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n8.1-dev
libswresample/swresample_internal.h
220 строк
13 KB
Andreas Rheinhardt
swresample/rematrix: Constify
03 июл 2025, 19:17
03 июл 2025, 19:17
2db60f9
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/* * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at) * * This file is part of libswresample * * libswresample is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * libswresample is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with libswresample; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #ifndef SWRESAMPLE_SWRESAMPLE_INTERNAL_H #define SWRESAMPLE_SWRESAMPLE_INTERNAL_H #include "swresample.h" #include "libavutil/channel_layout.h" #include "config.h" #define SWR_CH_MAX 64 #define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */ #define NS_TAPS 20 #if ARCH_X86_64 typedef int64_t integer; #else typedef int integer; #endif typedef void (mix_1_1_func_type)(void *out, const void *in, const void *coeffp, integer index, integer len); typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, const void *coeffp, integer index1, integer index2, integer len); typedef void (mix_any_func_type)(uint8_t *const *out, const uint8_t *const *in1, const void *coeffp, integer len); typedef struct AudioData{ uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel uint8_t *data; ///< samples buffer int ch_count; ///< number of channels int bps; ///< bytes per sample int count; ///< number of samples int planar; ///< 1 if planar audio, 0 otherwise enum AVSampleFormat fmt; ///< sample format } AudioData; struct DitherContext { int method; int noise_pos; float scale; float noise_scale; ///< Noise scale int ns_taps; ///< Noise shaping dither taps float ns_scale; ///< Noise shaping dither scale float ns_scale_1; ///< Noise shaping dither scale^-1 int ns_pos; ///< Noise shaping dither position float ns_coeffs[NS_TAPS]; ///< Noise shaping filter coefficients float ns_errors[SWR_CH_MAX][2*NS_TAPS]; AudioData noise; ///< noise used for dithering AudioData temp; ///< temporary storage when writing into the input buffer isn't possible int output_sample_bits; ///< the number of used output bits, needed to scale dither correctly }; typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby, int exact_rational); typedef void (* resample_free_func)(struct ResampleContext **c); typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed); typedef int (* resample_flush_func)(struct SwrContext *c); typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance); typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base); typedef int (* invert_initial_buffer_func)(struct ResampleContext *c, AudioData *dst, const AudioData *src, int src_size, int *dst_idx, int *dst_count); typedef int64_t (* get_out_samples_func)(struct SwrContext *s, int in_samples); struct Resampler { resample_init_func init; resample_free_func free; multiple_resample_func multiple_resample; resample_flush_func flush; set_compensation_func set_compensation; get_delay_func get_delay; invert_initial_buffer_func invert_initial_buffer; get_out_samples_func get_out_samples; }; extern struct Resampler const swri_resampler; extern struct Resampler const swri_soxr_resampler; struct SwrContext { const AVClass *av_class; ///< AVClass used for AVOption and av_log() int log_level_offset; ///< logging level offset void *log_ctx; ///< parent logging context enum AVSampleFormat in_sample_fmt; ///< input sample format enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P) enum AVSampleFormat out_sample_fmt; ///< output sample format AVChannelLayout used_ch_layout; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count) AVChannelLayout in_ch_layout; ///< input channel layout AVChannelLayout out_ch_layout; ///< output channel layout int in_sample_rate; ///< input sample rate int out_sample_rate; ///< output sample rate int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE float slev; ///< surround mixing level float clev; ///< center mixing level float lfe_mix_level; ///< LFE mixing level float rematrix_volume; ///< rematrixing volume coefficient float rematrix_maxval; ///< maximum value for rematrixing output int matrix_encoding; /**< matrixed stereo encoding */ const int *channel_map; ///< channel index (or -1 if muted channel) map int engine; AVChannelLayout user_used_chlayout; ///< User set used channel layout AVChannelLayout user_in_chlayout; ///< User set input channel layout AVChannelLayout user_out_chlayout; ///< User set output channel layout enum AVSampleFormat user_int_sample_fmt; ///< User set internal sample format int user_dither_method; ///< User set dither method struct DitherContext dither; int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */ int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */ int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */ int exact_rational; /**< if 1 then enable non power of 2 phase_count */ double cutoff; /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */ int filter_type; /**< swr resampling filter type */ double kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */ double precision; /**< soxr resampling precision (in bits) */ int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */ float min_compensation; ///< swr minimum below which no compensation will happen float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen float soft_compensation_duration; ///< swr duration over which soft compensation is applied float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration float async; ///< swr simple 1 parameter async, similar to ffmpegs -async int64_t firstpts_in_samples; ///< swr first pts in samples int resample_first; ///< 1 if resampling must come first, 0 if rematrixing int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch) int rematrix_custom; ///< flag to indicate that a custom matrix has been defined AudioData in; ///< input audio data AudioData postin; ///< post-input audio data: used for rematrix/resample AudioData midbuf; ///< intermediate audio data (postin/preout) AudioData preout; ///< pre-output audio data: used for rematrix/resample AudioData out; ///< converted output audio data AudioData in_buffer; ///< cached audio data (convert and resample purpose) AudioData silence; ///< temporary with silence AudioData drop_temp; ///< temporary used to discard output int in_buffer_index; ///< cached buffer position int in_buffer_count; ///< cached buffer length int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise int flushed; ///< 1 if data is to be flushed and no further input is expected int64_t outpts; ///< output PTS int64_t firstpts; ///< first PTS int drop_output; ///< number of output samples to drop double delayed_samples_fixup; ///< soxr 0.1.1: needed to fixup delayed_samples after flush has been called. struct AudioConvert *in_convert; ///< input conversion context struct AudioConvert *out_convert; ///< output conversion context struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output) struct ResampleContext *resample; ///< resampling context struct Resampler const *resampler; ///< resampler virtual function table double matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients float matrix_flt[SWR_CH_MAX][SWR_CH_MAX]; ///< single precision floating point rematrixing coefficients uint8_t *native_matrix; uint8_t *native_one; uint8_t *native_simd_one; uint8_t *native_simd_matrix; int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients mix_1_1_func_type *mix_1_1_f; mix_1_1_func_type *mix_1_1_simd; mix_2_1_func_type *mix_2_1_f; mix_2_1_func_type *mix_2_1_simd; mix_any_func_type *mix_any_f; /* TODO: callbacks for ASM optimizations */ }; av_warn_unused_result int swri_realloc_audio(AudioData *a, int count); void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); av_warn_unused_result int swri_rematrix_init(SwrContext *s); void swri_rematrix_free(SwrContext *s); int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy); int swri_rematrix_init_x86(struct SwrContext *s); av_warn_unused_result int swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt); av_warn_unused_result int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt); void swri_audio_convert_init_aarch64(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels); void swri_audio_convert_init_arm(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels); void swri_audio_convert_init_x86(struct AudioConvert *ac, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels); #endif