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n3.4.14
libavformat/rtpdec.c
924 строки
30 KB
Damien Riegel
rtp: rfc4175: add handler for YCbCr-4:2:2
05 апр 2017, 19:03
05 апр 2017, 19:03
01718dc
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/* * RTP input format * Copyright (c) 2002 Fabrice Bellard * * This file is part of FFmpeg. * * FFmpeg 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. * * FFmpeg 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 FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "libavutil/mathematics.h" #include "libavutil/avstring.h" #include "libavutil/intreadwrite.h" #include "libavutil/time.h" #include "avformat.h" #include "network.h" #include "srtp.h" #include "url.h" #include "rtpdec.h" #include "rtpdec_formats.h" #define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */ static RTPDynamicProtocolHandler l24_dynamic_handler = { .enc_name = "L24", .codec_type = AVMEDIA_TYPE_AUDIO, .codec_id = AV_CODEC_ID_PCM_S24BE, }; static RTPDynamicProtocolHandler gsm_dynamic_handler = { .enc_name = "GSM", .codec_type = AVMEDIA_TYPE_AUDIO, .codec_id = AV_CODEC_ID_GSM, }; static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = { .enc_name = "X-MP3-draft-00", .codec_type = AVMEDIA_TYPE_AUDIO, .codec_id = AV_CODEC_ID_MP3ADU, }; static RTPDynamicProtocolHandler speex_dynamic_handler = { .enc_name = "speex", .codec_type = AVMEDIA_TYPE_AUDIO, .codec_id = AV_CODEC_ID_SPEEX, }; static RTPDynamicProtocolHandler opus_dynamic_handler = { .enc_name = "opus", .codec_type = AVMEDIA_TYPE_AUDIO, .codec_id = AV_CODEC_ID_OPUS, }; static RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */ .enc_name = "t140", .codec_type = AVMEDIA_TYPE_SUBTITLE, .codec_id = AV_CODEC_ID_TEXT, }; static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL; void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) { handler->next = rtp_first_dynamic_payload_handler; rtp_first_dynamic_payload_handler = handler; } void ff_register_rtp_dynamic_payload_handlers(void) { ff_register_dynamic_payload_handler(&ff_ac3_dynamic_handler); ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler); ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler); ff_register_dynamic_payload_handler(&ff_dv_dynamic_handler); ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler); ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler); ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler); ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler); ff_register_dynamic_payload_handler(&ff_g726le_16_dynamic_handler); ff_register_dynamic_payload_handler(&ff_g726le_24_dynamic_handler); ff_register_dynamic_payload_handler(&ff_g726le_32_dynamic_handler); ff_register_dynamic_payload_handler(&ff_g726le_40_dynamic_handler); ff_register_dynamic_payload_handler(&ff_h261_dynamic_handler); ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler); ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler); ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler); ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler); ff_register_dynamic_payload_handler(&ff_hevc_dynamic_handler); ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler); ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler); ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler); ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler); ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler); ff_register_dynamic_payload_handler(&ff_mpeg_audio_robust_dynamic_handler); ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler); ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler); ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler); ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler); ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler); ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler); ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler); ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler); ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler); ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler); ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler); ff_register_dynamic_payload_handler(&ff_rfc4175_rtp_handler); ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler); ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler); ff_register_dynamic_payload_handler(&ff_vc2hq_dynamic_handler); ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler); ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler); ff_register_dynamic_payload_handler(&ff_vp9_dynamic_handler); ff_register_dynamic_payload_handler(&gsm_dynamic_handler); ff_register_dynamic_payload_handler(&l24_dynamic_handler); ff_register_dynamic_payload_handler(&opus_dynamic_handler); ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler); ff_register_dynamic_payload_handler(&speex_dynamic_handler); ff_register_dynamic_payload_handler(&t140_dynamic_handler); } RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name, enum AVMediaType codec_type) { RTPDynamicProtocolHandler *handler; for (handler = rtp_first_dynamic_payload_handler; handler; handler = handler->next) if (handler->enc_name && !av_strcasecmp(name, handler->enc_name) && codec_type == handler->codec_type) return handler; return NULL; } RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id, enum AVMediaType codec_type) { RTPDynamicProtocolHandler *handler; for (handler = rtp_first_dynamic_payload_handler; handler; handler = handler->next) if (handler->static_payload_id && handler->static_payload_id == id && codec_type == handler->codec_type) return handler; return NULL; } static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) { int payload_len; while (len >= 4) { payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4); switch (buf[1]) { case RTCP_SR: if (payload_len < 20) { av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n"); return AVERROR_INVALIDDATA; } s->last_rtcp_reception_time = av_gettime_relative(); s->last_rtcp_ntp_time = AV_RB64(buf + 8); s->last_rtcp_timestamp = AV_RB32(buf + 16); if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) { s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; if (!s->base_timestamp) s->base_timestamp = s->last_rtcp_timestamp; s->rtcp_ts_offset = (int32_t)(s->last_rtcp_timestamp - s->base_timestamp); } break; case RTCP_BYE: return -RTCP_BYE; } buf += payload_len; len -= payload_len; } return -1; } #define RTP_SEQ_MOD (1 << 16) static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) { memset(s, 0, sizeof(RTPStatistics)); s->max_seq = base_sequence; s->probation = 1; } /* * Called whenever there is a large jump in sequence numbers, * or when they get out of probation... */ static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) { s->max_seq = seq; s->cycles = 0; s->base_seq = seq - 1; s->bad_seq = RTP_SEQ_MOD + 1; s->received = 0; s->expected_prior = 0; s->received_prior = 0; s->jitter = 0; s->transit = 0; } /* Returns 1 if we should handle this packet. */ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) { uint16_t udelta = seq - s->max_seq; const int MAX_DROPOUT = 3000; const int MAX_MISORDER = 100; const int MIN_SEQUENTIAL = 2; /* source not valid until MIN_SEQUENTIAL packets with sequence * seq. numbers have been received */ if (s->probation) { if (seq == s->max_seq + 1) { s->probation--; s->max_seq = seq; if (s->probation == 0) { rtp_init_sequence(s, seq); s->received++; return 1; } } else { s->probation = MIN_SEQUENTIAL - 1; s->max_seq = seq; } } else if (udelta < MAX_DROPOUT) { // in order, with permissible gap if (seq < s->max_seq) { // sequence number wrapped; count another 64k cycles s->cycles += RTP_SEQ_MOD; } s->max_seq = seq; } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { // sequence made a large jump... if (seq == s->bad_seq) { /* two sequential packets -- assume that the other side * restarted without telling us; just resync. */ rtp_init_sequence(s, seq); } else { s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1); return 0; } } else { // duplicate or reordered packet... } s->received++; return 1; } static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp) { // Most of this is pretty straight from RFC 3550 appendix A.8 uint32_t transit = arrival_timestamp - sent_timestamp; uint32_t prev_transit = s->transit; int32_t d = transit - prev_transit; // Doing the FFABS() call directly on the "transit - prev_transit" // expression doesn't work, since it's an unsigned expression. Doing the // transit calculation in unsigned is desired though, since it most // probably will need to wrap around. d = FFABS(d); s->transit = transit; if (!prev_transit) return; s->jitter += d - (int32_t) ((s->jitter + 8) >> 4); } int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio, int count) { AVIOContext *pb; uint8_t *buf; int len; int rtcp_bytes; RTPStatistics *stats = &s->statistics; uint32_t lost; uint32_t extended_max; uint32_t expected_interval; uint32_t received_interval; int32_t lost_interval; uint32_t expected; uint32_t fraction; if ((!fd && !avio) || (count < 1)) return -1; /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */ s->octet_count += count; rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / RTCP_TX_RATIO_DEN; rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? if (rtcp_bytes < 28) return -1; s->last_octet_count = s->octet_count; if (!fd) pb = avio; else if (avio_open_dyn_buf(&pb) < 0) return -1; // Receiver Report avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ avio_w8(pb, RTCP_RR); avio_wb16(pb, 7); /* length in words - 1 */ // our own SSRC: we use the server's SSRC + 1 to avoid conflicts avio_wb32(pb, s->ssrc + 1); avio_wb32(pb, s->ssrc); // server SSRC // some placeholders we should really fill... // RFC 1889/p64 extended_max = stats->cycles + stats->max_seq; expected = extended_max - stats->base_seq; lost = expected - stats->received; lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... expected_interval = expected - stats->expected_prior; stats->expected_prior = expected; received_interval = stats->received - stats->received_prior; stats->received_prior = stats->received; lost_interval = expected_interval - received_interval; if (expected_interval == 0 || lost_interval <= 0) fraction = 0; else fraction = (lost_interval << 8) / expected_interval; fraction = (fraction << 24) | lost; avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ avio_wb32(pb, extended_max); /* max sequence received */ avio_wb32(pb, stats->jitter >> 4); /* jitter */ if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) { avio_wb32(pb, 0); /* last SR timestamp */ avio_wb32(pb, 0); /* delay since last SR */ } else { uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special? uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time, 65536, AV_TIME_BASE); avio_wb32(pb, middle_32_bits); /* last SR timestamp */ avio_wb32(pb, delay_since_last); /* delay since last SR */ } // CNAME avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ avio_w8(pb, RTCP_SDES); len = strlen(s->hostname); avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */ avio_wb32(pb, s->ssrc + 1); avio_w8(pb, 0x01); avio_w8(pb, len); avio_write(pb, s->hostname, len); avio_w8(pb, 0); /* END */ // padding for (len = (7 + len) % 4; len % 4; len++) avio_w8(pb, 0); avio_flush(pb); if (!fd) return 0; len = avio_close_dyn_buf(pb, &buf); if ((len > 0) && buf) { int av_unused result; av_log(s->ic, AV_LOG_TRACE, "sending %d bytes of RR\n", len); result = ffurl_write(fd, buf, len); av_log(s->ic, AV_LOG_TRACE, "result from ffurl_write: %d\n", result); av_free(buf); } return 0; } void ff_rtp_send_punch_packets(URLContext *rtp_handle) { AVIOContext *pb; uint8_t *buf; int len; /* Send a small RTP packet */ if (avio_open_dyn_buf(&pb) < 0) return; avio_w8(pb, (RTP_VERSION << 6)); avio_w8(pb, 0); /* Payload type */ avio_wb16(pb, 0); /* Seq */ avio_wb32(pb, 0); /* Timestamp */ avio_wb32(pb, 0); /* SSRC */ avio_flush(pb); len = avio_close_dyn_buf(pb, &buf); if ((len > 0) && buf) ffurl_write(rtp_handle, buf, len); av_free(buf); /* Send a minimal RTCP RR */ if (avio_open_dyn_buf(&pb) < 0) return; avio_w8(pb, (RTP_VERSION << 6)); avio_w8(pb, RTCP_RR); /* receiver report */ avio_wb16(pb, 1); /* length in words - 1 */ avio_wb32(pb, 0); /* our own SSRC */ avio_flush(pb); len = avio_close_dyn_buf(pb, &buf); if ((len > 0) && buf) ffurl_write(rtp_handle, buf, len); av_free(buf); } static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing, uint16_t *missing_mask) { int i; uint16_t next_seq = s->seq + 1; RTPPacket *pkt = s->queue; if (!pkt || pkt->seq == next_seq) return 0; *missing_mask = 0; for (i = 1; i <= 16; i++) { uint16_t missing_seq = next_seq + i; while (pkt) { int16_t diff = pkt->seq - missing_seq; if (diff >= 0) break; pkt = pkt->next; } if (!pkt) break; if (pkt->seq == missing_seq) continue; *missing_mask |= 1 << (i - 1); } *first_missing = next_seq; return 1; } int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, AVIOContext *avio) { int len, need_keyframe, missing_packets; AVIOContext *pb; uint8_t *buf; int64_t now; uint16_t first_missing = 0, missing_mask = 0; if (!fd && !avio) return -1; need_keyframe = s->handler && s->handler->need_keyframe && s->handler->need_keyframe(s->dynamic_protocol_context); missing_packets = find_missing_packets(s, &first_missing, &missing_mask); if (!need_keyframe && !missing_packets) return 0; /* Send new feedback if enough time has elapsed since the last * feedback packet. */ now = av_gettime_relative(); if (s->last_feedback_time && (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL) return 0; s->last_feedback_time = now; if (!fd) pb = avio; else if (avio_open_dyn_buf(&pb) < 0) return -1; if (need_keyframe) { avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */ avio_w8(pb, RTCP_PSFB); avio_wb16(pb, 2); /* length in words - 1 */ // our own SSRC: we use the server's SSRC + 1 to avoid conflicts avio_wb32(pb, s->ssrc + 1); avio_wb32(pb, s->ssrc); // server SSRC } if (missing_packets) { avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */ avio_w8(pb, RTCP_RTPFB); avio_wb16(pb, 3); /* length in words - 1 */ avio_wb32(pb, s->ssrc + 1); avio_wb32(pb, s->ssrc); // server SSRC avio_wb16(pb, first_missing); avio_wb16(pb, missing_mask); } avio_flush(pb); if (!fd) return 0; len = avio_close_dyn_buf(pb, &buf); if (len > 0 && buf) { ffurl_write(fd, buf, len); av_free(buf); } return 0; } /** * open a new RTP parse context for stream 'st'. 'st' can be NULL for * MPEG-2 TS streams. */ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, int queue_size) { RTPDemuxContext *s; s = av_mallocz(sizeof(RTPDemuxContext)); if (!s) return NULL; s->payload_type = payload_type; s->last_rtcp_ntp_time = AV_NOPTS_VALUE; s->first_rtcp_ntp_time = AV_NOPTS_VALUE; s->ic = s1; s->st = st; s->queue_size = queue_size; av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n", s->queue_size); rtp_init_statistics(&s->statistics, 0); if (st) { switch (st->codecpar->codec_id) { case AV_CODEC_ID_ADPCM_G722: /* According to RFC 3551, the stream clock rate is 8000 * even if the sample rate is 16000. */ if (st->codecpar->sample_rate == 8000) st->codecpar->sample_rate = 16000; break; default: break; } } // needed to send back RTCP RR in RTSP sessions gethostname(s->hostname, sizeof(s->hostname)); return s; } void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler) { s->dynamic_protocol_context = ctx; s->handler = handler; } void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, const char *params) { if (!ff_srtp_set_crypto(&s->srtp, suite, params)) s->srtp_enabled = 1; } /** * This was the second switch in rtp_parse packet. * Normalizes time, if required, sets stream_index, etc. */ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) { if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE) return; /* Timestamp already set by depacketizer */ if (timestamp == RTP_NOTS_VALUE) return; if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) { int64_t addend; int delta_timestamp; /* compute pts from timestamp with received ntp_time */ delta_timestamp = timestamp - s->last_rtcp_timestamp; /* convert to the PTS timebase */ addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t) s->st->time_base.num << 32); pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend + delta_timestamp; return; } if (!s->base_timestamp) s->base_timestamp = timestamp; /* assume that the difference is INT32_MIN < x < INT32_MAX, * but allow the first timestamp to exceed INT32_MAX */ if (!s->timestamp) s->unwrapped_timestamp += timestamp; else s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp); s->timestamp = timestamp; pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp; } static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, const uint8_t *buf, int len) { unsigned int ssrc; int payload_type, seq, flags = 0; int ext, csrc; AVStream *st; uint32_t timestamp; int rv = 0; csrc = buf[0] & 0x0f; ext = buf[0] & 0x10; payload_type = buf[1] & 0x7f; if (buf[1] & 0x80) flags |= RTP_FLAG_MARKER; seq = AV_RB16(buf + 2); timestamp = AV_RB32(buf + 4); ssrc = AV_RB32(buf + 8); /* store the ssrc in the RTPDemuxContext */ s->ssrc = ssrc; /* NOTE: we can handle only one payload type */ if (s->payload_type != payload_type) return -1; st = s->st; // only do something with this if all the rtp checks pass... if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) { av_log(s->ic, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", payload_type, seq, ((s->seq + 1) & 0xffff)); return -1; } if (buf[0] & 0x20) { int padding = buf[len - 1]; if (len >= 12 + padding) len -= padding; } s->seq = seq; len -= 12; buf += 12; len -= 4 * csrc; buf += 4 * csrc; if (len < 0) return AVERROR_INVALIDDATA; /* RFC 3550 Section 5.3.1 RTP Header Extension handling */ if (ext) { if (len < 4) return -1; /* calculate the header extension length (stored as number * of 32-bit words) */ ext = (AV_RB16(buf + 2) + 1) << 2; if (len < ext) return -1; // skip past RTP header extension len -= ext; buf += ext; } if (s->handler && s->handler->parse_packet) { rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context, s->st, pkt, ×tamp, buf, len, seq, flags); } else if (st) { if ((rv = av_new_packet(pkt, len)) < 0) return rv; memcpy(pkt->data, buf, len); pkt->stream_index = st->index; } else { return AVERROR(EINVAL); } // now perform timestamp things.... finalize_packet(s, pkt, timestamp); return rv; } void ff_rtp_reset_packet_queue(RTPDemuxContext *s) { while (s->queue) { RTPPacket *next = s->queue->next; av_freep(&s->queue->buf); av_freep(&s->queue); s->queue = next; } s->seq = 0; s->queue_len = 0; s->prev_ret = 0; } static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len) { uint16_t seq = AV_RB16(buf + 2); RTPPacket **cur = &s->queue, *packet; /* Find the correct place in the queue to insert the packet */ while (*cur) { int16_t diff = seq - (*cur)->seq; if (diff < 0) break; cur = &(*cur)->next; } packet = av_mallocz(sizeof(*packet)); if (!packet) return AVERROR(ENOMEM); packet->recvtime = av_gettime_relative(); packet->seq = seq; packet->len = len; packet->buf = buf; packet->next = *cur; *cur = packet; s->queue_len++; return 0; } static int has_next_packet(RTPDemuxContext *s) { return s->queue && s->queue->seq == (uint16_t) (s->seq + 1); } int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s) { return s->queue ? s->queue->recvtime : 0; } static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt) { int rv; RTPPacket *next; if (s->queue_len <= 0) return -1; if (!has_next_packet(s)) av_log(s->ic, AV_LOG_WARNING, "RTP: missed %d packets\n", s->queue->seq - s->seq - 1); /* Parse the first packet in the queue, and dequeue it */ rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len); next = s->queue->next; av_freep(&s->queue->buf); av_freep(&s->queue); s->queue = next; s->queue_len--; return rv; } static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len) { uint8_t *buf = bufptr ? *bufptr : NULL; int flags = 0; uint32_t timestamp; int rv = 0; if (!buf) { /* If parsing of the previous packet actually returned 0 or an error, * there's nothing more to be parsed from that packet, but we may have * indicated that we can return the next enqueued packet. */ if (s->prev_ret <= 0) return rtp_parse_queued_packet(s, pkt); /* return the next packets, if any */ if (s->handler && s->handler->parse_packet) { /* timestamp should be overwritten by parse_packet, if not, * the packet is left with pts == AV_NOPTS_VALUE */ timestamp = RTP_NOTS_VALUE; rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context, s->st, pkt, ×tamp, NULL, 0, 0, flags); finalize_packet(s, pkt, timestamp); return rv; } } if (len < 12) return -1; if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) return -1; if (RTP_PT_IS_RTCP(buf[1])) { return rtcp_parse_packet(s, buf, len); } if (s->st) { int64_t received = av_gettime_relative(); uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q, s->st->time_base); timestamp = AV_RB32(buf + 4); // Calculate the jitter immediately, before queueing the packet // into the reordering queue. rtcp_update_jitter(&s->statistics, timestamp, arrival_ts); } if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) { /* First packet, or no reordering */ return rtp_parse_packet_internal(s, pkt, buf, len); } else { uint16_t seq = AV_RB16(buf + 2); int16_t diff = seq - s->seq; if (diff < 0) { /* Packet older than the previously emitted one, drop */ av_log(s->ic, AV_LOG_WARNING, "RTP: dropping old packet received too late\n"); return -1; } else if (diff <= 1) { /* Correct packet */ rv = rtp_parse_packet_internal(s, pkt, buf, len); return rv; } else { /* Still missing some packet, enqueue this one. */ rv = enqueue_packet(s, buf, len); if (rv < 0) return rv; *bufptr = NULL; /* Return the first enqueued packet if the queue is full, * even if we're missing something */ if (s->queue_len >= s->queue_size) { av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n"); return rtp_parse_queued_packet(s, pkt); } return -1; } } } /** * Parse an RTP or RTCP packet directly sent as a buffer. * @param s RTP parse context. * @param pkt returned packet * @param bufptr pointer to the input buffer or NULL to read the next packets * @param len buffer len * @return 0 if a packet is returned, 1 if a packet is returned and more can follow * (use buf as NULL to read the next). -1 if no packet (error or no more packet). */ int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len) { int rv; if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0) return -1; rv = rtp_parse_one_packet(s, pkt, bufptr, len); s->prev_ret = rv; while (rv < 0 && has_next_packet(s)) rv = rtp_parse_queued_packet(s, pkt); return rv ? rv : has_next_packet(s); } void ff_rtp_parse_close(RTPDemuxContext *s) { ff_rtp_reset_packet_queue(s); ff_srtp_free(&s->srtp); av_free(s); } int ff_parse_fmtp(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *p, int (*parse_fmtp)(AVFormatContext *s, AVStream *stream, PayloadContext *data, const char *attr, const char *value)) { char attr[256]; char *value; int res; int value_size = strlen(p) + 1; if (!(value = av_malloc(value_size))) { av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n"); return AVERROR(ENOMEM); } // remove protocol identifier while (*p && *p == ' ') p++; // strip spaces while (*p && *p != ' ') p++; // eat protocol identifier while (*p && *p == ' ') p++; // strip trailing spaces while (ff_rtsp_next_attr_and_value(&p, attr, sizeof(attr), value, value_size)) { res = parse_fmtp(s, stream, data, attr, value); if (res < 0 && res != AVERROR_PATCHWELCOME) { av_free(value); return res; } } av_free(value); return 0; } int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx) { int ret; av_init_packet(pkt); pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data); pkt->stream_index = stream_idx; *dyn_buf = NULL; if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) { av_freep(&pkt->data); return ret; } return pkt->size; }