & PCM_CAP_OUTPUT) != 0)) || (deviceType == ma_device_type_capture && ((ai.caps & PCM_CAP_INPUT) != 0))) { unsigned int formatMask; /* ID */ ma_strncpy_s(pDeviceInfo->id.oss, sizeof(pDeviceInfo->id.oss), ai.devnode, (size_t)-1); /* The human readable device name should be in the "ai.handle" variable, but it can sometimes be empty in which case we just fall back to "ai.name" which is less user friendly, but usually has a value. */ if (ai.handle[0] != '\0') { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), ai.handle, (size_t)-1); } else { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), ai.name, (size_t)-1); } pDeviceInfo->nativeDataFormatCount = 0; if (deviceType == ma_device_type_playback) { formatMask = ai.oformats; } else { formatMask = ai.iformats; } if (((formatMask & AFMT_S16_LE) != 0 && ma_is_little_endian()) || (AFMT_S16_BE && ma_is_big_endian())) { ma_context_add_native_data_format__oss(pContext, &ai, ma_format_s16, pDeviceInfo); } if (((formatMask & AFMT_S32_LE) != 0 && ma_is_little_endian()) || (AFMT_S32_BE && ma_is_big_endian())) { ma_context_add_native_data_format__oss(pContext, &ai, ma_format_s32, pDeviceInfo); } if ((formatMask & AFMT_U8) != 0) { ma_context_add_native_data_format__oss(pContext, &ai, ma_format_u8, pDeviceInfo); } foundDevice = MA_TRUE; break; } } } } } else { close(fdTemp); ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve system information for device enumeration."); return MA_NO_BACKEND; } close(fdTemp); if (!foundDevice) { return MA_NO_DEVICE; } return MA_SUCCESS; } static ma_result ma_device_uninit__oss(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { close(pDevice->oss.fdCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { close(pDevice->oss.fdPlayback); } return MA_SUCCESS; } static int ma_format_to_oss(ma_format format) { int ossFormat = AFMT_U8; switch (format) { case ma_format_s16: ossFormat = (ma_is_little_endian()) ? AFMT_S16_LE : AFMT_S16_BE; break; case ma_format_s24: ossFormat = (ma_is_little_endian()) ? AFMT_S32_LE : AFMT_S32_BE; break; case ma_format_s32: ossFormat = (ma_is_little_endian()) ? AFMT_S32_LE : AFMT_S32_BE; break; case ma_format_f32: ossFormat = (ma_is_little_endian()) ? AFMT_S16_LE : AFMT_S16_BE; break; case ma_format_u8: default: ossFormat = AFMT_U8; break; } return ossFormat; } static ma_format ma_format_from_oss(int ossFormat) { if (ossFormat == AFMT_U8) { return ma_format_u8; } else { if (ma_is_little_endian()) { switch (ossFormat) { case AFMT_S16_LE: return ma_format_s16; case AFMT_S32_LE: return ma_format_s32; default: return ma_format_unknown; } } else { switch (ossFormat) { case AFMT_S16_BE: return ma_format_s16; case AFMT_S32_BE: return ma_format_s32; default: return ma_format_unknown; } } } return ma_format_unknown; } static ma_result ma_device_init_fd__oss(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType) { ma_result result; int ossResult; int fd; const ma_device_id* pDeviceID = NULL; ma_share_mode shareMode; int ossFormat; int ossChannels; int ossSampleRate; int ossFragment; MA_ASSERT(pDevice != NULL); MA_ASSERT(pConfig != NULL); MA_ASSERT(deviceType != ma_device_type_duplex); pDeviceID = pDescriptor->pDeviceID; shareMode = pDescriptor->shareMode; ossFormat = ma_format_to_oss((pDescriptor->format != ma_format_unknown) ? pDescriptor->format : ma_format_s16); /* Use s16 by default because OSS doesn't like floating point. */ ossChannels = (int)(pDescriptor->channels > 0) ? pDescriptor->channels : MA_DEFAULT_CHANNELS; ossSampleRate = (int)(pDescriptor->sampleRate > 0) ? pDescriptor->sampleRate : MA_DEFAULT_SAMPLE_RATE; result = ma_context_open_device__oss(pDevice->pContext, deviceType, pDeviceID, shareMode, &fd); if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device."); return result; } /* The OSS documentation is very clear about the order we should be initializing the device's properties: 1) Format 2) Channels 3) Sample rate. */ /* Format. */ ossResult = ioctl(fd, SNDCTL_DSP_SETFMT, &ossFormat); if (ossResult == -1) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set format."); return ma_result_from_errno(errno); } /* Channels. */ ossResult = ioctl(fd, SNDCTL_DSP_CHANNELS, &ossChannels); if (ossResult == -1) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set channel count."); return ma_result_from_errno(errno); } /* Sample Rate. */ ossResult = ioctl(fd, SNDCTL_DSP_SPEED, &ossSampleRate); if (ossResult == -1) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set sample rate."); return ma_result_from_errno(errno); } /* Buffer. The documentation says that the fragment settings should be set as soon as possible, but I'm not sure if it should be done before or after format/channels/rate. OSS wants the fragment size in bytes and a power of 2. When setting, we specify the power, not the actual value. */ { ma_uint32 periodSizeInFrames; ma_uint32 periodSizeInBytes; ma_uint32 ossFragmentSizePower; periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, (ma_uint32)ossSampleRate, pConfig->performanceProfile); periodSizeInBytes = ma_round_to_power_of_2(periodSizeInFrames * ma_get_bytes_per_frame(ma_format_from_oss(ossFormat), ossChannels)); if (periodSizeInBytes < 16) { periodSizeInBytes = 16; } ossFragmentSizePower = 4; periodSizeInBytes >>= 4; while (periodSizeInBytes >>= 1) { ossFragmentSizePower += 1; } ossFragment = (int)((pConfig->periods << 16) | ossFragmentSizePower); ossResult = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &ossFragment); if (ossResult == -1) { close(fd); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set fragment size and period count."); return ma_result_from_errno(errno); } } /* Internal settings. */ if (deviceType == ma_device_type_capture) { pDevice->oss.fdCapture = fd; } else { pDevice->oss.fdPlayback = fd; } pDescriptor->format = ma_format_from_oss(ossFormat); pDescriptor->channels = ossChannels; pDescriptor->sampleRate = ossSampleRate; ma_channel_map_init_standard(ma_standard_channel_map_sound4, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), pDescriptor->channels); pDescriptor->periodCount = (ma_uint32)(ossFragment >> 16); pDescriptor->periodSizeInFrames = (ma_uint32)(1 << (ossFragment & 0xFFFF)) / ma_get_bytes_per_frame(pDescriptor->format, pDescriptor->channels); if (pDescriptor->format == ma_format_unknown) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] The device's internal format is not supported by miniaudio."); return MA_FORMAT_NOT_SUPPORTED; } return MA_SUCCESS; } static ma_result ma_device_init__oss(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { MA_ASSERT(pDevice != NULL); MA_ASSERT(pConfig != NULL); MA_ZERO_OBJECT(&pDevice->oss); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { ma_result result = ma_device_init_fd__oss(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture); if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device."); return result; } } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { ma_result result = ma_device_init_fd__oss(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback); if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device."); return result; } } return MA_SUCCESS; } /* Note on Starting and Stopping ============================= In the past I was using SNDCTL_DSP_HALT to stop the device, however this results in issues when trying to resume the device again. If we use SNDCTL_DSP_HALT, the next write() or read() will fail. Instead what we need to do is just not write or read to and from the device when the device is not running. As a result, both the start and stop functions for OSS are just empty stubs. The starting and stopping logic is handled by ma_device_write__oss() and ma_device_read__oss(). These will check the device state, and if the device is stopped they will simply not do any kind of processing. The downside to this technique is that I've noticed a fairly lengthy delay in stopping the device, up to a second. This is on a virtual machine, and as such might just be due to the virtual drivers, but I'm not fully sure. I am not sure how to work around this problem so for the moment that's just how it's going to have to be. When starting the device, OSS will automatically start it when write() or read() is called. */ static ma_result ma_device_start__oss(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); /* The device is automatically started with reading and writing. */ (void)pDevice; return MA_SUCCESS; } static ma_result ma_device_stop__oss(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); /* See note above on why this is empty. */ (void)pDevice; return MA_SUCCESS; } static ma_result ma_device_write__oss(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { int resultOSS; ma_uint32 deviceState; if (pFramesWritten != NULL) { *pFramesWritten = 0; } /* Don't do any processing if the device is stopped. */ deviceState = ma_device_get_state(pDevice); if (deviceState != ma_device_state_started && deviceState != ma_device_state_starting) { return MA_SUCCESS; } resultOSS = write(pDevice->oss.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); if (resultOSS < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to send data from the client to the device."); return ma_result_from_errno(errno); } if (pFramesWritten != NULL) { *pFramesWritten = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); } return MA_SUCCESS; } static ma_result ma_device_read__oss(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) { int resultOSS; ma_uint32 deviceState; if (pFramesRead != NULL) { *pFramesRead = 0; } /* Don't do any processing if the device is stopped. */ deviceState = ma_device_get_state(pDevice); if (deviceState != ma_device_state_started && deviceState != ma_device_state_starting) { return MA_SUCCESS; } resultOSS = read(pDevice->oss.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); if (resultOSS < 0) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to read data from the device to be sent to the client."); return ma_result_from_errno(errno); } if (pFramesRead != NULL) { *pFramesRead = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); } return MA_SUCCESS; } static ma_result ma_context_uninit__oss(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_oss); (void)pContext; return MA_SUCCESS; } static ma_result ma_context_init__oss(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { int fd; int ossVersion; int result; MA_ASSERT(pContext != NULL); (void)pConfig; /* Try opening a temporary device first so we can get version information. This is closed at the end. */ fd = ma_open_temp_device__oss(); if (fd == -1) { ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open temporary device for retrieving system properties."); /* Looks liks OSS isn't installed, or there are no available devices. */ return MA_NO_BACKEND; } /* Grab the OSS version. */ ossVersion = 0; result = ioctl(fd, OSS_GETVERSION, &ossVersion); if (result == -1) { close(fd); ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve OSS version."); return MA_NO_BACKEND; } /* The file handle to temp device is no longer needed. Close ASAP. */ close(fd); pContext->oss.versionMajor = ((ossVersion & 0xFF0000) >> 16); pContext->oss.versionMinor = ((ossVersion & 0x00FF00) >> 8); pCallbacks->onContextInit = ma_context_init__oss; pCallbacks->onContextUninit = ma_context_uninit__oss; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__oss; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__oss; pCallbacks->onDeviceInit = ma_device_init__oss; pCallbacks->onDeviceUninit = ma_device_uninit__oss; pCallbacks->onDeviceStart = ma_device_start__oss; pCallbacks->onDeviceStop = ma_device_stop__oss; pCallbacks->onDeviceRead = ma_device_read__oss; pCallbacks->onDeviceWrite = ma_device_write__oss; pCallbacks->onDeviceDataLoop = NULL; return MA_SUCCESS; } #endif /* MA_HAS_OSS */ /****************************************************************************** AAudio Backend ******************************************************************************/ #ifdef MA_HAS_AAUDIO #ifdef MA_NO_RUNTIME_LINKING #include <AAudio/AAudio.h> #endif typedef int32_t ma_aaudio_result_t; typedef int32_t ma_aaudio_direction_t; typedef int32_t ma_aaudio_sharing_mode_t; typedef int32_t ma_aaudio_format_t; typedef int32_t ma_aaudio_stream_state_t; typedef int32_t ma_aaudio_performance_mode_t; typedef int32_t ma_aaudio_usage_t; typedef int32_t ma_aaudio_content_type_t; typedef int32_t ma_aaudio_input_preset_t; typedef int32_t ma_aaudio_allowed_capture_policy_t; typedef int32_t ma_aaudio_data_callback_result_t; typedef struct ma_AAudioStreamBuilder_t* ma_AAudioStreamBuilder; typedef struct ma_AAudioStream_t* ma_AAudioStream; #define MA_AAUDIO_UNSPECIFIED 0 /* Result codes. miniaudio only cares about the success code. */ #define MA_AAUDIO_OK 0 /* Directions. */ #define MA_AAUDIO_DIRECTION_OUTPUT 0 #define MA_AAUDIO_DIRECTION_INPUT 1 /* Sharing modes. */ #define MA_AAUDIO_SHARING_MODE_EXCLUSIVE 0 #define MA_AAUDIO_SHARING_MODE_SHARED 1 /* Formats. */ #define MA_AAUDIO_FORMAT_PCM_I16 1 #define MA_AAUDIO_FORMAT_PCM_FLOAT 2 /* Stream states. */ #define MA_AAUDIO_STREAM_STATE_UNINITIALIZED 0 #define MA_AAUDIO_STREAM_STATE_UNKNOWN 1 #define MA_AAUDIO_STREAM_STATE_OPEN 2 #define MA_AAUDIO_STREAM_STATE_STARTING 3 #define MA_AAUDIO_STREAM_STATE_STARTED 4 #define MA_AAUDIO_STREAM_STATE_PAUSING 5 #define MA_AAUDIO_STREAM_STATE_PAUSED 6 #define MA_AAUDIO_STREAM_STATE_FLUSHING 7 #define MA_AAUDIO_STREAM_STATE_FLUSHED 8 #define MA_AAUDIO_STREAM_STATE_STOPPING 9 #define MA_AAUDIO_STREAM_STATE_STOPPED 10 #define MA_AAUDIO_STREAM_STATE_CLOSING 11 #define MA_AAUDIO_STREAM_STATE_CLOSED 12 #define MA_AAUDIO_STREAM_STATE_DISCONNECTED 13 /* Performance modes. */ #define MA_AAUDIO_PERFORMANCE_MODE_NONE 10 #define MA_AAUDIO_PERFORMANCE_MODE_POWER_SAVING 11 #define MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY 12 /* Usage types. */ #define MA_AAUDIO_USAGE_MEDIA 1 #define MA_AAUDIO_USAGE_VOICE_COMMUNICATION 2 #define MA_AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING 3 #define MA_AAUDIO_USAGE_ALARM 4 #define MA_AAUDIO_USAGE_NOTIFICATION 5 #define MA_AAUDIO_USAGE_NOTIFICATION_RINGTONE 6 #define MA_AAUDIO_USAGE_NOTIFICATION_EVENT 10 #define MA_AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY 11 #define MA_AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE 12 #define MA_AAUDIO_USAGE_ASSISTANCE_SONIFICATION 13 #define MA_AAUDIO_USAGE_GAME 14 #define MA_AAUDIO_USAGE_ASSISTANT 16 #define MA_AAUDIO_SYSTEM_USAGE_EMERGENCY 1000 #define MA_AAUDIO_SYSTEM_USAGE_SAFETY 1001 #define MA_AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS 1002 #define MA_AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT 1003 /* Content types. */ #define MA_AAUDIO_CONTENT_TYPE_SPEECH 1 #define MA_AAUDIO_CONTENT_TYPE_MUSIC 2 #define MA_AAUDIO_CONTENT_TYPE_MOVIE 3 #define MA_AAUDIO_CONTENT_TYPE_SONIFICATION 4 /* Input presets. */ #define MA_AAUDIO_INPUT_PRESET_GENERIC 1 #define MA_AAUDIO_INPUT_PRESET_CAMCORDER 5 #define MA_AAUDIO_INPUT_PRESET_VOICE_RECOGNITION 6 #define MA_AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION 7 #define MA_AAUDIO_INPUT_PRESET_UNPROCESSED 9 #define MA_AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE 10 /* Allowed Capture Policies */ #define MA_AAUDIO_ALLOW_CAPTURE_BY_ALL 1 #define MA_AAUDIO_ALLOW_CAPTURE_BY_SYSTEM 2 #define MA_AAUDIO_ALLOW_CAPTURE_BY_NONE 3 /* Callback results. */ #define MA_AAUDIO_CALLBACK_RESULT_CONTINUE 0 #define MA_AAUDIO_CALLBACK_RESULT_STOP 1 typedef ma_aaudio_data_callback_result_t (* ma_AAudioStream_dataCallback) (ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t numFrames); typedef void (* ma_AAudioStream_errorCallback)(ma_AAudioStream *pStream, void *pUserData, ma_aaudio_result_t error); typedef ma_aaudio_result_t (* MA_PFN_AAudio_createStreamBuilder) (ma_AAudioStreamBuilder** ppBuilder); typedef ma_aaudio_result_t (* MA_PFN_AAudioStreamBuilder_delete) (ma_AAudioStreamBuilder* pBuilder); typedef void (* MA_PFN_AAudioStreamBuilder_setDeviceId) (ma_AAudioStreamBuilder* pBuilder, int32_t deviceId); typedef void (* MA_PFN_AAudioStreamBuilder_setDirection) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_direction_t direction); typedef void (* MA_PFN_AAudioStreamBuilder_setSharingMode) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_sharing_mode_t sharingMode); typedef void (* MA_PFN_AAudioStreamBuilder_setFormat) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_format_t format); typedef void (* MA_PFN_AAudioStreamBuilder_setChannelCount) (ma_AAudioStreamBuilder* pBuilder, int32_t channelCount); typedef void (* MA_PFN_AAudioStreamBuilder_setSampleRate) (ma_AAudioStreamBuilder* pBuilder, int32_t sampleRate); typedef void (* MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)(ma_AAudioStreamBuilder* pBuilder, int32_t numFrames); typedef void (* MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback) (ma_AAudioStreamBuilder* pBuilder, int32_t numFrames); typedef void (* MA_PFN_AAudioStreamBuilder_setDataCallback) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream_dataCallback callback, void* pUserData); typedef void (* MA_PFN_AAudioStreamBuilder_setErrorCallback) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream_errorCallback callback, void* pUserData); typedef void (* MA_PFN_AAudioStreamBuilder_setPerformanceMode) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_performance_mode_t mode); typedef void (* MA_PFN_AAudioStreamBuilder_setUsage) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_usage_t contentType); typedef void (* MA_PFN_AAudioStreamBuilder_setContentType) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_content_type_t contentType); typedef void (* MA_PFN_AAudioStreamBuilder_setInputPreset) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_input_preset_t inputPreset); typedef void (* MA_PFN_AAudioStreamBuilder_setAllowedCapturePolicy) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_allowed_capture_policy_t policy); typedef ma_aaudio_result_t (* MA_PFN_AAudioStreamBuilder_openStream) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream** ppStream); typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_close) (ma_AAudioStream* pStream); typedef ma_aaudio_stream_state_t (* MA_PFN_AAudioStream_getState) (ma_AAudioStream* pStream); typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_waitForStateChange) (ma_AAudioStream* pStream, ma_aaudio_stream_state_t inputState, ma_aaudio_stream_state_t* pNextState, int64_t timeoutInNanoseconds); typedef ma_aaudio_format_t (* MA_PFN_AAudioStream_getFormat) (ma_AAudioStream* pStream); typedef int32_t (* MA_PFN_AAudioStream_getChannelCount) (ma_AAudioStream* pStream); typedef int32_t (* MA_PFN_AAudioStream_getSampleRate) (ma_AAudioStream* pStream); typedef int32_t (* MA_PFN_AAudioStream_getBufferCapacityInFrames) (ma_AAudioStream* pStream); typedef int32_t (* MA_PFN_AAudioStream_getFramesPerDataCallback) (ma_AAudioStream* pStream); typedef int32_t (* MA_PFN_AAudioStream_getFramesPerBurst) (ma_AAudioStream* pStream); typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_requestStart) (ma_AAudioStream* pStream); typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_requestStop) (ma_AAudioStream* pStream); static ma_result ma_result_from_aaudio(ma_aaudio_result_t resultAA) { switch (resultAA) { case MA_AAUDIO_OK: return MA_SUCCESS; default: break; } return MA_ERROR; } static ma_aaudio_usage_t ma_to_usage__aaudio(ma_aaudio_usage usage) { switch (usage) { case ma_aaudio_usage_media: return MA_AAUDIO_USAGE_MEDIA; case ma_aaudio_usage_voice_communication: return MA_AAUDIO_USAGE_VOICE_COMMUNICATION; case ma_aaudio_usage_voice_communication_signalling: return MA_AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING; case ma_aaudio_usage_alarm: return MA_AAUDIO_USAGE_ALARM; case ma_aaudio_usage_notification: return MA_AAUDIO_USAGE_NOTIFICATION; case ma_aaudio_usage_notification_ringtone: return MA_AAUDIO_USAGE_NOTIFICATION_RINGTONE; case ma_aaudio_usage_notification_event: return MA_AAUDIO_USAGE_NOTIFICATION_EVENT; case ma_aaudio_usage_assistance_accessibility: return MA_AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY; case ma_aaudio_usage_assistance_navigation_guidance: return MA_AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE; case ma_aaudio_usage_assistance_sonification: return MA_AAUDIO_USAGE_ASSISTANCE_SONIFICATION; case ma_aaudio_usage_game: return MA_AAUDIO_USAGE_GAME; case ma_aaudio_usage_assitant: return MA_AAUDIO_USAGE_ASSISTANT; case ma_aaudio_usage_emergency: return MA_AAUDIO_SYSTEM_USAGE_EMERGENCY; case ma_aaudio_usage_safety: return MA_AAUDIO_SYSTEM_USAGE_SAFETY; case ma_aaudio_usage_vehicle_status: return MA_AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS; case ma_aaudio_usage_announcement: return MA_AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT; default: break; } return MA_AAUDIO_USAGE_MEDIA; } static ma_aaudio_content_type_t ma_to_content_type__aaudio(ma_aaudio_content_type contentType) { switch (contentType) { case ma_aaudio_content_type_speech: return MA_AAUDIO_CONTENT_TYPE_SPEECH; case ma_aaudio_content_type_music: return MA_AAUDIO_CONTENT_TYPE_MUSIC; case ma_aaudio_content_type_movie: return MA_AAUDIO_CONTENT_TYPE_MOVIE; case ma_aaudio_content_type_sonification: return MA_AAUDIO_CONTENT_TYPE_SONIFICATION; default: break; } return MA_AAUDIO_CONTENT_TYPE_SPEECH; } static ma_aaudio_input_preset_t ma_to_input_preset__aaudio(ma_aaudio_input_preset inputPreset) { switch (inputPreset) { case ma_aaudio_input_preset_generic: return MA_AAUDIO_INPUT_PRESET_GENERIC; case ma_aaudio_input_preset_camcorder: return MA_AAUDIO_INPUT_PRESET_CAMCORDER; case ma_aaudio_input_preset_voice_recognition: return MA_AAUDIO_INPUT_PRESET_VOICE_RECOGNITION; case ma_aaudio_input_preset_voice_communication: return MA_AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION; case ma_aaudio_input_preset_unprocessed: return MA_AAUDIO_INPUT_PRESET_UNPROCESSED; case ma_aaudio_input_preset_voice_performance: return MA_AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE; default: break; } return MA_AAUDIO_INPUT_PRESET_GENERIC; } static ma_aaudio_allowed_capture_policy_t ma_to_allowed_capture_policy__aaudio(ma_aaudio_allowed_capture_policy allowedCapturePolicy) { switch (allowedCapturePolicy) { case ma_aaudio_allow_capture_by_all: return MA_AAUDIO_ALLOW_CAPTURE_BY_ALL; case ma_aaudio_allow_capture_by_system: return MA_AAUDIO_ALLOW_CAPTURE_BY_SYSTEM; case ma_aaudio_allow_capture_by_none: return MA_AAUDIO_ALLOW_CAPTURE_BY_NONE; default: break; } return MA_AAUDIO_ALLOW_CAPTURE_BY_ALL; } static void ma_stream_error_callback__aaudio(ma_AAudioStream* pStream, void* pUserData, ma_aaudio_result_t error) { ma_result result; ma_job job; ma_device* pDevice = (ma_device*)pUserData; MA_ASSERT(pDevice != NULL); (void)error; ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[AAudio] ERROR CALLBACK: error=%d, AAudioStream_getState()=%d\n", error, ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream)); /* When we get an error, we'll assume that the stream is in an erroneous state and needs to be restarted. From the documentation, we cannot do this from the error callback. Therefore we are going to use an event thread for the AAudio backend to do this cleanly and safely. */ if (ma_atomic_bool32_get(&pDevice->aaudio.isTearingDown)) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[AAudio] Device Disconnected. Tearing down device.\n"); } else { job = ma_job_init(MA_JOB_TYPE_DEVICE_AAUDIO_REROUTE); job.data.device.aaudio.reroute.pDevice = pDevice; if (pStream == pDevice->aaudio.pStreamCapture) { job.data.device.aaudio.reroute.deviceType = ma_device_type_capture; } else { job.data.device.aaudio.reroute.deviceType = ma_device_type_playback; } result = ma_device_job_thread_post(&pDevice->pContext->aaudio.jobThread, &job); if (result != MA_SUCCESS) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[AAudio] Device Disconnected. Failed to post job for rerouting.\n"); return; } } } static ma_aaudio_data_callback_result_t ma_stream_data_callback_capture__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount) { ma_device* pDevice = (ma_device*)pUserData; MA_ASSERT(pDevice != NULL); if (frameCount > 0) { ma_device_handle_backend_data_callback(pDevice, NULL, pAudioData, (ma_uint32)frameCount); } (void)pStream; return MA_AAUDIO_CALLBACK_RESULT_CONTINUE; } static ma_aaudio_data_callback_result_t ma_stream_data_callback_playback__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount) { ma_device* pDevice = (ma_device*)pUserData; MA_ASSERT(pDevice != NULL); /* I've had a report that AAudio can sometimes post a frame count of 0. We need to check for that here so we don't get any errors at a deeper level. I'm doing the same with the capture side for safety, though I've not yet had any reports about that one. */ if (frameCount > 0) { ma_device_handle_backend_data_callback(pDevice, pAudioData, NULL, (ma_uint32)frameCount); } (void)pStream; return MA_AAUDIO_CALLBACK_RESULT_CONTINUE; } static ma_result ma_create_and_configure_AAudioStreamBuilder__aaudio(ma_context* pContext, const ma_device_id* pDeviceID, ma_device_type deviceType, ma_share_mode shareMode, const ma_device_descriptor* pDescriptor, const ma_device_config* pConfig, ma_device* pDevice, ma_AAudioStreamBuilder** ppBuilder) { ma_AAudioStreamBuilder* pBuilder; ma_aaudio_result_t resultAA; /* Safety. */ *ppBuilder = NULL; resultAA = ((MA_PFN_AAudio_createStreamBuilder)pContext->aaudio.AAudio_createStreamBuilder)(&pBuilder); if (resultAA != MA_AAUDIO_OK) { return ma_result_from_aaudio(resultAA); } if (pDeviceID != NULL) { ((MA_PFN_AAudioStreamBuilder_setDeviceId)pContext->aaudio.AAudioStreamBuilder_setDeviceId)(pBuilder, pDeviceID->aaudio); } ((MA_PFN_AAudioStreamBuilder_setDirection)pContext->aaudio.AAudioStreamBuilder_setDirection)(pBuilder, (deviceType == ma_device_type_playback) ? MA_AAUDIO_DIRECTION_OUTPUT : MA_AAUDIO_DIRECTION_INPUT); ((MA_PFN_AAudioStreamBuilder_setSharingMode)pContext->aaudio.AAudioStreamBuilder_setSharingMode)(pBuilder, (shareMode == ma_share_mode_shared) ? MA_AAUDIO_SHARING_MODE_SHARED : MA_AAUDIO_SHARING_MODE_EXCLUSIVE); /* If we have a device descriptor make sure we configure the stream builder to take our requested parameters. */ if (pDescriptor != NULL) { MA_ASSERT(pConfig != NULL); /* We must have a device config if we also have a descriptor. The config is required for AAudio specific configuration options. */ if (pDescriptor->sampleRate != 0) { ((MA_PFN_AAudioStreamBuilder_setSampleRate)pContext->aaudio.AAudioStreamBuilder_setSampleRate)(pBuilder, pDescriptor->sampleRate); } if (pDescriptor->channels != 0) { ((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pDescriptor->channels); } if (pDescriptor->format != ma_format_unknown) { ((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pDescriptor->format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT); } /* There have been reports where setting the frames per data callback results in an error. In particular, re-routing may inadvertently switch from low-latency mode, resulting in a less stable stream from the legacy path (AudioStreamLegacy). To address this, we simply don't set the value. It can still be set if it's explicitly requested via the aaudio.allowSetBufferCapacity variable in the device config. */ if ((!pConfig->aaudio.enableCompatibilityWorkarounds || ma_android_sdk_version() > 30) && pConfig->aaudio.allowSetBufferCapacity) { /* AAudio is annoying when it comes to its buffer calculation stuff because it doesn't let you retrieve the actual sample rate until after you've opened the stream. But you need to configure the buffer capacity before you open the stream... :/ To solve, we're just going to assume MA_DEFAULT_SAMPLE_RATE (48000) and move on. */ ma_uint32 bufferCapacityInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, pDescriptor->sampleRate, pConfig->performanceProfile) * pDescriptor->periodCount; ((MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames)(pBuilder, bufferCapacityInFrames); ((MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback)pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback)(pBuilder, bufferCapacityInFrames / pDescriptor->periodCount); } if (deviceType == ma_device_type_capture) { if (pConfig->aaudio.inputPreset != ma_aaudio_input_preset_default && pContext->aaudio.AAudioStreamBuilder_setInputPreset != NULL) { ((MA_PFN_AAudioStreamBuilder_setInputPreset)pContext->aaudio.AAudioStreamBuilder_setInputPreset)(pBuilder, ma_to_input_preset__aaudio(pConfig->aaudio.inputPreset)); } ((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_capture__aaudio, (void*)pDevice); } else { if (pConfig->aaudio.usage != ma_aaudio_usage_default && pContext->aaudio.AAudioStreamBuilder_setUsage != NULL) { ((MA_PFN_AAudioStreamBuilder_setUsage)pContext->aaudio.AAudioStreamBuilder_setUsage)(pBuilder, ma_to_usage__aaudio(pConfig->aaudio.usage)); } if (pConfig->aaudio.contentType != ma_aaudio_content_type_default && pContext->aaudio.AAudioStreamBuilder_setContentType != NULL) { ((MA_PFN_AAudioStreamBuilder_setContentType)pContext->aaudio.AAudioStreamBuilder_setContentType)(pBuilder, ma_to_content_type__aaudio(pConfig->aaudio.contentType)); } if (pConfig->aaudio.allowedCapturePolicy != ma_aaudio_allow_capture_default && pContext->aaudio.AAudioStreamBuilder_setAllowedCapturePolicy != NULL) { ((MA_PFN_AAudioStreamBuilder_setAllowedCapturePolicy)pContext->aaudio.AAudioStreamBuilder_setAllowedCapturePolicy)(pBuilder, ma_to_allowed_capture_policy__aaudio(pConfig->aaudio.allowedCapturePolicy)); } ((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_playback__aaudio, (void*)pDevice); } /* If we set AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, we allow for MMAP (non-legacy path). Since there's a mapping between miniaudio's performance profiles and AAudio's performance modes, let's use it. Beware though, with a conservative performance profile, AAudio will indeed take the legacy path. */ ((MA_PFN_AAudioStreamBuilder_setPerformanceMode)pContext->aaudio.AAudioStreamBuilder_setPerformanceMode)(pBuilder, (pConfig->performanceProfile == ma_performance_profile_low_latency) ? MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY : MA_AAUDIO_PERFORMANCE_MODE_NONE); /* We need to set an error callback to detect device changes. */ if (pDevice != NULL) { /* <-- pDevice should never be null if pDescriptor is not null, which is always the case if we hit this branch. Check anyway for safety. */ ((MA_PFN_AAudioStreamBuilder_setErrorCallback)pContext->aaudio.AAudioStreamBuilder_setErrorCallback)(pBuilder, ma_stream_error_callback__aaudio, (void*)pDevice); } } *ppBuilder = pBuilder; return MA_SUCCESS; } static ma_result ma_open_stream_and_close_builder__aaudio(ma_context* pContext, ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream** ppStream) { ma_result result; result = ma_result_from_aaudio(((MA_PFN_AAudioStreamBuilder_openStream)pContext->aaudio.AAudioStreamBuilder_openStream)(pBuilder, ppStream)); ((MA_PFN_AAudioStreamBuilder_delete)pContext->aaudio.AAudioStreamBuilder_delete)(pBuilder); return result; } static ma_result ma_open_stream_basic__aaudio(ma_context* pContext, const ma_device_id* pDeviceID, ma_device_type deviceType, ma_share_mode shareMode, ma_AAudioStream** ppStream) { ma_result result; ma_AAudioStreamBuilder* pBuilder; *ppStream = NULL; result = ma_create_and_configure_AAudioStreamBuilder__aaudio(pContext, pDeviceID, deviceType, shareMode, NULL, NULL, NULL, &pBuilder); if (result != MA_SUCCESS) { return result; } /* Let's give AAudio a hint to avoid the legacy path (AudioStreamLegacy). */ ((MA_PFN_AAudioStreamBuilder_setPerformanceMode)pContext->aaudio.AAudioStreamBuilder_setPerformanceMode)(pBuilder, MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY); return ma_open_stream_and_close_builder__aaudio(pContext, pBuilder, ppStream); } static ma_result ma_open_stream__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_type deviceType, const ma_device_descriptor* pDescriptor, ma_AAudioStream** ppStream) { ma_result result; ma_AAudioStreamBuilder* pBuilder; MA_ASSERT(pDevice != NULL); MA_ASSERT(pDescriptor != NULL); MA_ASSERT(deviceType != ma_device_type_duplex); /* This function should not be called for a full-duplex device type. */ *ppStream = NULL; result = ma_create_and_configure_AAudioStreamBuilder__aaudio(pDevice->pContext, pDescriptor->pDeviceID, deviceType, pDescriptor->shareMode, pDescriptor, pConfig, pDevice, &pBuilder); if (result != MA_SUCCESS) { return result; } return ma_open_stream_and_close_builder__aaudio(pDevice->pContext, pBuilder, ppStream); } static ma_result ma_close_stream__aaudio(ma_context* pContext, ma_AAudioStream* pStream) { if (pStream == NULL) { return MA_INVALID_ARGS; } return ma_result_from_aaudio(((MA_PFN_AAudioStream_close)pContext->aaudio.AAudioStream_close)(pStream)); } static ma_bool32 ma_has_default_device__aaudio(ma_context* pContext, ma_device_type deviceType) { /* The only way to know this is to try creating a stream. */ ma_AAudioStream* pStream; ma_result result = ma_open_stream_basic__aaudio(pContext, NULL, deviceType, ma_share_mode_shared, &pStream); if (result != MA_SUCCESS) { return MA_FALSE; } ma_close_stream__aaudio(pContext, pStream); return MA_TRUE; } static ma_result ma_wait_for_simple_state_transition__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_aaudio_stream_state_t oldState, ma_aaudio_stream_state_t newState) { ma_aaudio_stream_state_t actualNewState; ma_aaudio_result_t resultAA = ((MA_PFN_AAudioStream_waitForStateChange)pContext->aaudio.AAudioStream_waitForStateChange)(pStream, oldState, &actualNewState, 5000000000); /* 5 second timeout. */ if (resultAA != MA_AAUDIO_OK) { return ma_result_from_aaudio(resultAA); } if (newState != actualNewState) { return MA_ERROR; /* Failed to transition into the expected state. */ } return MA_SUCCESS; } static ma_result ma_context_enumerate_devices__aaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { ma_bool32 cbResult = MA_TRUE; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); /* Unfortunately AAudio does not have an enumeration API. Therefore I'm only going to report default devices, but only if it can instantiate a stream. */ /* Playback. */ if (cbResult) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); deviceInfo.id.aaudio = MA_AAUDIO_UNSPECIFIED; ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); if (ma_has_default_device__aaudio(pContext, ma_device_type_playback)) { cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } } /* Capture. */ if (cbResult) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); deviceInfo.id.aaudio = MA_AAUDIO_UNSPECIFIED; ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); if (ma_has_default_device__aaudio(pContext, ma_device_type_capture)) { cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); } } return MA_SUCCESS; } static void ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_format format, ma_uint32 flags, ma_device_info* pDeviceInfo) { MA_ASSERT(pContext != NULL); MA_ASSERT(pStream != NULL); MA_ASSERT(pDeviceInfo != NULL); pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)(pStream); pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)(pStream); pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = flags; pDeviceInfo->nativeDataFormatCount += 1; } static void ma_context_add_native_data_format_from_AAudioStream__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_uint32 flags, ma_device_info* pDeviceInfo) { /* AAudio supports s16 and f32. */ ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(pContext, pStream, ma_format_f32, flags, pDeviceInfo); ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(pContext, pStream, ma_format_s16, flags, pDeviceInfo); } static ma_result ma_context_get_device_info__aaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { ma_AAudioStream* pStream; ma_result result; MA_ASSERT(pContext != NULL); /* ID */ if (pDeviceID != NULL) { pDeviceInfo->id.aaudio = pDeviceID->aaudio; } else { pDeviceInfo->id.aaudio = MA_AAUDIO_UNSPECIFIED; } /* Name */ if (deviceType == ma_device_type_playback) { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); } else { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); } pDeviceInfo->nativeDataFormatCount = 0; /* We'll need to open the device to get accurate sample rate and channel count information. */ result = ma_open_stream_basic__aaudio(pContext, pDeviceID, deviceType, ma_share_mode_shared, &pStream); if (result != MA_SUCCESS) { return result; } ma_context_add_native_data_format_from_AAudioStream__aaudio(pContext, pStream, 0, pDeviceInfo); ma_close_stream__aaudio(pContext, pStream); pStream = NULL; return MA_SUCCESS; } static ma_result ma_close_streams__aaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); /* When rerouting, streams may have been closed and never re-opened. Hence the extra checks below. */ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); pDevice->aaudio.pStreamCapture = NULL; } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); pDevice->aaudio.pStreamPlayback = NULL; } return MA_SUCCESS; } static ma_result ma_device_uninit__aaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); /* Note: Closing the streams may cause a timeout error, which would then trigger rerouting in our error callback. We must not schedule a reroute when device is getting destroyed. */ ma_atomic_bool32_set(&pDevice->aaudio.isTearingDown, MA_TRUE); /* Wait for any rerouting to finish before attempting to close the streams. */ ma_mutex_lock(&pDevice->aaudio.rerouteLock); { ma_close_streams__aaudio(pDevice); } ma_mutex_unlock(&pDevice->aaudio.rerouteLock); /* Destroy rerouting lock. */ ma_mutex_uninit(&pDevice->aaudio.rerouteLock); return MA_SUCCESS; } static ma_result ma_device_init_by_type__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_type deviceType, ma_device_descriptor* pDescriptor, ma_AAudioStream** ppStream) { ma_result result; int32_t bufferCapacityInFrames; int32_t framesPerDataCallback; ma_AAudioStream* pStream; MA_ASSERT(pDevice != NULL); MA_ASSERT(pConfig != NULL); MA_ASSERT(pDescriptor != NULL); *ppStream = NULL; /* Safety. */ /* First step is to open the stream. From there we'll be able to extract the internal configuration. */ result = ma_open_stream__aaudio(pDevice, pConfig, deviceType, pDescriptor, &pStream); if (result != MA_SUCCESS) { return result; /* Failed to open the AAudio stream. */ } /* Now extract the internal configuration. */ pDescriptor->format = (((MA_PFN_AAudioStream_getFormat)pDevice->pContext->aaudio.AAudioStream_getFormat)(pStream) == MA_AAUDIO_FORMAT_PCM_I16) ? ma_format_s16 : ma_format_f32; pDescriptor->channels = ((MA_PFN_AAudioStream_getChannelCount)pDevice->pContext->aaudio.AAudioStream_getChannelCount)(pStream); pDescriptor->sampleRate = ((MA_PFN_AAudioStream_getSampleRate)pDevice->pContext->aaudio.AAudioStream_getSampleRate)(pStream); /* For the channel map we need to be sure we don't overflow any buffers. */ if (pDescriptor->channels <= MA_MAX_CHANNELS) { ma_channel_map_init_standard(ma_standard_channel_map_default, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), pDescriptor->channels); /* <-- Cannot find info on channel order, so assuming a default. */ } else { ma_channel_map_init_blank(pDescriptor->channelMap, MA_MAX_CHANNELS); /* Too many channels. Use a blank channel map. */ } bufferCapacityInFrames = ((MA_PFN_AAudioStream_getBufferCapacityInFrames)pDevice->pContext->aaudio.AAudioStream_getBufferCapacityInFrames)(pStream); framesPerDataCallback = ((MA_PFN_AAudioStream_getFramesPerDataCallback)pDevice->pContext->aaudio.AAudioStream_getFramesPerDataCallback)(pStream); if (framesPerDataCallback > 0) { pDescriptor->periodSizeInFrames = framesPerDataCallback; pDescriptor->periodCount = bufferCapacityInFrames / framesPerDataCallback; } else { pDescriptor->periodSizeInFrames = bufferCapacityInFrames; pDescriptor->periodCount = 1; } *ppStream = pStream; return MA_SUCCESS; } static ma_result ma_device_init_streams__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { ma_result result; MA_ASSERT(pDevice != NULL); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } pDevice->aaudio.usage = pConfig->aaudio.usage; pDevice->aaudio.contentType = pConfig->aaudio.contentType; pDevice->aaudio.inputPreset = pConfig->aaudio.inputPreset; pDevice->aaudio.allowedCapturePolicy = pConfig->aaudio.allowedCapturePolicy; pDevice->aaudio.noAutoStartAfterReroute = pConfig->aaudio.noAutoStartAfterReroute; if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { result = ma_device_init_by_type__aaudio(pDevice, pConfig, ma_device_type_capture, pDescriptorCapture, (ma_AAudioStream**)&pDevice->aaudio.pStreamCapture); if (result != MA_SUCCESS) { return result; } } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { result = ma_device_init_by_type__aaudio(pDevice, pConfig, ma_device_type_playback, pDescriptorPlayback, (ma_AAudioStream**)&pDevice->aaudio.pStreamPlayback); if (result != MA_SUCCESS) { return result; } } return MA_SUCCESS; } static ma_result ma_device_init__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { ma_result result; MA_ASSERT(pDevice != NULL); result = ma_device_init_streams__aaudio(pDevice, pConfig, pDescriptorPlayback, pDescriptorCapture); if (result != MA_SUCCESS) { return result; } result = ma_mutex_init(&pDevice->aaudio.rerouteLock); if (result != MA_SUCCESS) { return result; } return MA_SUCCESS; } static ma_result ma_device_start_stream__aaudio(ma_device* pDevice, ma_AAudioStream* pStream) { ma_aaudio_result_t resultAA; ma_aaudio_stream_state_t currentState; MA_ASSERT(pDevice != NULL); if (pStream == NULL) { return MA_INVALID_ARGS; } resultAA = ((MA_PFN_AAudioStream_requestStart)pDevice->pContext->aaudio.AAudioStream_requestStart)(pStream); if (resultAA != MA_AAUDIO_OK) { return ma_result_from_aaudio(resultAA); } /* Do we actually need to wait for the device to transition into its started state? */ /* The device should be in either a starting or started state. If it's not set to started we need to wait for it to transition. It should go from starting to started. */ currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream); if (currentState != MA_AAUDIO_STREAM_STATE_STARTED) { ma_result result; if (currentState != MA_AAUDIO_STREAM_STATE_STARTING) { return MA_ERROR; /* Expecting the stream to be a starting or started state. */ } result = ma_wait_for_simple_state_transition__aaudio(pDevice->pContext, pStream, currentState, MA_AAUDIO_STREAM_STATE_STARTED); if (result != MA_SUCCESS) { return result; } } return MA_SUCCESS; } static ma_result ma_device_stop_stream__aaudio(ma_device* pDevice, ma_AAudioStream* pStream) { ma_aaudio_result_t resultAA; ma_aaudio_stream_state_t currentState; MA_ASSERT(pDevice != NULL); if (pStream == NULL) { return MA_INVALID_ARGS; } /* From the AAudio documentation: The stream will stop after all of the data currently buffered has been played. This maps with miniaudio's requirement that device's be drained which means we don't need to implement any draining logic. */ currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream); if (currentState == MA_AAUDIO_STREAM_STATE_DISCONNECTED) { return MA_SUCCESS; /* The device is disconnected. Don't try stopping it. */ } resultAA = ((MA_PFN_AAudioStream_requestStop)pDevice->pContext->aaudio.AAudioStream_requestStop)(pStream); if (resultAA != MA_AAUDIO_OK) { return ma_result_from_aaudio(resultAA); } /* The device should be in either a stopping or stopped state. If it's not set to started we need to wait for it to transition. It should go from stopping to stopped. */ currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream); if (currentState != MA_AAUDIO_STREAM_STATE_STOPPED) { ma_result result; if (currentState != MA_AAUDIO_STREAM_STATE_STOPPING) { return MA_ERROR; /* Expecting the stream to be a stopping or stopped state. */ } result = ma_wait_for_simple_state_transition__aaudio(pDevice->pContext, pStream, currentState, MA_AAUDIO_STREAM_STATE_STOPPED); if (result != MA_SUCCESS) { return result; } } return MA_SUCCESS; } static ma_result ma_device_start__aaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ma_result result = ma_device_start_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); if (result != MA_SUCCESS) { return result; } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ma_result result = ma_device_start_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); if (result != MA_SUCCESS) { if (pDevice->type == ma_device_type_duplex) { ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); } return result; } } return MA_SUCCESS; } static ma_result ma_device_stop__aaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ma_result result = ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); if (result != MA_SUCCESS) { return result; } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ma_result result = ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); if (result != MA_SUCCESS) { return result; } } ma_device__on_notification_stopped(pDevice); return MA_SUCCESS; } static ma_result ma_device_reinit__aaudio(ma_device* pDevice, ma_device_type deviceType) { const ma_int32 maxAttempts = 4; /* Reasonable retry limit. */ ma_result result; ma_int32 iAttempt; MA_ASSERT(pDevice != NULL); /* We got disconnected! Retry a few times, until we find a connected device! */ iAttempt = 0; while (iAttempt++ < maxAttempts) { /* Device tearing down? No need to reroute! */ if (ma_atomic_bool32_get(&pDevice->aaudio.isTearingDown)) { result = MA_SUCCESS; /* Caller should continue as normal. */ break; } /* The first thing to do is close the streams. */ ma_close_streams__aaudio(pDevice); /* Now we need to reinitialize each streams. The hardest part with this is just filling output the config and descriptors. */ ma_device_config deviceConfig; ma_device_descriptor descriptorPlayback; ma_device_descriptor descriptorCapture; deviceConfig = ma_device_config_init(deviceType); deviceConfig.playback.pDeviceID = NULL; /* Only doing rerouting with default devices. */ deviceConfig.playback.shareMode = pDevice->playback.shareMode; deviceConfig.playback.format = pDevice->playback.format; deviceConfig.playback.channels = pDevice->playback.channels; deviceConfig.capture.pDeviceID = NULL; /* Only doing rerouting with default devices. */ deviceConfig.capture.shareMode = pDevice->capture.shareMode; deviceConfig.capture.format = pDevice->capture.format; deviceConfig.capture.channels = pDevice->capture.channels; deviceConfig.sampleRate = pDevice->sampleRate; deviceConfig.aaudio.usage = pDevice->aaudio.usage; deviceConfig.aaudio.contentType = pDevice->aaudio.contentType; deviceConfig.aaudio.inputPreset = pDevice->aaudio.inputPreset; deviceConfig.aaudio.allowedCapturePolicy = pDevice->aaudio.allowedCapturePolicy; deviceConfig.aaudio.noAutoStartAfterReroute = pDevice->aaudio.noAutoStartAfterReroute; deviceConfig.periods = 1; /* Try to get an accurate period size. */ if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) { deviceConfig.periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames; } else { deviceConfig.periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames; } if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) { descriptorCapture.pDeviceID = deviceConfig.capture.pDeviceID; descriptorCapture.shareMode = deviceConfig.capture.shareMode; descriptorCapture.format = deviceConfig.capture.format; descriptorCapture.channels = deviceConfig.capture.channels; descriptorCapture.sampleRate = deviceConfig.sampleRate; descriptorCapture.periodSizeInFrames = deviceConfig.periodSizeInFrames; descriptorCapture.periodCount = deviceConfig.periods; } if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) { descriptorPlayback.pDeviceID = deviceConfig.playback.pDeviceID; descriptorPlayback.shareMode = deviceConfig.playback.shareMode; descriptorPlayback.format = deviceConfig.playback.format; descriptorPlayback.channels = deviceConfig.playback.channels; descriptorPlayback.sampleRate = deviceConfig.sampleRate; descriptorPlayback.periodSizeInFrames = deviceConfig.periodSizeInFrames; descriptorPlayback.periodCount = deviceConfig.periods; } result = ma_device_init_streams__aaudio(pDevice, &deviceConfig, &descriptorPlayback, &descriptorCapture); if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[AAudio] Failed to create stream after route change."); /* Reroute failed! */ break; } result = ma_device_post_init(pDevice, deviceType, &descriptorPlayback, &descriptorCapture); if (result != MA_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[AAudio] Failed to initialize device after route change."); ma_close_streams__aaudio(pDevice); /* Reroute failed! */ break; } /* We'll only ever do this in response to a reroute. */ ma_device__on_notification_rerouted(pDevice); /* If the device is started, start the streams. Maybe make this configurable? */ if (ma_device_get_state(pDevice) == ma_device_state_started) { if (pDevice->aaudio.noAutoStartAfterReroute == MA_FALSE) { result = ma_device_start__aaudio(pDevice); if (result != MA_SUCCESS) { if (iAttempt < maxAttempts) { ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[AAudio] Failed to start stream after route change, retrying(%d)", iAttempt); } else { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[AAudio] Failed to start stream after route change, giving up."); } } } else { ma_device_stop(pDevice); /* Do a full device stop so we set internal state correctly. */ } } if (result == MA_SUCCESS) { /* Reroute successful! */ break; } } return result; } static ma_result ma_device_get_info__aaudio(ma_device* pDevice, ma_device_type type, ma_device_info* pDeviceInfo) { ma_AAudioStream* pStream = NULL; MA_ASSERT(pDevice != NULL); MA_ASSERT(type != ma_device_type_duplex); MA_ASSERT(pDeviceInfo != NULL); if (type == ma_device_type_capture) { pStream = (ma_AAudioStream*)pDevice->aaudio.pStreamCapture; pDeviceInfo->id.aaudio = pDevice->capture.id.aaudio; ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); /* Only supporting default devices. */ } if (type == ma_device_type_playback) { pStream = (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback; pDeviceInfo->id.aaudio = pDevice->playback.id.aaudio; ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); /* Only supporting default devices. */ } /* Safety. Should never happen. */ if (pStream == NULL) { return MA_INVALID_OPERATION; } pDeviceInfo->nativeDataFormatCount = 0; ma_context_add_native_data_format_from_AAudioStream__aaudio(pDevice->pContext, pStream, 0, pDeviceInfo); return MA_SUCCESS; } static ma_result ma_context_uninit__aaudio(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_aaudio); ma_device_job_thread_uninit(&pContext->aaudio.jobThread, &pContext->allocationCallbacks); ma_dlclose(ma_context_get_log(pContext), pContext->aaudio.hAAudio); pContext->aaudio.hAAudio = NULL; return MA_SUCCESS; } static ma_result ma_context_init__aaudio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { #if !defined(MA_NO_RUNTIME_LINKING) size_t i; const char* libNames[] = { "libaaudio.so" }; for (i = 0; i < ma_countof(libNames); ++i) { pContext->aaudio.hAAudio = ma_dlopen(ma_context_get_log(pContext), libNames[i]); if (pContext->aaudio.hAAudio != NULL) { break; } } if (pContext->aaudio.hAAudio == NULL) { return MA_FAILED_TO_INIT_BACKEND; } pContext->aaudio.AAudio_createStreamBuilder = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudio_createStreamBuilder"); pContext->aaudio.AAudioStreamBuilder_delete = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_delete"); pContext->aaudio.AAudioStreamBuilder_setDeviceId = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDeviceId"); pContext->aaudio.AAudioStreamBuilder_setDirection = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDirection"); pContext->aaudio.AAudioStreamBuilder_setSharingMode = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setSharingMode"); pContext->aaudio.AAudioStreamBuilder_setFormat = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setFormat"); pContext->aaudio.AAudioStreamBuilder_setChannelCount = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setChannelCount"); pContext->aaudio.AAudioStreamBuilder_setSampleRate = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setSampleRate"); pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setBufferCapacityInFrames"); pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setFramesPerDataCallback"); pContext->aaudio.AAudioStreamBuilder_setDataCallback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDataCallback"); pContext->aaudio.AAudioStreamBuilder_setErrorCallback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setErrorCallback"); pContext->aaudio.AAudioStreamBuilder_setPerformanceMode = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setPerformanceMode"); pContext->aaudio.AAudioStreamBuilder_setUsage = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setUsage"); pContext->aaudio.AAudioStreamBuilder_setContentType = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setContentType"); pContext->aaudio.AAudioStreamBuilder_setInputPreset = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setInputPreset"); pContext->aaudio.AAudioStreamBuilder_setAllowedCapturePolicy = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_setAllowedCapturePolicy"); pContext->aaudio.AAudioStreamBuilder_openStream = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStreamBuilder_openStream"); pContext->aaudio.AAudioStream_close = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStream_close"); pContext->aaudio.AAudioStream_getState = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStream_getState"); pContext->aaudio.AAudioStream_waitForStateChange = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStream_waitForStateChange"); pContext->aaudio.AAudioStream_getFormat = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStream_getFormat"); pContext->aaudio.AAudioStream_getChannelCount = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStream_getChannelCount"); pContext->aaudio.AAudioStream_getSampleRate = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStream_getSampleRate"); pContext->aaudio.AAudioStream_getBufferCapacityInFrames = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStream_getBufferCapacityInFrames"); pContext->aaudio.AAudioStream_getFramesPerDataCallback = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStream_getFramesPerDataCallback"); pContext->aaudio.AAudioStream_getFramesPerBurst = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStream_getFramesPerBurst"); pContext->aaudio.AAudioStream_requestStart = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStream_requestStart"); pContext->aaudio.AAudioStream_requestStop = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->aaudio.hAAudio, "AAudioStream_requestStop"); #else pContext->aaudio.AAudio_createStreamBuilder = (ma_proc)AAudio_createStreamBuilder; pContext->aaudio.AAudioStreamBuilder_delete = (ma_proc)AAudioStreamBuilder_delete; pContext->aaudio.AAudioStreamBuilder_setDeviceId = (ma_proc)AAudioStreamBuilder_setDeviceId; pContext->aaudio.AAudioStreamBuilder_setDirection = (ma_proc)AAudioStreamBuilder_setDirection; pContext->aaudio.AAudioStreamBuilder_setSharingMode = (ma_proc)AAudioStreamBuilder_setSharingMode; pContext->aaudio.AAudioStreamBuilder_setFormat = (ma_proc)AAudioStreamBuilder_setFormat; pContext->aaudio.AAudioStreamBuilder_setChannelCount = (ma_proc)AAudioStreamBuilder_setChannelCount; pContext->aaudio.AAudioStreamBuilder_setSampleRate = (ma_proc)AAudioStreamBuilder_setSampleRate; pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames = (ma_proc)AAudioStreamBuilder_setBufferCapacityInFrames; pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback = (ma_proc)AAudioStreamBuilder_setFramesPerDataCallback; pContext->aaudio.AAudioStreamBuilder_setDataCallback = (ma_proc)AAudioStreamBuilder_setDataCallback; pContext->aaudio.AAudioStreamBuilder_setErrorCallback = (ma_proc)AAudioStreamBuilder_setErrorCallback; pContext->aaudio.AAudioStreamBuilder_setPerformanceMode = (ma_proc)AAudioStreamBuilder_setPerformanceMode; pContext->aaudio.AAudioStreamBuilder_setUsage = (ma_proc)AAudioStreamBuilder_setUsage; pContext->aaudio.AAudioStreamBuilder_setContentType = (ma_proc)AAudioStreamBuilder_setContentType; pContext->aaudio.AAudioStreamBuilder_setInputPreset = (ma_proc)AAudioStreamBuilder_setInputPreset; #if defined(__ANDROID_API__) && __ANDROID_API__ >= 29 pContext->aaudio.AAudioStreamBuilder_setAllowedCapturePolicy = (ma_proc)AAudioStreamBuilder_setAllowedCapturePolicy; #endif pContext->aaudio.AAudioStreamBuilder_openStream = (ma_proc)AAudioStreamBuilder_openStream; pContext->aaudio.AAudioStream_close = (ma_proc)AAudioStream_close; pContext->aaudio.AAudioStream_getState = (ma_proc)AAudioStream_getState; pContext->aaudio.AAudioStream_waitForStateChange = (ma_proc)AAudioStream_waitForStateChange; pContext->aaudio.AAudioStream_getFormat = (ma_proc)AAudioStream_getFormat; pContext->aaudio.AAudioStream_getChannelCount = (ma_proc)AAudioStream_getChannelCount; pContext->aaudio.AAudioStream_getSampleRate = (ma_proc)AAudioStream_getSampleRate; pContext->aaudio.AAudioStream_getBufferCapacityInFrames = (ma_proc)AAudioStream_getBufferCapacityInFrames; pContext->aaudio.AAudioStream_getFramesPerDataCallback = (ma_proc)AAudioStream_getFramesPerDataCallback; pContext->aaudio.AAudioStream_getFramesPerBurst = (ma_proc)AAudioStream_getFramesPerBurst; pContext->aaudio.AAudioStream_requestStart = (ma_proc)AAudioStream_requestStart; pContext->aaudio.AAudioStream_requestStop = (ma_proc)AAudioStream_requestStop; #endif pCallbacks->onContextInit = ma_context_init__aaudio; pCallbacks->onContextUninit = ma_context_uninit__aaudio; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__aaudio; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__aaudio; pCallbacks->onDeviceInit = ma_device_init__aaudio; pCallbacks->onDeviceUninit = ma_device_uninit__aaudio; pCallbacks->onDeviceStart = ma_device_start__aaudio; pCallbacks->onDeviceStop = ma_device_stop__aaudio; pCallbacks->onDeviceRead = NULL; /* Not used because AAudio is asynchronous. */ pCallbacks->onDeviceWrite = NULL; /* Not used because AAudio is asynchronous. */ pCallbacks->onDeviceDataLoop = NULL; /* Not used because AAudio is asynchronous. */ pCallbacks->onDeviceGetInfo = ma_device_get_info__aaudio; /* We need a job thread so we can deal with rerouting. */ { ma_result result; ma_device_job_thread_config jobThreadConfig; jobThreadConfig = ma_device_job_thread_config_init(); result = ma_device_job_thread_init(&jobThreadConfig, &pContext->allocationCallbacks, &pContext->aaudio.jobThread); if (result != MA_SUCCESS) { ma_dlclose(ma_context_get_log(pContext), pContext->aaudio.hAAudio); pContext->aaudio.hAAudio = NULL; return result; } } (void)pConfig; return MA_SUCCESS; } static ma_result ma_job_process__device__aaudio_reroute(ma_job* pJob) { ma_result result = MA_SUCCESS; ma_device* pDevice; MA_ASSERT(pJob != NULL); pDevice = (ma_device*)pJob->data.device.aaudio.reroute.pDevice; MA_ASSERT(pDevice != NULL); ma_mutex_lock(&pDevice->aaudio.rerouteLock); { /* Here is where we need to reroute the device. To do this we need to uninitialize the stream and reinitialize it. */ result = ma_device_reinit__aaudio(pDevice, (ma_device_type)pJob->data.device.aaudio.reroute.deviceType); if (result != MA_SUCCESS) { /* Getting here means we failed to reroute the device. The best thing I can think of here is to just stop the device. */ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[AAudio] Stopping device due to reroute failure."); ma_device_stop(pDevice); } } ma_mutex_unlock(&pDevice->aaudio.rerouteLock); return result; } #else /* Getting here means there is no AAudio backend so we need a no-op job implementation. */ static ma_result ma_job_process__device__aaudio_reroute(ma_job* pJob) { return ma_job_process__noop(pJob); } #endif /* AAudio */ /****************************************************************************** OpenSL|ES Backend ******************************************************************************/ #ifdef MA_HAS_OPENSL #include <SLES/OpenSLES.h> #ifdef MA_ANDROID #include <SLES/OpenSLES_Android.h> #endif typedef SLresult (SLAPIENTRY * ma_slCreateEngine_proc)(SLObjectItf* pEngine, SLuint32 numOptions, SLEngineOption* pEngineOptions, SLuint32 numInterfaces, SLInterfaceID* pInterfaceIds, SLboolean* pInterfaceRequired); /* OpenSL|ES has one-per-application objects :( */ static SLObjectItf g_maEngineObjectSL = NULL; static SLEngineItf g_maEngineSL = NULL; static ma_uint32 g_maOpenSLInitCounter = 0; static ma_spinlock g_maOpenSLSpinlock = 0; /* For init/uninit. */ #define MA_OPENSL_OBJ(p) (*((SLObjectItf)(p))) #define MA_OPENSL_OUTPUTMIX(p) (*((SLOutputMixItf)(p))) #define MA_OPENSL_PLAY(p) (*((SLPlayItf)(p))) #define MA_OPENSL_RECORD(p) (*((SLRecordItf)(p))) #ifdef MA_ANDROID #define MA_OPENSL_BUFFERQUEUE(p) (*((SLAndroidSimpleBufferQueueItf)(p))) #else #define MA_OPENSL_BUFFERQUEUE(p) (*((SLBufferQueueItf)(p))) #endif static ma_result ma_result_from_OpenSL(SLuint32 result) { switch (result) { case SL_RESULT_SUCCESS: return MA_SUCCESS; case SL_RESULT_PRECONDITIONS_VIOLATED: return MA_ERROR; case SL_RESULT_PARAMETER_INVALID: return MA_INVALID_ARGS; case SL_RESULT_MEMORY_FAILURE: return MA_OUT_OF_MEMORY; case SL_RESULT_RESOURCE_ERROR: return MA_INVALID_DATA; case SL_RESULT_RESOURCE_LOST: return MA_ERROR; case SL_RESULT_IO_ERROR: return MA_IO_ERROR; case SL_RESULT_BUFFER_INSUFFICIENT: return MA_NO_SPACE; case SL_RESULT_CONTENT_CORRUPTED: return MA_INVALID_DATA; case SL_RESULT_CONTENT_UNSUPPORTED: return MA_FORMAT_NOT_SUPPORTED; case SL_RESULT_CONTENT_NOT_FOUND: return MA_ERROR; case SL_RESULT_PERMISSION_DENIED: return MA_ACCESS_DENIED; case SL_RESULT_FEATURE_UNSUPPORTED: return MA_NOT_IMPLEMENTED; case SL_RESULT_INTERNAL_ERROR: return MA_ERROR; case SL_RESULT_UNKNOWN_ERROR: return MA_ERROR; case SL_RESULT_OPERATION_ABORTED: return MA_ERROR; case SL_RESULT_CONTROL_LOST: return MA_ERROR; default: return MA_ERROR; } } /* Converts an individual OpenSL-style channel identifier (SL_SPEAKER_FRONT_LEFT, etc.) to miniaudio. */ static ma_uint8 ma_channel_id_to_ma__opensl(SLuint32 id) { switch (id) { case SL_SPEAKER_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT; case SL_SPEAKER_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT; case SL_SPEAKER_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER; case SL_SPEAKER_LOW_FREQUENCY: return MA_CHANNEL_LFE; case SL_SPEAKER_BACK_LEFT: return MA_CHANNEL_BACK_LEFT; case SL_SPEAKER_BACK_RIGHT: return MA_CHANNEL_BACK_RIGHT; case SL_SPEAKER_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER; case SL_SPEAKER_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER; case SL_SPEAKER_BACK_CENTER: return MA_CHANNEL_BACK_CENTER; case SL_SPEAKER_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT; case SL_SPEAKER_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT; case SL_SPEAKER_TOP_CENTER: return MA_CHANNEL_TOP_CENTER; case SL_SPEAKER_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT; case SL_SPEAKER_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER; case SL_SPEAKER_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT; case SL_SPEAKER_TOP_BACK_LEFT: return MA_CHANNEL_TOP_BACK_LEFT; case SL_SPEAKER_TOP_BACK_CENTER: return MA_CHANNEL_TOP_BACK_CENTER; case SL_SPEAKER_TOP_BACK_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT; default: return 0; } } /* Converts an individual miniaudio channel identifier (MA_CHANNEL_FRONT_LEFT, etc.) to OpenSL-style. */ static SLuint32 ma_channel_id_to_opensl(ma_uint8 id) { switch (id) { case MA_CHANNEL_MONO: return SL_SPEAKER_FRONT_CENTER; case MA_CHANNEL_FRONT_LEFT: return SL_SPEAKER_FRONT_LEFT; case MA_CHANNEL_FRONT_RIGHT: return SL_SPEAKER_FRONT_RIGHT; case MA_CHANNEL_FRONT_CENTER: return SL_SPEAKER_FRONT_CENTER; case MA_CHANNEL_LFE: return SL_SPEAKER_LOW_FREQUENCY; case MA_CHANNEL_BACK_LEFT: return SL_SPEAKER_BACK_LEFT; case MA_CHANNEL_BACK_RIGHT: return SL_SPEAKER_BACK_RIGHT; case MA_CHANNEL_FRONT_LEFT_CENTER: return SL_SPEAKER_FRONT_LEFT_OF_CENTER; case MA_CHANNEL_FRONT_RIGHT_CENTER: return SL_SPEAKER_FRONT_RIGHT_OF_CENTER; case MA_CHANNEL_BACK_CENTER: return SL_SPEAKER_BACK_CENTER; case MA_CHANNEL_SIDE_LEFT: return SL_SPEAKER_SIDE_LEFT; case MA_CHANNEL_SIDE_RIGHT: return SL_SPEAKER_SIDE_RIGHT; case MA_CHANNEL_TOP_CENTER: return SL_SPEAKER_TOP_CENTER; case MA_CHANNEL_TOP_FRONT_LEFT: return SL_SPEAKER_TOP_FRONT_LEFT; case MA_CHANNEL_TOP_FRONT_CENTER: return SL_SPEAKER_TOP_FRONT_CENTER; case MA_CHANNEL_TOP_FRONT_RIGHT: return SL_SPEAKER_TOP_FRONT_RIGHT; case MA_CHANNEL_TOP_BACK_LEFT: return SL_SPEAKER_TOP_BACK_LEFT; case MA_CHANNEL_TOP_BACK_CENTER: return SL_SPEAKER_TOP_BACK_CENTER; case MA_CHANNEL_TOP_BACK_RIGHT: return SL_SPEAKER_TOP_BACK_RIGHT; default: return 0; } } /* Converts a channel mapping to an OpenSL-style channel mask. */ static SLuint32 ma_channel_map_to_channel_mask__opensl(const ma_channel* pChannelMap, ma_uint32 channels) { SLuint32 channelMask = 0; ma_uint32 iChannel; for (iChannel = 0; iChannel < channels; ++iChannel) { channelMask |= ma_channel_id_to_opensl(pChannelMap[iChannel]); } return channelMask; } /* Converts an OpenSL-style channel mask to a miniaudio channel map. */ static void ma_channel_mask_to_channel_map__opensl(SLuint32 channelMask, ma_uint32 channels, ma_channel* pChannelMap) { if (channels == 1 && channelMask == 0) { pChannelMap[0] = MA_CHANNEL_MONO; } else if (channels == 2 && channelMask == 0) { pChannelMap[0] = MA_CHANNEL_FRONT_LEFT; pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT; } else { if (channels == 1 && (channelMask & SL_SPEAKER_FRONT_CENTER) != 0) { pChannelMap[0] = MA_CHANNEL_MONO; } else { /* Just iterate over each bit. */ ma_uint32 iChannel = 0; ma_uint32 iBit; for (iBit = 0; iBit < 32 && iChannel < channels; ++iBit) { SLuint32 bitValue = (channelMask & (1UL << iBit)); if (bitValue != 0) { /* The bit is set. */ pChannelMap[iChannel] = ma_channel_id_to_ma__opensl(bitValue); iChannel += 1; } } } } } static SLuint32 ma_round_to_standard_sample_rate__opensl(SLuint32 samplesPerSec) { if (samplesPerSec <= SL_SAMPLINGRATE_8) { return SL_SAMPLINGRATE_8; } if (samplesPerSec <= SL_SAMPLINGRATE_11_025) { return SL_SAMPLINGRATE_11_025; } if (samplesPerSec <= SL_SAMPLINGRATE_12) { return SL_SAMPLINGRATE_12; } if (samplesPerSec <= SL_SAMPLINGRATE_16) { return SL_SAMPLINGRATE_16; } if (samplesPerSec <= SL_SAMPLINGRATE_22_05) { return SL_SAMPLINGRATE_22_05; } if (samplesPerSec <= SL_SAMPLINGRATE_24) { return SL_SAMPLINGRATE_24; } if (samplesPerSec <= SL_SAMPLINGRATE_32) { return SL_SAMPLINGRATE_32; } if (samplesPerSec <= SL_SAMPLINGRATE_44_1) { return SL_SAMPLINGRATE_44_1; } if (samplesPerSec <= SL_SAMPLINGRATE_48) { return SL_SAMPLINGRATE_48; } /* Android doesn't support more than 48000. */ #ifndef MA_ANDROID if (samplesPerSec <= SL_SAMPLINGRATE_64) { return SL_SAMPLINGRATE_64; } if (samplesPerSec <= SL_SAMPLINGRATE_88_2) { return SL_SAMPLINGRATE_88_2; } if (samplesPerSec <= SL_SAMPLINGRATE_96) { return SL_SAMPLINGRATE_96; } if (samplesPerSec <= SL_SAMPLINGRATE_192) { return SL_SAMPLINGRATE_192; } #endif return SL_SAMPLINGRATE_16; } static SLint32 ma_to_stream_type__opensl(ma_opensl_stream_type streamType) { switch (streamType) { case ma_opensl_stream_type_voice: return SL_ANDROID_STREAM_VOICE; case ma_opensl_stream_type_system: return SL_ANDROID_STREAM_SYSTEM; case ma_opensl_stream_type_ring: return SL_ANDROID_STREAM_RING; case ma_opensl_stream_type_media: return SL_ANDROID_STREAM_MEDIA; case ma_opensl_stream_type_alarm: return SL_ANDROID_STREAM_ALARM; case ma_opensl_stream_type_notification: return SL_ANDROID_STREAM_NOTIFICATION; default: break; } return SL_ANDROID_STREAM_VOICE; } static SLint32 ma_to_recording_preset__opensl(ma_opensl_recording_preset recordingPreset) { switch (recordingPreset) { case ma_opensl_recording_preset_generic: return SL_ANDROID_RECORDING_PRESET_GENERIC; case ma_opensl_recording_preset_camcorder: return SL_ANDROID_RECORDING_PRESET_CAMCORDER; case ma_opensl_recording_preset_voice_recognition: return SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION; case ma_opensl_recording_preset_voice_communication: return SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION; case ma_opensl_recording_preset_voice_unprocessed: return SL_ANDROID_RECORDING_PRESET_UNPROCESSED; default: break; } return SL_ANDROID_RECORDING_PRESET_NONE; } static ma_result ma_context_enumerate_devices__opensl(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { ma_bool32 cbResult; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to enumerate devices. */ if (g_maOpenSLInitCounter == 0) { return MA_INVALID_OPERATION; } /* TODO: Test Me. This is currently untested, so for now we are just returning default devices. */ #if 0 && !defined(MA_ANDROID) ma_bool32 isTerminated = MA_FALSE; SLuint32 pDeviceIDs[128]; SLint32 deviceCount = sizeof(pDeviceIDs) / sizeof(pDeviceIDs[0]); SLAudioIODeviceCapabilitiesItf deviceCaps; SLresult resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, (SLInterfaceID)pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES, &deviceCaps); if (resultSL != SL_RESULT_SUCCESS) { /* The interface may not be supported so just report a default device. */ goto return_default_device; } /* Playback */ if (!isTerminated) { resultSL = (*deviceCaps)->GetAvailableAudioOutputs(deviceCaps, &deviceCount, pDeviceIDs); if (resultSL != SL_RESULT_SUCCESS) { return ma_result_from_OpenSL(resultSL); } for (SLint32 iDevice = 0; iDevice < deviceCount; ++iDevice) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); deviceInfo.id.opensl = pDeviceIDs[iDevice]; SLAudioOutputDescriptor desc; resultSL = (*deviceCaps)->QueryAudioOutputCapabilities(deviceCaps, deviceInfo.id.opensl, &desc); if (resultSL == SL_RESULT_SUCCESS) { ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), (const char*)desc.pDeviceName, (size_t)-1); ma_bool32 cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); if (cbResult == MA_FALSE) { isTerminated = MA_TRUE; break; } } } } /* Capture */ if (!isTerminated) { resultSL = (*deviceCaps)->GetAvailableAudioInputs(deviceCaps, &deviceCount, pDeviceIDs); if (resultSL != SL_RESULT_SUCCESS) { return ma_result_from_OpenSL(resultSL); } for (SLint32 iDevice = 0; iDevice < deviceCount; ++iDevice) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); deviceInfo.id.opensl = pDeviceIDs[iDevice]; SLAudioInputDescriptor desc; resultSL = (*deviceCaps)->QueryAudioInputCapabilities(deviceCaps, deviceInfo.id.opensl, &desc); if (resultSL == SL_RESULT_SUCCESS) { ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), (const char*)desc.deviceName, (size_t)-1); ma_bool32 cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); if (cbResult == MA_FALSE) { isTerminated = MA_TRUE; break; } } } } return MA_SUCCESS; #else goto return_default_device; #endif return_default_device:; cbResult = MA_TRUE; /* Playback. */ if (cbResult) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); deviceInfo.id.opensl = SL_DEFAULTDEVICEID_AUDIOOUTPUT; ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } /* Capture. */ if (cbResult) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); deviceInfo.id.opensl = SL_DEFAULTDEVICEID_AUDIOINPUT; ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); } return MA_SUCCESS; } static void ma_context_add_data_format_ex__opensl(ma_context* pContext, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_device_info* pDeviceInfo) { MA_ASSERT(pContext != NULL); MA_ASSERT(pDeviceInfo != NULL); pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate; pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0; pDeviceInfo->nativeDataFormatCount += 1; } static void ma_context_add_data_format__opensl(ma_context* pContext, ma_format format, ma_device_info* pDeviceInfo) { ma_uint32 minChannels = 1; ma_uint32 maxChannels = 2; ma_uint32 minSampleRate = (ma_uint32)ma_standard_sample_rate_8000; ma_uint32 maxSampleRate = (ma_uint32)ma_standard_sample_rate_48000; ma_uint32 iChannel; ma_uint32 iSampleRate; MA_ASSERT(pContext != NULL); MA_ASSERT(pDeviceInfo != NULL); /* Each sample format can support mono and stereo, and we'll support a small subset of standard rates (up to 48000). A better solution would be to somehow find a native sample rate. */ for (iChannel = minChannels; iChannel < maxChannels; iChannel += 1) { for (iSampleRate = 0; iSampleRate < ma_countof(g_maStandardSampleRatePriorities); iSampleRate += 1) { ma_uint32 standardSampleRate = g_maStandardSampleRatePriorities[iSampleRate]; if (standardSampleRate >= minSampleRate && standardSampleRate <= maxSampleRate) { ma_context_add_data_format_ex__opensl(pContext, format, iChannel, standardSampleRate, pDeviceInfo); } } } } static ma_result ma_context_get_device_info__opensl(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { MA_ASSERT(pContext != NULL); MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to get device info. */ if (g_maOpenSLInitCounter == 0) { return MA_INVALID_OPERATION; } /* TODO: Test Me. This is currently untested, so for now we are just returning default devices. */ #if 0 && !defined(MA_ANDROID) SLAudioIODeviceCapabilitiesItf deviceCaps; SLresult resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, (SLInterfaceID)pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES, &deviceCaps); if (resultSL != SL_RESULT_SUCCESS) { /* The interface may not be supported so just report a default device. */ goto return_default_device; } if (deviceType == ma_device_type_playback) { SLAudioOutputDescriptor desc; resultSL = (*deviceCaps)->QueryAudioOutputCapabilities(deviceCaps, pDeviceID->opensl, &desc); if (resultSL != SL_RESULT_SUCCESS) { return ma_result_from_OpenSL(resultSL); } ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (const char*)desc.pDeviceName, (size_t)-1); } else { SLAudioInputDescriptor desc; resultSL = (*deviceCaps)->QueryAudioInputCapabilities(deviceCaps, pDeviceID->opensl, &desc); if (resultSL != SL_RESULT_SUCCESS) { return ma_result_from_OpenSL(resultSL); } ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (const char*)desc.deviceName, (size_t)-1); } goto return_detailed_info; #else goto return_default_device; #endif return_default_device: if (pDeviceID != NULL) { if ((deviceType == ma_device_type_playback && pDeviceID->opensl != SL_DEFAULTDEVICEID_AUDIOOUTPUT) || (deviceType == ma_device_type_capture && pDeviceID->opensl != SL_DEFAULTDEVICEID_AUDIOINPUT)) { return MA_NO_DEVICE; /* Don't know the device. */ } } /* ID and Name / Description */ if (deviceType == ma_device_type_playback) { pDeviceInfo->id.opensl = SL_DEFAULTDEVICEID_AUDIOOUTPUT; ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); } else { pDeviceInfo->id.opensl = SL_DEFAULTDEVICEID_AUDIOINPUT; ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); } pDeviceInfo->isDefault = MA_TRUE; goto return_detailed_info; return_detailed_info: /* For now we're just outputting a set of values that are supported by the API but not necessarily supported by the device natively. Later on we should work on this so that it more closely reflects the device's actual native format. */ pDeviceInfo->nativeDataFormatCount = 0; #if defined(MA_ANDROID) && __ANDROID_API__ >= 21 ma_context_add_data_format__opensl(pContext, ma_format_f32, pDeviceInfo); #endif ma_context_add_data_format__opensl(pContext, ma_format_s16, pDeviceInfo); ma_context_add_data_format__opensl(pContext, ma_format_u8, pDeviceInfo); return MA_SUCCESS; } #ifdef MA_ANDROID /*void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, SLuint32 eventFlags, const void* pBuffer, SLuint32 bufferSize, SLuint32 dataUsed, void* pContext)*/ static void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData) { ma_device* pDevice = (ma_device*)pUserData; size_t periodSizeInBytes; ma_uint8* pBuffer; SLresult resultSL; MA_ASSERT(pDevice != NULL); (void)pBufferQueue; /* For now, don't do anything unless the buffer was fully processed. From what I can tell, it looks like OpenSL|ES 1.1 improves on buffer queues to the point that we could much more intelligently handle this, but unfortunately it looks like Android is only supporting OpenSL|ES 1.0.1 for now :( */ /* Don't do anything if the device is not started. */ if (ma_device_get_state(pDevice) != ma_device_state_started) { return; } /* Don't do anything if the device is being drained. */ if (pDevice->opensl.isDrainingCapture) { return; } periodSizeInBytes = pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); pBuffer = pDevice->opensl.pBufferCapture + (pDevice->opensl.currentBufferIndexCapture * periodSizeInBytes); ma_device_handle_backend_data_callback(pDevice, NULL, pBuffer, pDevice->capture.internalPeriodSizeInFrames); resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pBuffer, periodSizeInBytes); if (resultSL != SL_RESULT_SUCCESS) { return; } pDevice->opensl.currentBufferIndexCapture = (pDevice->opensl.currentBufferIndexCapture + 1) % pDevice->capture.internalPeriods; } static void ma_buffer_queue_callback_playback__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData) { ma_device* pDevice = (ma_device*)pUserData; size_t periodSizeInBytes; ma_uint8* pBuffer; SLresult resultSL; MA_ASSERT(pDevice != NULL); (void)pBufferQueue; /* Don't do anything if the device is not started. */ if (ma_device_get_state(pDevice) != ma_device_state_started) { return; } /* Don't do anything if the device is being drained. */ if (pDevice->opensl.isDrainingPlayback) { return; } periodSizeInBytes = pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); pBuffer = pDevice->opensl.pBufferPlayback + (pDevice->opensl.currentBufferIndexPlayback * periodSizeInBytes); ma_device_handle_backend_data_callback(pDevice, pBuffer, NULL, pDevice->playback.internalPeriodSizeInFrames); resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pBuffer, periodSizeInBytes); if (resultSL != SL_RESULT_SUCCESS) { return; } pDevice->opensl.currentBufferIndexPlayback = (pDevice->opensl.currentBufferIndexPlayback + 1) % pDevice->playback.internalPeriods; } #endif static ma_result ma_device_uninit__opensl(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before uninitializing the device. */ if (g_maOpenSLInitCounter == 0) { return MA_INVALID_OPERATION; } if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { if (pDevice->opensl.pAudioRecorderObj) { MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioRecorderObj); } ma_free(pDevice->opensl.pBufferCapture, &pDevice->pContext->allocationCallbacks); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { if (pDevice->opensl.pAudioPlayerObj) { MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioPlayerObj); } if (pDevice->opensl.pOutputMixObj) { MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Destroy((SLObjectItf)pDevice->opensl.pOutputMixObj); } ma_free(pDevice->opensl.pBufferPlayback, &pDevice->pContext->allocationCallbacks); } return MA_SUCCESS; } #if defined(MA_ANDROID) && __ANDROID_API__ >= 21 typedef SLAndroidDataFormat_PCM_EX ma_SLDataFormat_PCM; #else typedef SLDataFormat_PCM ma_SLDataFormat_PCM; #endif static ma_result ma_SLDataFormat_PCM_init__opensl(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* channelMap, ma_SLDataFormat_PCM* pDataFormat) { /* We need to convert our format/channels/rate so that they aren't set to default. */ if (format == ma_format_unknown) { format = MA_DEFAULT_FORMAT; } if (channels == 0) { channels = MA_DEFAULT_CHANNELS; } if (sampleRate == 0) { sampleRate = MA_DEFAULT_SAMPLE_RATE; } #if defined(MA_ANDROID) && __ANDROID_API__ >= 21 if (format == ma_format_f32) { pDataFormat->formatType = SL_ANDROID_DATAFORMAT_PCM_EX; pDataFormat->representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT; } else { pDataFormat->formatType = SL_DATAFORMAT_PCM; } #else pDataFormat->formatType = SL_DATAFORMAT_PCM; #endif pDataFormat->numChannels = channels; ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec = ma_round_to_standard_sample_rate__opensl(sampleRate * 1000); /* In millihertz. Annoyingly, the sample rate variable is named differently between SLAndroidDataFormat_PCM_EX and SLDataFormat_PCM */ pDataFormat->bitsPerSample = ma_get_bytes_per_sample(format) * 8; pDataFormat->channelMask = ma_channel_map_to_channel_mask__opensl(channelMap, channels); pDataFormat->endianness = (ma_is_little_endian()) ? SL_BYTEORDER_LITTLEENDIAN : SL_BYTEORDER_BIGENDIAN; /* Android has a few restrictions on the format as documented here: https://developer.android.com/ndk/guides/audio/opensl-for-android.html - Only mono and stereo is supported. - Only u8 and s16 formats are supported. - Maximum sample rate of 48000. */ #ifdef MA_ANDROID if (pDataFormat->numChannels > 2) { pDataFormat->numChannels = 2; } #if __ANDROID_API__ >= 21 if (pDataFormat->formatType == SL_ANDROID_DATAFORMAT_PCM_EX) { /* It's floating point. */ MA_ASSERT(pDataFormat->representation == SL_ANDROID_PCM_REPRESENTATION_FLOAT); if (pDataFormat->bitsPerSample > 32) { pDataFormat->bitsPerSample = 32; } } else { if (pDataFormat->bitsPerSample > 16) { pDataFormat->bitsPerSample = 16; } } #else if (pDataFormat->bitsPerSample > 16) { pDataFormat->bitsPerSample = 16; } #endif if (((SLDataFormat_PCM*)pDataFormat)->samplesPerSec > SL_SAMPLINGRATE_48) { ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec = SL_SAMPLINGRATE_48; } #endif pDataFormat->containerSize = pDataFormat->bitsPerSample; /* Always tightly packed for now. */ return MA_SUCCESS; } static ma_result ma_deconstruct_SLDataFormat_PCM__opensl(ma_SLDataFormat_PCM* pDataFormat, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap) { ma_bool32 isFloatingPoint = MA_FALSE; #if defined(MA_ANDROID) && __ANDROID_API__ >= 21 if (pDataFormat->formatType == SL_ANDROID_DATAFORMAT_PCM_EX) { MA_ASSERT(pDataFormat->representation == SL_ANDROID_PCM_REPRESENTATION_FLOAT); isFloatingPoint = MA_TRUE; } #endif if (isFloatingPoint) { if (pDataFormat->bitsPerSample == 32) { *pFormat = ma_format_f32; } } else { if (pDataFormat->bitsPerSample == 8) { *pFormat = ma_format_u8; } else if (pDataFormat->bitsPerSample == 16) { *pFormat = ma_format_s16; } else if (pDataFormat->bitsPerSample == 24) { *pFormat = ma_format_s24; } else if (pDataFormat->bitsPerSample == 32) { *pFormat = ma_format_s32; } } *pChannels = pDataFormat->numChannels; *pSampleRate = ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec / 1000; ma_channel_mask_to_channel_map__opensl(pDataFormat->channelMask, ma_min(pDataFormat->numChannels, channelMapCap), pChannelMap); return MA_SUCCESS; } static ma_result ma_device_init__opensl(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { #ifdef MA_ANDROID SLDataLocator_AndroidSimpleBufferQueue queue; SLresult resultSL; size_t bufferSizeInBytes; SLInterfaceID itfIDs[2]; const SLboolean itfIDsRequired[] = { SL_BOOLEAN_TRUE, /* SL_IID_ANDROIDSIMPLEBUFFERQUEUE */ SL_BOOLEAN_FALSE /* SL_IID_ANDROIDCONFIGURATION */ }; #endif MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to initialize a new device. */ if (g_maOpenSLInitCounter == 0) { return MA_INVALID_OPERATION; } if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } /* For now, only supporting Android implementations of OpenSL|ES since that's the only one I've been able to test with and I currently depend on Android-specific extensions (simple buffer queues). */ #ifdef MA_ANDROID itfIDs[0] = (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE; itfIDs[1] = (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDCONFIGURATION; /* No exclusive mode with OpenSL|ES. */ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) || ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) { return MA_SHARE_MODE_NOT_SUPPORTED; } /* Now we can start initializing the device properly. */ MA_ASSERT(pDevice != NULL); MA_ZERO_OBJECT(&pDevice->opensl); queue.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE; if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { ma_SLDataFormat_PCM pcm; SLDataLocator_IODevice locatorDevice; SLDataSource source; SLDataSink sink; SLAndroidConfigurationItf pRecorderConfig; ma_SLDataFormat_PCM_init__opensl(pDescriptorCapture->format, pDescriptorCapture->channels, pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, &pcm); locatorDevice.locatorType = SL_DATALOCATOR_IODEVICE; locatorDevice.deviceType = SL_IODEVICE_AUDIOINPUT; locatorDevice.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT; /* Must always use the default device with Android. */ locatorDevice.device = NULL; source.pLocator = &locatorDevice; source.pFormat = NULL; queue.numBuffers = pDescriptorCapture->periodCount; sink.pLocator = &queue; sink.pFormat = (SLDataFormat_PCM*)&pcm; resultSL = (*g_maEngineSL)->CreateAudioRecorder(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioRecorderObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired); if (resultSL == SL_RESULT_CONTENT_UNSUPPORTED || resultSL == SL_RESULT_PARAMETER_INVALID) { /* Unsupported format. Fall back to something safer and try again. If this fails, just abort. */ pcm.formatType = SL_DATAFORMAT_PCM; pcm.numChannels = 1; ((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16; /* The name of the sample rate variable is different between SLAndroidDataFormat_PCM_EX and SLDataFormat_PCM. */ pcm.bitsPerSample = 16; pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */ pcm.channelMask = 0; resultSL = (*g_maEngineSL)->CreateAudioRecorder(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioRecorderObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired); } if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio recorder."); return ma_result_from_OpenSL(resultSL); } /* Set the recording preset before realizing the player. */ if (pConfig->opensl.recordingPreset != ma_opensl_recording_preset_default) { resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDCONFIGURATION, &pRecorderConfig); if (resultSL == SL_RESULT_SUCCESS) { SLint32 recordingPreset = ma_to_recording_preset__opensl(pConfig->opensl.recordingPreset); resultSL = (*pRecorderConfig)->SetConfiguration(pRecorderConfig, SL_ANDROID_KEY_RECORDING_PRESET, &recordingPreset, sizeof(SLint32)); if (resultSL != SL_RESULT_SUCCESS) { /* Failed to set the configuration. Just keep going. */ } } } resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Realize((SLObjectItf)pDevice->opensl.pAudioRecorderObj, SL_BOOLEAN_FALSE); if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio recorder."); return ma_result_from_OpenSL(resultSL); } resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_RECORD, &pDevice->opensl.pAudioRecorder); if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_RECORD interface."); return ma_result_from_OpenSL(resultSL); } resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueueCapture); if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface."); return ma_result_from_OpenSL(resultSL); } resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, ma_buffer_queue_callback_capture__opensl_android, pDevice); if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback."); return ma_result_from_OpenSL(resultSL); } /* The internal format is determined by the "pcm" object. */ ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDescriptorCapture->format, &pDescriptorCapture->channels, &pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap)); /* Buffer. */ pDescriptorCapture->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile); pDevice->opensl.currentBufferIndexCapture = 0; bufferSizeInBytes = pDescriptorCapture->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) * pDescriptorCapture->periodCount; pDevice->opensl.pBufferCapture = (ma_uint8*)ma_calloc(bufferSizeInBytes, &pDevice->pContext->allocationCallbacks); if (pDevice->opensl.pBufferCapture == NULL) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer."); return MA_OUT_OF_MEMORY; } MA_ZERO_MEMORY(pDevice->opensl.pBufferCapture, bufferSizeInBytes); } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { ma_SLDataFormat_PCM pcm; SLDataSource source; SLDataLocator_OutputMix outmixLocator; SLDataSink sink; SLAndroidConfigurationItf pPlayerConfig; ma_SLDataFormat_PCM_init__opensl(pDescriptorPlayback->format, pDescriptorPlayback->channels, pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, &pcm); resultSL = (*g_maEngineSL)->CreateOutputMix(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pOutputMixObj, 0, NULL, NULL); if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create output mix."); return ma_result_from_OpenSL(resultSL); } resultSL = MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Realize((SLObjectItf)pDevice->opensl.pOutputMixObj, SL_BOOLEAN_FALSE); if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize output mix object."); return ma_result_from_OpenSL(resultSL); } resultSL = MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->GetInterface((SLObjectItf)pDevice->opensl.pOutputMixObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_OUTPUTMIX, &pDevice->opensl.pOutputMix); if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_OUTPUTMIX interface."); return ma_result_from_OpenSL(resultSL); } /* Set the output device. */ if (pDescriptorPlayback->pDeviceID != NULL) { SLuint32 deviceID_OpenSL = pDescriptorPlayback->pDeviceID->opensl; MA_OPENSL_OUTPUTMIX(pDevice->opensl.pOutputMix)->ReRoute((SLOutputMixItf)pDevice->opensl.pOutputMix, 1, &deviceID_OpenSL); } queue.numBuffers = pDescriptorPlayback->periodCount; source.pLocator = &queue; source.pFormat = (SLDataFormat_PCM*)&pcm; outmixLocator.locatorType = SL_DATALOCATOR_OUTPUTMIX; outmixLocator.outputMix = (SLObjectItf)pDevice->opensl.pOutputMixObj; sink.pLocator = &outmixLocator; sink.pFormat = NULL; resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired); if (resultSL == SL_RESULT_CONTENT_UNSUPPORTED || resultSL == SL_RESULT_PARAMETER_INVALID) { /* Unsupported format. Fall back to something safer and try again. If this fails, just abort. */ pcm.formatType = SL_DATAFORMAT_PCM; pcm.numChannels = 2; ((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16; pcm.bitsPerSample = 16; pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */ pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired); } if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio player."); return ma_result_from_OpenSL(resultSL); } /* Set the stream type before realizing the player. */ if (pConfig->opensl.streamType != ma_opensl_stream_type_default) { resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDCONFIGURATION, &pPlayerConfig); if (resultSL == SL_RESULT_SUCCESS) { SLint32 streamType = ma_to_stream_type__opensl(pConfig->opensl.streamType); resultSL = (*pPlayerConfig)->SetConfiguration(pPlayerConfig, SL_ANDROID_KEY_STREAM_TYPE, &streamType, sizeof(SLint32)); if (resultSL != SL_RESULT_SUCCESS) { /* Failed to set the configuration. Just keep going. */ } } } resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Realize((SLObjectItf)pDevice->opensl.pAudioPlayerObj, SL_BOOLEAN_FALSE); if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio player."); return ma_result_from_OpenSL(resultSL); } resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_PLAY, &pDevice->opensl.pAudioPlayer); if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_PLAY interface."); return ma_result_from_OpenSL(resultSL); } resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueuePlayback); if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface."); return ma_result_from_OpenSL(resultSL); } resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, ma_buffer_queue_callback_playback__opensl_android, pDevice); if (resultSL != SL_RESULT_SUCCESS) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback."); return ma_result_from_OpenSL(resultSL); } /* The internal format is determined by the "pcm" object. */ ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDescriptorPlayback->format, &pDescriptorPlayback->channels, &pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, ma_countof(pDescriptorPlayback->channelMap)); /* Buffer. */ pDescriptorPlayback->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile); pDevice->opensl.currentBufferIndexPlayback = 0; bufferSizeInBytes = pDescriptorPlayback->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels) * pDescriptorPlayback->periodCount; pDevice->opensl.pBufferPlayback = (ma_uint8*)ma_calloc(bufferSizeInBytes, &pDevice->pContext->allocationCallbacks); if (pDevice->opensl.pBufferPlayback == NULL) { ma_device_uninit__opensl(pDevice); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer."); return MA_OUT_OF_MEMORY; } MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, bufferSizeInBytes); } return MA_SUCCESS; #else return MA_NO_BACKEND; /* Non-Android implementations are not supported. */ #endif } static ma_result ma_device_start__opensl(ma_device* pDevice) { SLresult resultSL; size_t periodSizeInBytes; ma_uint32 iPeriod; MA_ASSERT(pDevice != NULL); MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to start the device. */ if (g_maOpenSLInitCounter == 0) { return MA_INVALID_OPERATION; } if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { resultSL = MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_RECORDING); if (resultSL != SL_RESULT_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal capture device."); return ma_result_from_OpenSL(resultSL); } periodSizeInBytes = pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) { resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pDevice->opensl.pBufferCapture + (periodSizeInBytes * iPeriod), periodSizeInBytes); if (resultSL != SL_RESULT_SUCCESS) { MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_STOPPED); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for capture device."); return ma_result_from_OpenSL(resultSL); } } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { resultSL = MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_PLAYING); if (resultSL != SL_RESULT_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal playback device."); return ma_result_from_OpenSL(resultSL); } /* In playback mode (no duplex) we need to load some initial buffers. In duplex mode we need to enqueue silent buffers. */ if (pDevice->type == ma_device_type_duplex) { MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); } else { ma_device__read_frames_from_client(pDevice, pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods, pDevice->opensl.pBufferPlayback); } periodSizeInBytes = pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; ++iPeriod) { resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pDevice->opensl.pBufferPlayback + (periodSizeInBytes * iPeriod), periodSizeInBytes); if (resultSL != SL_RESULT_SUCCESS) { MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_STOPPED); ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for playback device."); return ma_result_from_OpenSL(resultSL); } } } return MA_SUCCESS; } static ma_result ma_device_drain__opensl(ma_device* pDevice, ma_device_type deviceType) { SLAndroidSimpleBufferQueueItf pBufferQueue; MA_ASSERT(deviceType == ma_device_type_capture || deviceType == ma_device_type_playback); if (pDevice->type == ma_device_type_capture) { pBufferQueue = (SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture; pDevice->opensl.isDrainingCapture = MA_TRUE; } else { pBufferQueue = (SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback; pDevice->opensl.isDrainingPlayback = MA_TRUE; } for (;;) { SLAndroidSimpleBufferQueueState state; MA_OPENSL_BUFFERQUEUE(pBufferQueue)->GetState(pBufferQueue, &state); if (state.count == 0) { break; } ma_sleep(10); } if (pDevice->type == ma_device_type_capture) { pDevice->opensl.isDrainingCapture = MA_FALSE; } else { pDevice->opensl.isDrainingPlayback = MA_FALSE; } return MA_SUCCESS; } static ma_result ma_device_stop__opensl(ma_device* pDevice) { SLresult resultSL; MA_ASSERT(pDevice != NULL); MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before stopping/uninitializing the device. */ if (g_maOpenSLInitCounter == 0) { return MA_INVALID_OPERATION; } if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { ma_device_drain__opensl(pDevice, ma_device_type_capture); resultSL = MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_STOPPED); if (resultSL != SL_RESULT_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to stop internal capture device."); return ma_result_from_OpenSL(resultSL); } MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Clear((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { ma_device_drain__opensl(pDevice, ma_device_type_playback); resultSL = MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_STOPPED); if (resultSL != SL_RESULT_SUCCESS) { ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to stop internal playback device."); return ma_result_from_OpenSL(resultSL); } MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Clear((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback); } /* Make sure the client is aware that the device has stopped. There may be an OpenSL|ES callback for this, but I haven't found it. */ ma_device__on_notification_stopped(pDevice); return MA_SUCCESS; } static ma_result ma_context_uninit__opensl(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_opensl); (void)pContext; /* Uninit global data. */ ma_spinlock_lock(&g_maOpenSLSpinlock); { MA_ASSERT(g_maOpenSLInitCounter > 0); /* If you've triggered this, it means you have ma_context_init/uninit mismatch. Each successful call to ma_context_init() must be matched up with a call to ma_context_uninit(). */ g_maOpenSLInitCounter -= 1; if (g_maOpenSLInitCounter == 0) { (*g_maEngineObjectSL)->Destroy(g_maEngineObjectSL); } } ma_spinlock_unlock(&g_maOpenSLSpinlock); return MA_SUCCESS; } static ma_result ma_dlsym_SLInterfaceID__opensl(ma_context* pContext, const char* pName, ma_handle* pHandle) { /* We need to return an error if the symbol cannot be found. This is important because there have been reports that some symbols do not exist. */ ma_handle* p = (ma_handle*)ma_dlsym(ma_context_get_log(pContext), pContext->opensl.libOpenSLES, pName); if (p == NULL) { ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "[OpenSL] Cannot find symbol %s", pName); return MA_NO_BACKEND; } *pHandle = *p; return MA_SUCCESS; } static ma_result ma_context_init_engine_nolock__opensl(ma_context* pContext) { g_maOpenSLInitCounter += 1; if (g_maOpenSLInitCounter == 1) { SLresult resultSL; resultSL = ((ma_slCreateEngine_proc)pContext->opensl.slCreateEngine)(&g_maEngineObjectSL, 0, NULL, 0, NULL, NULL); if (resultSL != SL_RESULT_SUCCESS) { g_maOpenSLInitCounter -= 1; return ma_result_from_OpenSL(resultSL); } (*g_maEngineObjectSL)->Realize(g_maEngineObjectSL, SL_BOOLEAN_FALSE); resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, (SLInterfaceID)pContext->opensl.SL_IID_ENGINE, &g_maEngineSL); if (resultSL != SL_RESULT_SUCCESS) { (*g_maEngineObjectSL)->Destroy(g_maEngineObjectSL); g_maOpenSLInitCounter -= 1; return ma_result_from_OpenSL(resultSL); } } return MA_SUCCESS; } static ma_result ma_context_init__opensl(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { ma_result result; #if !defined(MA_NO_RUNTIME_LINKING) size_t i; const char* libOpenSLESNames[] = { "libOpenSLES.so" }; #endif MA_ASSERT(pContext != NULL); (void)pConfig; #if !defined(MA_NO_RUNTIME_LINKING) /* Dynamically link against libOpenSLES.so. I have now had multiple reports that SL_IID_ANDROIDSIMPLEBUFFERQUEUE cannot be found. One report was happening at compile time and another at runtime. To try working around this, I'm going to link to libOpenSLES at runtime and extract the symbols rather than reference them directly. This should, hopefully, fix these issues as the compiler won't see any references to the symbols and will hopefully skip the checks. */ for (i = 0; i < ma_countof(libOpenSLESNames); i += 1) { pContext->opensl.libOpenSLES = ma_dlopen(ma_context_get_log(pContext), libOpenSLESNames[i]); if (pContext->opensl.libOpenSLES != NULL) { break; } } if (pContext->opensl.libOpenSLES == NULL) { ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "[OpenSL] Could not find libOpenSLES.so"); return MA_NO_BACKEND; } result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_ENGINE", &pContext->opensl.SL_IID_ENGINE); if (result != MA_SUCCESS) { ma_dlclose(ma_context_get_log(pContext), pContext->opensl.libOpenSLES); return result; } result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_AUDIOIODEVICECAPABILITIES", &pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES); if (result != MA_SUCCESS) { ma_dlclose(ma_context_get_log(pContext), pContext->opensl.libOpenSLES); return result; } result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_ANDROIDSIMPLEBUFFERQUEUE", &pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE); if (result != MA_SUCCESS) { ma_dlclose(ma_context_get_log(pContext), pContext->opensl.libOpenSLES); return result; } result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_RECORD", &pContext->opensl.SL_IID_RECORD); if (result != MA_SUCCESS) { ma_dlclose(ma_context_get_log(pContext), pContext->opensl.libOpenSLES); return result; } result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_PLAY", &pContext->opensl.SL_IID_PLAY); if (result != MA_SUCCESS) { ma_dlclose(ma_context_get_log(pContext), pContext->opensl.libOpenSLES); return result; } result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_OUTPUTMIX", &pContext->opensl.SL_IID_OUTPUTMIX); if (result != MA_SUCCESS) { ma_dlclose(ma_context_get_log(pContext), pContext->opensl.libOpenSLES); return result; } result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_ANDROIDCONFIGURATION", &pContext->opensl.SL_IID_ANDROIDCONFIGURATION); if (result != MA_SUCCESS) { ma_dlclose(ma_context_get_log(pContext), pContext->opensl.libOpenSLES); return result; } pContext->opensl.slCreateEngine = (ma_proc)ma_dlsym(ma_context_get_log(pContext), pContext->opensl.libOpenSLES, "slCreateEngine"); if (pContext->opensl.slCreateEngine == NULL) { ma_dlclose(ma_context_get_log(pContext), pContext->opensl.libOpenSLES); ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "[OpenSL] Cannot find symbol slCreateEngine."); return MA_NO_BACKEND; } #else pContext->opensl.SL_IID_ENGINE = (ma_handle)SL_IID_ENGINE; pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES = (ma_handle)SL_IID_AUDIOIODEVICECAPABILITIES; pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE = (ma_handle)SL_IID_ANDROIDSIMPLEBUFFERQUEUE; pContext->opensl.SL_IID_RECORD = (ma_handle)SL_IID_RECORD; pContext->opensl.SL_IID_PLAY = (ma_handle)SL_IID_PLAY; pContext->opensl.SL_IID_OUTPUTMIX = (ma_handle)SL_IID_OUTPUTMIX; pContext->opensl.SL_IID_ANDROIDCONFIGURATION = (ma_handle)SL_IID_ANDROIDCONFIGURATION; pContext->opensl.slCreateEngine = (ma_proc)slCreateEngine; #endif /* Initialize global data first if applicable. */ ma_spinlock_lock(&g_maOpenSLSpinlock); { result = ma_context_init_engine_nolock__opensl(pContext); } ma_spinlock_unlock(&g_maOpenSLSpinlock); if (result != MA_SUCCESS) { ma_dlclose(ma_context_get_log(pContext), pContext->opensl.libOpenSLES); ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "[OpenSL] Failed to initialize OpenSL engine."); return result; } pCallbacks->onContextInit = ma_context_init__opensl; pCallbacks->onContextUninit = ma_context_uninit__opensl; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__opensl; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__opensl; pCallbacks->onDeviceInit = ma_device_init__opensl; pCallbacks->onDeviceUninit = ma_device_uninit__opensl; pCallbacks->onDeviceStart = ma_device_start__opensl; pCallbacks->onDeviceStop = ma_device_stop__opensl; pCallbacks->onDeviceRead = NULL; /* Not needed because OpenSL|ES is asynchronous. */ pCallbacks->onDeviceWrite = NULL; /* Not needed because OpenSL|ES is asynchronous. */ pCallbacks->onDeviceDataLoop = NULL; /* Not needed because OpenSL|ES is asynchronous. */ return MA_SUCCESS; } #endif /* OpenSL|ES */ /****************************************************************************** Web Audio Backend ******************************************************************************/ #ifdef MA_HAS_WEBAUDIO #include <emscripten/emscripten.h> #if (__EMSCRIPTEN_major__ > 3) || (__EMSCRIPTEN_major__ == 3 && (__EMSCRIPTEN_minor__ > 1 || (__EMSCRIPTEN_minor__ == 1 && __EMSCRIPTEN_tiny__ >= 32))) #include <emscripten/webaudio.h> #define MA_SUPPORT_AUDIO_WORKLETS #if (__EMSCRIPTEN_major__ > 3) || (__EMSCRIPTEN_major__ == 3 && (__EMSCRIPTEN_minor__ > 1 || (__EMSCRIPTEN_minor__ == 1 && __EMSCRIPTEN_tiny__ >= 70))) #define MA_SUPPORT_AUDIO_WORKLETS_VARIABLE_BUFFER_SIZE #endif #endif /* TODO: Version 0.12: Swap this logic around so that AudioWorklets are used by default. Add MA_NO_AUDIO_WORKLETS. */ #if defined(MA_ENABLE_AUDIO_WORKLETS) && defined(MA_SUPPORT_AUDIO_WORKLETS) #define MA_USE_AUDIO_WORKLETS #endif /* The thread stack size must be a multiple of 16. */ #ifndef MA_AUDIO_WORKLETS_THREAD_STACK_SIZE #define MA_AUDIO_WORKLETS_THREAD_STACK_SIZE 131072 #endif #if defined(MA_USE_AUDIO_WORKLETS) #define MA_WEBAUDIO_LATENCY_HINT_BALANCED "balanced" #define MA_WEBAUDIO_LATENCY_HINT_INTERACTIVE "interactive" #define MA_WEBAUDIO_LATENCY_HINT_PLAYBACK "playback" #endif static ma_bool32 ma_is_capture_supported__webaudio() { return EM_ASM_INT({ return (navigator.mediaDevices !== undefined && navigator.mediaDevices.getUserMedia !== undefined); }, 0) != 0; /* Must pass in a dummy argument for C99 compatibility. */ } #ifdef __cplusplus extern "C" { #endif void* EMSCRIPTEN_KEEPALIVE ma_malloc_emscripten(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks) { return ma_malloc(sz, pAllocationCallbacks); } void EMSCRIPTEN_KEEPALIVE ma_free_emscripten(void* p, const ma_allocation_callbacks* pAllocationCallbacks) { ma_free(p, pAllocationCallbacks); } void EMSCRIPTEN_KEEPALIVE ma_device_process_pcm_frames_capture__webaudio(ma_device* pDevice, int frameCount, float* pFrames) { ma_device_handle_backend_data_callback(pDevice, NULL, pFrames, (ma_uint32)frameCount); } void EMSCRIPTEN_KEEPALIVE ma_device_process_pcm_frames_playback__webaudio(ma_device* pDevice, int frameCount, float* pFrames) { ma_device_handle_backend_data_callback(pDevice, pFrames, NULL, (ma_uint32)frameCount); } #ifdef __cplusplus } #endif static ma_result ma_context_enumerate_devices__webaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { ma_bool32 cbResult = MA_TRUE; MA_ASSERT(pContext != NULL); MA_ASSERT(callback != NULL); /* Only supporting default devices for now. */ /* Playback. */ if (cbResult) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); deviceInfo.isDefault = MA_TRUE; /* Only supporting default devices. */ cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } /* Capture. */ if (cbResult) { if (ma_is_capture_supported__webaudio()) { ma_device_info deviceInfo; MA_ZERO_OBJECT(&deviceInfo); ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); deviceInfo.isDefault = MA_TRUE; /* Only supporting default devices. */ cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); } } return MA_SUCCESS; } static ma_result ma_context_get_device_info__webaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo) { MA_ASSERT(pContext != NULL); if (deviceType == ma_device_type_capture && !ma_is_capture_supported__webaudio()) { return MA_NO_DEVICE; } MA_ZERO_MEMORY(pDeviceInfo->id.webaudio, sizeof(pDeviceInfo->id.webaudio)); /* Only supporting default devices for now. */ (void)pDeviceID; if (deviceType == ma_device_type_playback) { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); } else { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); } /* Only supporting default devices. */ pDeviceInfo->isDefault = MA_TRUE; /* Web Audio can support any number of channels and sample rates. It only supports f32 formats, however. */ pDeviceInfo->nativeDataFormats[0].flags = 0; pDeviceInfo->nativeDataFormats[0].format = ma_format_unknown; pDeviceInfo->nativeDataFormats[0].channels = 0; /* All channels are supported. */ pDeviceInfo->nativeDataFormats[0].sampleRate = EM_ASM_INT({ try { var temp = new (window.AudioContext || window.webkitAudioContext)(); var sampleRate = temp.sampleRate; temp.close(); return sampleRate; } catch(e) { return 0; } }, 0); /* Must pass in a dummy argument for C99 compatibility. */ if (pDeviceInfo->nativeDataFormats[0].sampleRate == 0) { return MA_NO_DEVICE; } pDeviceInfo->nativeDataFormatCount = 1; return MA_SUCCESS; } static ma_result ma_device_uninit__webaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); #if defined(MA_USE_AUDIO_WORKLETS) { EM_ASM({ var device = window.miniaudio.get_device_by_index($0); if (device.streamNode !== undefined) { device.streamNode.disconnect(); device.streamNode = undefined; } device.pDevice = undefined; }, pDevice->webaudio.deviceIndex); emscripten_destroy_web_audio_node(pDevice->webaudio.audioWorklet); emscripten_destroy_audio_context(pDevice->webaudio.audioContext); ma_free(pDevice->webaudio.pStackBuffer, &pDevice->pContext->allocationCallbacks); } #else { EM_ASM({ var device = window.miniaudio.get_device_by_index($0); /* Make sure all nodes are disconnected and marked for collection. */ if (device.scriptNode !== undefined) { device.scriptNode.onaudioprocess = function(e) {}; /* We want to reset the callback to ensure it doesn't get called after AudioContext.close() has returned. Shouldn't happen since we're disconnecting, but just to be safe... */ device.scriptNode.disconnect(); device.scriptNode = undefined; } if (device.streamNode !== undefined) { device.streamNode.disconnect(); device.streamNode = undefined; } /* Stop the device. I think there is a chance the callback could get fired after calling this, hence why we want to clear the callback before closing. */ device.webaudio.close(); device.webaudio = undefined; device.pDevice = undefined; }, pDevice->webaudio.deviceIndex); } #endif /* Clean up the device on the JS side. */ EM_ASM({ window.miniaudio.untrack_device_by_index($0); }, pDevice->webaudio.deviceIndex); ma_free(pDevice->webaudio.pIntermediaryBuffer, &pDevice->pContext->allocationCallbacks); return MA_SUCCESS; } #if !defined(MA_USE_AUDIO_WORKLETS) static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__webaudio(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile) { /* There have been reports of the default buffer size being too small on some browsers. If we're using the default buffer size, we'll make sure the period size is bigger than our standard defaults. */ ma_uint32 periodSizeInFrames; if (nativeSampleRate == 0) { nativeSampleRate = MA_DEFAULT_SAMPLE_RATE; } if (pDescriptor->periodSizeInFrames == 0) { if (pDescriptor->periodSizeInMilliseconds == 0) { if (performanceProfile == ma_performance_profile_low_latency) { periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(33, nativeSampleRate); /* 1 frame @ 30 FPS */ } else { periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(333, nativeSampleRate); } } else { periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptor->periodSizeInMilliseconds, nativeSampleRate); } } else { periodSizeInFrames = pDescriptor->periodSizeInFrames; } /* The size of the buffer must be a power of 2 and between 256 and 16384. */ if (periodSizeInFrames < 256) { periodSizeInFrames = 256; } else if (periodSizeInFrames > 16384) { periodSizeInFrames = 16384; } else { periodSizeInFrames = ma_next_power_of_2(periodSizeInFrames); } return periodSizeInFrames; } #endif #if defined(MA_USE_AUDIO_WORKLETS) typedef struct { ma_device* pDevice; const ma_device_config* pConfig; ma_device_descriptor* pDescriptorPlayback; ma_device_descriptor* pDescriptorCapture; } ma_audio_worklet_thread_initialized_data; static EM_BOOL ma_audio_worklet_process_callback__webaudio(int inputCount, const AudioSampleFrame* pInputs, int outputCount, AudioSampleFrame* pOutputs, int paramCount, const AudioParamFrame* pParams, void* pUserData) { ma_device* pDevice = (ma_device*)pUserData; ma_uint32 frameCount; (void)paramCount; (void)pParams; /* The Emscripten documentation says that it'll always be 128 frames being passed in. Hard coding it like that feels like a very bad idea to me. Even if it's hard coded in the backend, the API and documentation should always refer to variables instead of a hard coded number. In any case, will follow along for the time being. Unfortunately the audio data is not interleaved so we'll need to convert it before we give the data to miniaudio for further processing. */ if (pDevice->type == ma_device_type_playback) { frameCount = pDevice->playback.internalPeriodSizeInFrames; } else { frameCount = pDevice->capture.internalPeriodSizeInFrames; } /* If this is called by the device has not yet been started we need to return early, making sure we output silence to the output buffer. */ if (ma_device_get_state(pDevice) != ma_device_state_started) { for (int i = 0; i < outputCount; i += 1) { MA_ZERO_MEMORY(pOutputs[i].data, pOutputs[i].numberOfChannels * frameCount * sizeof(float)); } return EM_TRUE; } if (inputCount > 0) { /* Input data needs to be interleaved before we hand it to the client. */ for (ma_uint32 iChannel = 0; iChannel < pDevice->capture.internalChannels; iChannel += 1) { for (ma_uint32 iFrame = 0; iFrame < frameCount; iFrame += 1) { pDevice->webaudio.pIntermediaryBuffer[iFrame*pDevice->capture.internalChannels + iChannel] = pInputs[0].data[frameCount*iChannel + iFrame]; } } ma_device_process_pcm_frames_capture__webaudio(pDevice, frameCount, pDevice->webaudio.pIntermediaryBuffer); } if (outputCount > 0) { /* If it's a capture-only device, we'll need to output silence. */ if (pDevice->type == ma_device_type_capture) { for (int i = 0; i < outputCount; i += 1) { MA_ZERO_MEMORY(pOutputs[i].data, pOutputs[i].numberOfChannels * frameCount * sizeof(float)); } } else { ma_device_process_pcm_frames_playback__webaudio(pDevice, frameCount, pDevice->webaudio.pIntermediaryBuffer); /* We've read the data from the client. Now we need to deinterleave the buffer and output to the output buffer. */ for (ma_uint32 iChannel = 0; iChannel < pDevice->playback.internalChannels; iChannel += 1) { for (ma_uint32 iFrame = 0; iFrame < frameCount; iFrame += 1) { pOutputs[0].data[frameCount*iChannel + iFrame] = pDevice->webaudio.pIntermediaryBuffer[iFrame*pDevice->playback.internalChannels + iChannel]; } } /* Just above we output data to the first output buffer. Here we just make sure we're putting silence into any remaining output buffers. */ for (int i = 1; i < outputCount; i += 1) { /* <-- Note that the counter starts at 1 instead of 0. */ MA_ZERO_MEMORY(pOutputs[i].data, pOutputs[i].numberOfChannels * frameCount * sizeof(float)); } } } return EM_TRUE; } static void ma_audio_worklet_processor_created__webaudio(EMSCRIPTEN_WEBAUDIO_T audioContext, EM_BOOL success, void* pUserData) { ma_audio_worklet_thread_initialized_data* pParameters = (ma_audio_worklet_thread_initialized_data*)pUserData; EmscriptenAudioWorkletNodeCreateOptions audioWorkletOptions; int channels = 0; size_t intermediaryBufferSizeInFrames; int sampleRate; if (success == EM_FALSE) { pParameters->pDevice->webaudio.initResult = MA_ERROR; ma_free(pParameters, &pParameters->pDevice->pContext->allocationCallbacks); return; } /* The next step is to initialize the audio worklet node. */ MA_ZERO_OBJECT(&audioWorkletOptions); /* The way channel counts work with Web Audio is confusing. As far as I can tell, there's no way to know the channel count from MediaStreamAudioSourceNode (what we use for capture)? The only way to have control is to configure an output channel count on the capture side. This is slightly confusing for capture mode because intuitively you wouldn't actually connect an output to an input-only node, but this is what we'll have to do in order to have proper control over the channel count. In the capture case, we'll have to output silence to its output node. */ if (pParameters->pConfig->deviceType == ma_device_type_capture) { channels = (int)((pParameters->pDescriptorCapture->channels > 0) ? pParameters->pDescriptorCapture->channels : MA_DEFAULT_CHANNELS); audioWorkletOptions.numberOfInputs = 1; } else { channels = (int)((pParameters->pDescriptorPlayback->channels > 0) ? pParameters->pDescriptorPlayback->channels : MA_DEFAULT_CHANNELS); if (pParameters->pConfig->deviceType == ma_device_type_duplex) { audioWorkletOptions.numberOfInputs = 1; } else { audioWorkletOptions.numberOfInputs = 0; } } audioWorkletOptions.numberOfOutputs = 1; audioWorkletOptions.outputChannelCounts = &channels; /* Now that we know the channel count to use we can allocate the intermediary buffer. The intermediary buffer is used for interleaving and deinterleaving. */ #if defined(MA_SUPPORT_AUDIO_WORKLETS_VARIABLE_BUFFER_SIZE) { intermediaryBufferSizeInFrames = (size_t)emscripten_audio_context_quantum_size(audioContext); } #else { intermediaryBufferSizeInFrames = 128; } #endif pParameters->pDevice->webaudio.pIntermediaryBuffer = (float*)ma_malloc(intermediaryBufferSizeInFrames * (ma_uint32)channels * sizeof(float), &pParameters->pDevice->pContext->allocationCallbacks); if (pParameters->pDevice->webaudio.pIntermediaryBuffer == NULL) { pParameters->pDevice->webaudio.initResult = MA_OUT_OF_MEMORY; ma_free(pParameters, &pParameters->pDevice->pContext->allocationCallbacks); return; } pParameters->pDevice->webaudio.audioWorklet = emscripten_create_wasm_audio_worklet_node(audioContext, "miniaudio", &audioWorkletOptions, &ma_audio_worklet_process_callback__webaudio, pParameters->pDevice); /* With the audio worklet initialized we can now attach it to the graph. */ if (pParameters->pConfig->deviceType == ma_device_type_capture || pParameters->pConfig->deviceType == ma_device_type_duplex) { ma_result attachmentResult = (ma_result)EM_ASM_INT({ var getUserMediaResult = 0; var audioWorklet = emscriptenGetAudioObject($0); var audioContext = emscriptenGetAudioObject($1); navigator.mediaDevices.getUserMedia({audio:true, video:false}) .then(function(stream) { audioContext.streamNode = audioContext.createMediaStreamSource(stream); audioContext.streamNode.connect(audioWorklet); audioWorklet.connect(audioContext.destination); getUserMediaResult = 0; /* 0 = MA_SUCCESS */ }) .catch(function(error) { console.log("navigator.mediaDevices.getUserMedia Failed: " + error); getUserMediaResult = -1; /* -1 = MA_ERROR */ }); return getUserMediaResult; }, pParameters->pDevice->webaudio.audioWorklet, audioContext); if (attachmentResult != MA_SUCCESS) { ma_log_postf(ma_device_get_log(pParameters->pDevice), MA_LOG_LEVEL_ERROR, "Web Audio: Failed to connect capture node."); emscripten_destroy_web_audio_node(pParameters->pDevice->webaudio.audioWorklet); pParameters->pDevice->webaudio.initResult = attachmentResult; ma_free(pParameters, &pParameters->pDevice->pContext->allocationCallbacks); return; } } /* If it's playback only we can now attach the worklet node to the graph. This has already been done for the duplex case. */ if (pParameters->pConfig->deviceType == ma_device_type_playback) { ma_result attachmentResult = (ma_result)EM_ASM_INT({ var audioWorklet = emscriptenGetAudioObject($0); var audioContext = emscriptenGetAudioObject($1); audioWorklet.connect(audioContext.destination); return 0; /* 0 = MA_SUCCESS */ }, pParameters->pDevice->webaudio.audioWorklet, audioContext); if (attachmentResult != MA_SUCCESS) { ma_log_postf(ma_device_get_log(pParameters->pDevice), MA_LOG_LEVEL_ERROR, "Web Audio: Failed to connect playback node."); pParameters->pDevice->webaudio.initResult = attachmentResult; ma_free(pParameters, &pParameters->pDevice->pContext->allocationCallbacks); return; } } /* We need to update the descriptors so that they reflect the internal data format. Both capture and playback should be the same. */ sampleRate = EM_ASM_INT({ return emscriptenGetAudioObject($0).sampleRate; }, audioContext); if (pParameters->pDescriptorCapture != NULL) { pParameters->pDescriptorCapture->format = ma_format_f32; pParameters->pDescriptorCapture->channels = (ma_uint32)channels; pParameters->pDescriptorCapture->sampleRate = (ma_uint32)sampleRate; ma_channel_map_init_standard(ma_standard_channel_map_webaudio, pParameters->pDescriptorCapture->channelMap, ma_countof(pParameters->pDescriptorCapture->channelMap), pParameters->pDescriptorCapture->channels); pParameters->pDescriptorCapture->periodSizeInFrames = intermediaryBufferSizeInFrames; pParameters->pDescriptorCapture->periodCount = 1; } if (pParameters->pDescriptorPlayback != NULL) { pParameters->pDescriptorPlayback->format = ma_format_f32; pParameters->pDescriptorPlayback->channels = (ma_uint32)channels; pParameters->pDescriptorPlayback->sampleRate = (ma_uint32)sampleRate; ma_channel_map_init_standard(ma_standard_channel_map_webaudio, pParameters->pDescriptorPlayback->channelMap, ma_countof(pParameters->pDescriptorPlayback->channelMap), pParameters->pDescriptorPlayback->channels); pParameters->pDescriptorPlayback->periodSizeInFrames = intermediaryBufferSizeInFrames; pParameters->pDescriptorPlayback->periodCount = 1; } /* At this point we're done and we can return. */ ma_log_postf(ma_device_get_log(pParameters->pDevice), MA_LOG_LEVEL_DEBUG, "AudioWorklets: Created worklet node: %d\n", pParameters->pDevice->webaudio.audioWorklet); pParameters->pDevice->webaudio.initResult = MA_SUCCESS; ma_free(pParameters, &pParameters->pDevice->pContext->allocationCallbacks); } static void ma_audio_worklet_thread_initialized__webaudio(EMSCRIPTEN_WEBAUDIO_T audioContext, EM_BOOL success, void* pUserData) { ma_audio_worklet_thread_initialized_data* pParameters = (ma_audio_worklet_thread_initialized_data*)pUserData; WebAudioWorkletProcessorCreateOptions workletProcessorOptions; MA_ASSERT(pParameters != NULL); if (success == EM_FALSE) { pParameters->pDevice->webaudio.initResult = MA_ERROR; return; } MA_ZERO_OBJECT(&workletProcessorOptions); workletProcessorOptions.name = "miniaudio"; /* I'm not entirely sure what to call this. Does this need to be globally unique, or does it need only be unique for a given AudioContext? */ emscripten_create_wasm_audio_worklet_processor_async(audioContext, &workletProcessorOptions, ma_audio_worklet_processor_created__webaudio, pParameters); } #endif static ma_result ma_device_init__webaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture) { if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } /* No exclusive mode with Web Audio. */ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) || ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) { return MA_SHARE_MODE_NOT_SUPPORTED; } /* With AudioWorklets we'll have just a single AudioContext. I'm not sure why I'm not doing this for ScriptProcessorNode so it might be worthwhile to look into that as well. */ #if defined(MA_USE_AUDIO_WORKLETS) { EmscriptenWebAudioCreateAttributes audioContextAttributes; ma_audio_worklet_thread_initialized_data* pInitParameters; void* pStackBuffer; if (pConfig->performanceProfile == ma_performance_profile_conservative) { audioContextAttributes.latencyHint = MA_WEBAUDIO_LATENCY_HINT_PLAYBACK; } else { audioContextAttributes.latencyHint = MA_WEBAUDIO_LATENCY_HINT_INTERACTIVE; } /* In my testing, Firefox does not seem to capture audio data properly if the sample rate is set to anything other than 48K. This does not seem to be the case for other browsers. For this reason, if the device type is anything other than playback, we'll leave the sample rate as-is and let the browser pick the appropriate rate for us. */ if (pConfig->deviceType == ma_device_type_playback) { audioContextAttributes.sampleRate = pDescriptorPlayback->sampleRate; } else { audioContextAttributes.sampleRate = 0; } /* It's not clear if this can return an error. None of the tests in the Emscripten repository check for this, so neither am I for now. */ pDevice->webaudio.audioContext = emscripten_create_audio_context(&audioContextAttributes); /* With the context created we can now create the worklet. We can only have a single worklet per audio context which means we'll need to craft this appropriately to handle duplex devices correctly. */ /* We now need to create a worker thread. This is a bit weird because we need to allocate our own buffer for the thread's stack. The stack needs to be aligned to 16 bytes. I'm going to allocate this on the heap to keep it simple. */ pStackBuffer = ma_aligned_malloc(MA_AUDIO_WORKLETS_THREAD_STACK_SIZE, 16, &pDevice->pContext->allocationCallbacks); if (pStackBuffer == NULL) { emscripten_destroy_audio_context(pDevice->webaudio.audioContext); return MA_OUT_OF_MEMORY; } /* Our thread initialization parameters need to be allocated on the heap so they don't go out of scope. */ pInitParameters = (ma_audio_worklet_thread_initialized_data*)ma_malloc(sizeof(*pInitParameters), &pDevice->pContext->allocationCallbacks); if (pInitParameters == NULL) { ma_free(pStackBuffer, &pDevice->pContext->allocationCallbacks); emscripten_destroy_audio_context(pDevice->webaudio.audioContext); return MA_OUT_OF_MEMORY; } pInitParameters->pDevice = pDevice; pInitParameters->pConfig = pConfig; pInitParameters->pDescriptorPlayback = pDescriptorPlayback; pInitParameters->pDescriptorCapture = pDescriptorCapture; /* We need to flag the device as not yet initialized so we can wait on it later. Unfortunately all of the Emscripten WebAudio stuff is asynchronous. */ pDevice->webaudio.initResult = MA_BUSY; { emscripten_start_wasm_audio_worklet_thread_async(pDevice->webaudio.audioContext, pStackBuffer, MA_AUDIO_WORKLETS_THREAD_STACK_SIZE, ma_audio_worklet_thread_initialized__webaudio, pInitParameters); } while (pDevice->webaudio.initResult == MA_BUSY) { emscripten_sleep(1); } /* We must wait for initialization to complete. We're just spinning here. The emscripten_sleep() call is why we need to build with `-sASYNCIFY`. */ /* Initialization is now complete. Descriptors were updated when the worklet was initialized. */ if (pDevice->webaudio.initResult != MA_SUCCESS) { ma_free(pStackBuffer, &pDevice->pContext->allocationCallbacks); emscripten_destroy_audio_context(pDevice->webaudio.audioContext); return pDevice->webaudio.initResult; } /* We need to add an entry to the miniaudio.devices list on the JS side so we can do some JS/C interop. */ pDevice->webaudio.deviceIndex = EM_ASM_INT({ return window.miniaudio.track_device({ webaudio: emscriptenGetAudioObject($0), state: 1, /* 1 = ma_device_state_stopped */ pDevice: $1 }); }, pDevice->webaudio.audioContext, pDevice); return MA_SUCCESS; } #else { /* ScriptProcessorNode. This path requires us to do almost everything in JS, but we'll do as much as we can in C. */ ma_uint32 deviceIndex; ma_uint32 channels; ma_uint32 sampleRate; ma_uint32 periodSizeInFrames; /* The channel count will depend on the device type. If it's a capture, use its, otherwise use the playback side. */ if (pConfig->deviceType == ma_device_type_capture) { channels = (pDescriptorCapture->channels > 0) ? pDescriptorCapture->channels : MA_DEFAULT_CHANNELS; } else { channels = (pDescriptorPlayback->channels > 0) ? pDescriptorPlayback->channels : MA_DEFAULT_CHANNELS; } /* When testing in Firefox, I've seen it where capture mode fails if the sample rate is changed to anything other than it's native rate. For this reason we're leaving the sample rate untouched for capture devices. */ if (pConfig->deviceType == ma_device_type_playback) { sampleRate = pDescriptorPlayback->sampleRate; } else { sampleRate = 0; /* Let the browser decide when capturing. */ } /* The period size needs to be a power of 2. */ if (pConfig->deviceType == ma_device_type_capture) { periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__webaudio(pDescriptorCapture, sampleRate, pConfig->performanceProfile); } else { periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__webaudio(pDescriptorPlayback, sampleRate, pConfig->performanceProfile); } /* We need an intermediary buffer for doing interleaving and deinterleaving. */ pDevice->webaudio.pIntermediaryBuffer = (float*)ma_malloc(periodSizeInFrames * channels * sizeof(float), &pDevice->pContext->allocationCallbacks); if (pDevice->webaudio.pIntermediaryBuffer == NULL) { return MA_OUT_OF_MEMORY; } deviceIndex = EM_ASM_INT({ var deviceType = $0; var channels = $1; var sampleRate = $2; var bufferSize = $3; var pIntermediaryBuffer = $4; var pDevice = $5; if (typeof(window.miniaudio) === 'undefined') { return -1; /* Context not initialized. */ } var device = {}; /* First thing we need is an AudioContext. */ var audioContextOptions = {}; if (deviceType == window.miniaudio.device_type.playback && sampleRate != 0) { audioContextOptions.sampleRate = sampleRate; } device.webaudio = new (window.AudioContext || window.webkitAudioContext)(audioContextOptions); device.webaudio.suspend(); /* The AudioContext must be created in a suspended state. */ device.state = window.miniaudio.device_state.stopped; /* We need to create a ScriptProcessorNode. The channel situation is the same as the AudioWorklet path in that we need to specify an output and configure the channel count there. */ var channelCountIn = 0; var channelCountOut = channels; if (deviceType != window.miniaudio.device_type.playback) { channelCountIn = channels; } device.scriptNode = device.webaudio.createScriptProcessor(bufferSize, channelCountIn, channelCountOut); /* The node processing callback. */ device.scriptNode.onaudioprocess = function(e) { if (device.intermediaryBufferView == null || device.intermediaryBufferView.length == 0) { device.intermediaryBufferView = new Float32Array(HEAPF32.buffer, pIntermediaryBuffer, bufferSize * channels); } /* Do the capture side first. */ if (deviceType == window.miniaudio.device_type.capture || deviceType == window.miniaudio.device_type.duplex) { /* The data must be interleaved before being processed miniaudio. */ for (var iChannel = 0; iChannel < channels; iChannel += 1) { var inputBuffer = e.inputBuffer.getChannelData(iChannel); var intermediaryBuffer = device.intermediaryBufferView; for (var iFrame = 0; iFrame < bufferSize; iFrame += 1) { intermediaryBuffer[iFrame*channels + iChannel] = inputBuffer[iFrame]; } } _ma_device_process_pcm_frames_capture__webaudio(pDevice, bufferSize, pIntermediaryBuffer); } if (deviceType == window.miniaudio.device_type.playback || deviceType == window.miniaudio.device_type.duplex) { _ma_device_process_pcm_frames_playback__webaudio(pDevice, bufferSize, pIntermediaryBuffer); for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) { var outputBuffer = e.outputBuffer.getChannelData(iChannel); var intermediaryBuffer = device.intermediaryBufferView; for (var iFrame = 0; iFrame < bufferSize; iFrame += 1) { outputBuffer[iFrame] = intermediaryBuffer[iFrame*channels + iChannel]; } } } else { /* It's a capture-only device. Make sure the output is silenced. */ for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) { e.outputBuffer.getChannelData(iChannel).fill(0.0); } } }; /* Now we need to connect our node to the graph. */ if (deviceType == window.miniaudio.device_type.capture || deviceType == window.miniaudio.device_type.duplex) { navigator.mediaDevices.getUserMedia({audio:true, video:false}) .then(function(stream) { device.streamNode = device.webaudio.createMediaStreamSource(stream); device.streamNode.connect(device.scriptNode); device.scriptNode.connect(device.webaudio.destination); }) .catch(function(error) { console.log("Failed to get user media: " + error); }); } if (deviceType == window.miniaudio.device_type.playback) { device.scriptNode.connect(device.webaudio.destination); } device.pDevice = pDevice; return window.miniaudio.track_device(device); }, pConfig->deviceType, channels, sampleRate, periodSizeInFrames, pDevice->webaudio.pIntermediaryBuffer, pDevice); if (deviceIndex < 0) { return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } pDevice->webaudio.deviceIndex = deviceIndex; /* Grab the sample rate from the audio context directly. */ sampleRate = (ma_uint32)EM_ASM_INT({ return window.miniaudio.get_device_by_index($0).webaudio.sampleRate; }, deviceIndex); if (pDescriptorCapture != NULL) { pDescriptorCapture->format = ma_format_f32; pDescriptorCapture->channels = channels; pDescriptorCapture->sampleRate = sampleRate; ma_channel_map_init_standard(ma_standard_channel_map_webaudio, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorCapture->channels); pDescriptorCapture->periodSizeInFrames = periodSizeInFrames; pDescriptorCapture->periodCount = 1; } if (pDescriptorPlayback != NULL) { pDescriptorPlayback->format = ma_format_f32; pDescriptorPlayback->channels = channels; pDescriptorPlayback->sampleRate = sampleRate; ma_channel_map_init_standard(ma_standard_channel_map_webaudio, pDescriptorPlayback->channelMap, ma_countof(pDescriptorPlayback->channelMap), pDescriptorPlayback->channels); pDescriptorPlayback->periodSizeInFrames = periodSizeInFrames; pDescriptorPlayback->periodCount = 1; } return MA_SUCCESS; } #endif } static ma_result ma_device_start__webaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); EM_ASM({ var device = window.miniaudio.get_device_by_index($0); device.webaudio.resume(); device.state = window.miniaudio.device_state.started; }, pDevice->webaudio.deviceIndex); return MA_SUCCESS; } static ma_result ma_device_stop__webaudio(ma_device* pDevice) { MA_ASSERT(pDevice != NULL); /* From the WebAudio API documentation for AudioContext.suspend(): Suspends the progression of AudioContext's currentTime, allows any current context processing blocks that are already processed to be played to the destination, and then allows the system to release its claim on audio hardware. I read this to mean that "any current context processing blocks" are processed by suspend() - i.e. They they are drained. We therefore shouldn't need to do any kind of explicit draining. */ EM_ASM({ var device = window.miniaudio.get_device_by_index($0); device.webaudio.suspend(); device.state = window.miniaudio.device_state.stopped; }, pDevice->webaudio.deviceIndex); ma_device__on_notification_stopped(pDevice); return MA_SUCCESS; } static ma_result ma_context_uninit__webaudio(ma_context* pContext) { MA_ASSERT(pContext != NULL); MA_ASSERT(pContext->backend == ma_backend_webaudio); (void)pContext; /* Unused. */ /* Remove the global miniaudio object from window if there are no more references to it. */ EM_ASM({ if (typeof(window.miniaudio) !== 'undefined') { window.miniaudio.unlock_event_types.map(function(event_type) { document.removeEventListener(event_type, window.miniaudio.unlock, true); }); window.miniaudio.referenceCount -= 1; if (window.miniaudio.referenceCount === 0) { delete window.miniaudio; } } }); return MA_SUCCESS; } static ma_result ma_context_init__webaudio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks) { int resultFromJS; MA_ASSERT(pContext != NULL); (void)pConfig; /* Unused. */ /* Here is where our global JavaScript object is initialized. */ resultFromJS = EM_ASM_INT({ if (typeof window === 'undefined' || (window.AudioContext || window.webkitAudioContext) === undefined) { return 0; /* Web Audio not supported. */ } if (typeof(window.miniaudio) === 'undefined') { window.miniaudio = { referenceCount: 0 }; /* Device types. */ window.miniaudio.device_type = {}; window.miniaudio.device_type.playback = $0; window.miniaudio.device_type.capture = $1; window.miniaudio.device_type.duplex = $2; /* Device states. */ window.miniaudio.device_state = {}; window.miniaudio.device_state.stopped = $3; window.miniaudio.device_state.started = $4; /* Device cache for mapping devices to indexes for JavaScript/C interop. */ let miniaudio = window.miniaudio; miniaudio.devices = []; miniaudio.track_device = function(device) { /* Try inserting into a free slot first. */ for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) { if (miniaudio.devices[iDevice] == null) { miniaudio.devices[iDevice] = device; return iDevice; } } /* Getting here means there is no empty slots in the array so we just push to the end. */ miniaudio.devices.push(device); return miniaudio.devices.length - 1; }; miniaudio.untrack_device_by_index = function(deviceIndex) { /* We just set the device's slot to null. The slot will get reused in the next call to ma_track_device. */ miniaudio.devices[deviceIndex] = null; /* Trim the array if possible. */ while (miniaudio.devices.length > 0) { if (miniaudio.devices[miniaudio.devices.length-1] == null) { miniaudio.devices.pop(); } else { break; } } }; miniaudio.untrack_device = function(device) { for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) { if (miniaudio.devices[iDevice] == device) { return miniaudio.untrack_device_by_index(iDevice); } } }; miniaudio.get_device_by_index = function(deviceIndex) { return miniaudio.devices[deviceIndex]; }; miniaudio.unlock_event_types = (function(){ return ['touchend', 'click']; })(); miniaudio.unlock = function() { for(var i = 0; i < miniaudio.devices.length; ++i) { var device = miniaudio.devices[i]; if (device != null && device.webaudio != null && device.state === miniaudio.device_state.started) { device.webaudio.resume().then(() => { _ma_device__on_notification_unlocked(device.pDevice); }, (error) => {console.error("Failed to resume audiocontext", error); }); } } miniaudio.unlock_event_types.map(function(event_type) { document.removeEventListener(event_type, miniaudio.unlock, true); }); }; miniaudio.unlock_event_types.map(function(event_type) { document.addEventListener(event_type, miniaudio.unlock, true); }); } window.miniaudio.referenceCount += 1; return 1; }, ma_device_type_playback, ma_device_type_capture, ma_device_type_duplex, ma_device_state_stopped, ma_device_state_started); if (resultFromJS != 1) { return MA_FAILED_TO_INIT_BACKEND; } pCallbacks->onContextInit = ma_context_init__webaudio; pCallbacks->onContextUninit = ma_context_uninit__webaudio; pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__webaudio; pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__webaudio; pCallbacks->onDeviceInit = ma_device_init__webaudio; pCallbacks->onDeviceUninit = ma_device_uninit__webaudio; pCallbacks->onDeviceStart = ma_device_start__webaudio; pCallbacks->onDeviceStop = ma_device_stop__webaudio; pCallbacks->onDeviceRead = NULL; /* Not needed because WebAudio is asynchronous. */ pCallbacks->onDeviceWrite = NULL; /* Not needed because WebAudio is asynchronous. */ pCallbacks->onDeviceDataLoop = NULL; /* Not needed because WebAudio is asynchronous. */ return MA_SUCCESS; } #endif /* MA_HAS_WEBAUDIO */ static ma_bool32 ma__is_channel_map_valid(const ma_channel* pChannelMap, ma_uint32 channels) { /* A blank channel map should be allowed, in which case it should use an appropriate default which will depend on context. */ if (pChannelMap != NULL && pChannelMap[0] != MA_CHANNEL_NONE) { ma_uint32 iChannel; if (channels == 0 || channels > MA_MAX_CHANNELS) { return MA_FALSE; /* Channel count out of range. */ } /* A channel cannot be present in the channel map more than once. */ for (iChannel = 0; iChannel < channels; ++iChannel) { ma_uint32 jChannel; for (jChannel = iChannel + 1; jChannel < channels; ++jChannel) { if (pChannelMap[iChannel] == pChannelMap[jChannel]) { return MA_FALSE; } } } } return MA_TRUE; } static ma_bool32 ma_context_is_backend_asynchronous(ma_context* pContext) { MA_ASSERT(pContext != NULL); if (pContext->callbacks.onDeviceRead == NULL && pContext->callbacks.onDeviceWrite == NULL) { if (pContext->callbacks.onDeviceDataLoop == NULL) { return MA_TRUE; } else { return MA_FALSE; } } else { return MA_FALSE; } } static ma_result ma_device__post_init_setup(ma_device* pDevice, ma_device_type deviceType) { ma_result result; MA_ASSERT(pDevice != NULL); if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) { if (pDevice->capture.format == ma_format_unknown) { pDevice->capture.format = pDevice->capture.internalFormat; } if (pDevice->capture.channels == 0) { pDevice->capture.channels = pDevice->capture.internalChannels; } if (pDevice->capture.channelMap[0] == MA_CHANNEL_NONE) { MA_ASSERT(pDevice->capture.channels <= MA_MAX_CHANNELS); if (pDevice->capture.internalChannels == pDevice->capture.channels) { ma_channel_map_copy(pDevice->capture.channelMap, pDevice->capture.internalChannelMap, pDevice->capture.channels); } else { if (pDevice->capture.channelMixMode == ma_channel_mix_mode_simple) { ma_channel_map_init_blank(pDevice->capture.channelMap, pDevice->capture.channels); } else { ma_channel_map_init_standard(ma_standard_channel_map_default, pDevice->capture.channelMap, ma_countof(pDevice->capture.channelMap), pDevice->capture.channels); } } } } if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) { if (pDevice->playback.format == ma_format_unknown) { pDevice->playback.format = pDevice->playback.internalFormat; } if (pDevice->playback.channels == 0) { pDevice->playback.channels = pDevice->playback.internalChannels; } if (pDevice->playback.channelMap[0] == MA_CHANNEL_NONE) { MA_ASSERT(pDevice->playback.channels <= MA_MAX_CHANNELS); if (pDevice->playback.internalChannels == pDevice->playback.channels) { ma_channel_map_copy(pDevice->playback.channelMap, pDevice->playback.internalChannelMap, pDevice->playback.channels); } else { if (pDevice->playback.channelMixMode == ma_channel_mix_mode_simple) { ma_channel_map_init_blank(pDevice->playback.channelMap, pDevice->playback.channels); } else { ma_channel_map_init_standard(ma_standard_channel_map_default, pDevice->playback.channelMap, ma_countof(pDevice->playback.channelMap), pDevice->playback.channels); } } } } if (pDevice->sampleRate == 0) { if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) { pDevice->sampleRate = pDevice->capture.internalSampleRate; } else { pDevice->sampleRate = pDevice->playback.internalSampleRate; } } /* Data converters. */ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) { /* Converting from internal device format to client format. */ ma_data_converter_config converterConfig = ma_data_converter_config_init_default(); converterConfig.formatIn = pDevice->capture.internalFormat; converterConfig.channelsIn = pDevice->capture.internalChannels; converterConfig.sampleRateIn = pDevice->capture.internalSampleRate; converterConfig.pChannelMapIn = pDevice->capture.internalChannelMap; converterConfig.formatOut = pDevice->capture.format; converterConfig.channelsOut = pDevice->capture.channels; converterConfig.sampleRateOut = pDevice->sampleRate; converterConfig.pChannelMapOut = pDevice->capture.channelMap; converterConfig.channelMixMode = pDevice->capture.channelMixMode; converterConfig.calculateLFEFromSpatialChannels = pDevice->capture.calculateLFEFromSpatialChannels; converterConfig.allowDynamicSampleRate = MA_FALSE; converterConfig.resampling.algorithm = pDevice->resampling.algorithm; converterConfig.resampling.linear.lpfOrder = pDevice->resampling.linear.lpfOrder; converterConfig.resampling.pBackendVTable = pDevice->resampling.pBackendVTable; converterConfig.resampling.pBackendUserData = pDevice->resampling.pBackendUserData; /* Make sure the old converter is uninitialized first. */ if (ma_device_get_state(pDevice) != ma_device_state_uninitialized) { ma_data_converter_uninit(&pDevice->capture.converter, &pDevice->pContext->allocationCallbacks); } result = ma_data_converter_init(&converterConfig, &pDevice->pContext->allocationCallbacks, &pDevice->capture.converter); if (result != MA_SUCCESS) { return result; } } if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) { /* Converting from client format to device format. */ ma_data_converter_config converterConfig = ma_data_converter_config_init_default(); converterConfig.formatIn = pDevice->playback.format; converterConfig.channelsIn = pDevice->playback.channels; converterConfig.sampleRateIn = pDevice->sampleRate; converterConfig.pChannelMapIn = pDevice->playback.channelMap; converterConfig.formatOut = pDevice->playback.internalFormat; converterConfig.channelsOut = pDevice->playback.internalChannels; converterConfig.sampleRateOut = pDevice->playback.internalSampleRate; converterConfig.pChannelMapOut = pDevice->playback.internalChannelMap; converterConfig.channelMixMode = pDevice->playback.channelMixMode; converterConfig.calculateLFEFromSpatialChannels = pDevice->playback.calculateLFEFromSpatialChannels; converterConfig.allowDynamicSampleRate = MA_FALSE; converterConfig.resampling.algorithm = pDevice->resampling.algorithm; converterConfig.resampling.linear.lpfOrder = pDevice->resampling.linear.lpfOrder; converterConfig.resampling.pBackendVTable = pDevice->resampling.pBackendVTable; converterConfig.resampling.pBackendUserData = pDevice->resampling.pBackendUserData; /* Make sure the old converter is uninitialized first. */ if (ma_device_get_state(pDevice) != ma_device_state_uninitialized) { ma_data_converter_uninit(&pDevice->playback.converter, &pDevice->pContext->allocationCallbacks); } result = ma_data_converter_init(&converterConfig, &pDevice->pContext->allocationCallbacks, &pDevice->playback.converter); if (result != MA_SUCCESS) { return result; } } /* If the device is doing playback (ma_device_type_playback or ma_device_type_duplex), there's a couple of situations where we'll need a heap allocated cache. The first is a duplex device for backends that use a callback for data delivery. The reason this is needed is that the input stage needs to have a buffer to place the input data while it waits for the playback stage, after which the miniaudio data callback will get fired. This is not needed for backends that use a blocking API because miniaudio manages temporary buffers on the stack to achieve this. The other situation is when the data converter does not have the ability to query the number of input frames that are required in order to process a given number of output frames. When performing data conversion, it's useful if miniaudio know exactly how many frames it needs from the client in order to generate a given number of output frames. This way, only exactly the number of frames are needed to be read from the client which means no cache is necessary. On the other hand, if miniaudio doesn't know how many frames to read, it is forced to read in fixed sized chunks and then cache any residual unused input frames, those of which will be processed at a later stage. */ if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) { ma_uint64 unused; pDevice->playback.inputCacheConsumed = 0; pDevice->playback.inputCacheRemaining = 0; if (pDevice->type == ma_device_type_duplex || /* Duplex. backend may decide to use ma_device_handle_backend_data_callback() which will require this cache. */ ma_data_converter_get_required_input_frame_count(&pDevice->playback.converter, 1, &unused) != MA_SUCCESS) /* Data conversion required input frame calculation not supported. */ { /* We need a heap allocated cache. We want to size this based on the period size. */ void* pNewInputCache; ma_uint64 newInputCacheCap; ma_uint64 newInputCacheSizeInBytes; newInputCacheCap = ma_calculate_frame_count_after_resampling(pDevice->playback.internalSampleRate, pDevice->sampleRate, pDevice->playback.internalPeriodSizeInFrames); newInputCacheSizeInBytes = newInputCacheCap * ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); if (newInputCacheSizeInBytes > MA_SIZE_MAX) { ma_free(pDevice->playback.pInputCache, &pDevice->pContext->allocationCallbacks); pDevice->playback.pInputCache = NULL; pDevice->playback.inputCacheCap = 0; return MA_OUT_OF_MEMORY; /* Allocation too big. Should never hit this, but makes the cast below safer for 32-bit builds. */ } pNewInputCache = ma_realloc(pDevice->playback.pInputCache, (size_t)newInputCacheSizeInBytes, &pDevice->pContext->allocationCallbacks); if (pNewInputCache == NULL) { ma_free(pDevice->playback.pInputCache, &pDevice->pContext->allocationCallbacks); pDevice->playback.pInputCache = NULL; pDevice->playback.inputCacheCap = 0; return MA_OUT_OF_MEMORY; } pDevice->playback.pInputCache = pNewInputCache; pDevice->playback.inputCacheCap = newInputCacheCap; } else { /* Heap allocation not required. Make sure we clear out the old cache just in case this function was called in response to a route change. */ ma_free(pDevice->playback.pInputCache, &pDevice->pContext->allocationCallbacks); pDevice->playback.pInputCache = NULL; pDevice->playback.inputCacheCap = 0; } } return MA_SUCCESS; } MA_API ma_result ma_device_post_init(ma_device* pDevice, ma_device_type deviceType, const ma_device_descriptor* pDescriptorPlayback, const ma_device_descriptor* pDescriptorCapture) { ma_result result; if (pDevice == NULL) { return MA_INVALID_ARGS; } /* Capture. */ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) { if (ma_device_descriptor_is_valid(pDescriptorCapture) == MA_FALSE) { return MA_INVALID_ARGS; } pDevice->capture.internalFormat = pDescriptorCapture->format; pDevice->capture.internalChannels = pDescriptorCapture->channels; pDevice->capture.internalSampleRate = pDescriptorCapture->sampleRate; MA_COPY_MEMORY(pDevice->capture.internalChannelMap, pDescriptorCapture->channelMap, sizeof(pDescriptorCapture->channelMap)); pDevice->capture.internalPeriodSizeInFrames = pDescriptorCapture->periodSizeInFrames; pDevice->capture.internalPeriods = pDescriptorCapture->periodCount; if (pDevice->capture.internalPeriodSizeInFrames == 0) { pDevice->capture.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptorCapture->periodSizeInMilliseconds, pDescriptorCapture->sampleRate); } } /* Playback. */ if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) { if (ma_device_descriptor_is_valid(pDescriptorPlayback) == MA_FALSE) { return MA_INVALID_ARGS; } pDevice->playback.internalFormat = pDescriptorPlayback->format; pDevice->playback.internalChannels = pDescriptorPlayback->channels; pDevice->playback.internalSampleRate = pDescriptorPlayback->sampleRate; MA_COPY_MEMORY(pDevice->playback.internalChannelMap, pDescriptorPlayback->channelMap, sizeof(pDescriptorPlayback->channelMap)); pDevice->playback.internalPeriodSizeInFrames = pDescriptorPlayback->periodSizeInFrames; pDevice->playback.internalPeriods = pDescriptorPlayback->periodCount; if (pDevice->playback.internalPeriodSizeInFrames == 0) { pDevice->playback.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptorPlayback->periodSizeInMilliseconds, pDescriptorPlayback->sampleRate); } } /* The name of the device can be retrieved from device info. This may be temporary and replaced with a `ma_device_get_info(pDevice, deviceType)` instead. For loopback devices, we need to retrieve the name of the playback device. */ { ma_device_info deviceInfo; if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) { result = ma_device_get_info(pDevice, ma_device_type_capture, &deviceInfo); if (result == MA_SUCCESS) { ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), deviceInfo.name, (size_t)-1); } else { /* We failed to retrieve the device info. Fall back to a default name. */ if (pDescriptorCapture->pDeviceID == NULL) { ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); } else { ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), "Capture Device", (size_t)-1); } } } if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) { result = ma_device_get_info(pDevice, ma_device_type_playback, &deviceInfo); if (result == MA_SUCCESS) { ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), deviceInfo.name, (size_t)-1); } else {