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Support composited bridges for 1:N protocols converting. v6.0.41 (#3392)
Co-authored-by: john <hondaxiao@tencent.com> Co-authored-by: chundonglinlin <chundonglinlin@163.com>
This commit is contained in:
Winlin
GitHub
parent
771ae0a1a6
commit
dcd02fe69c
19 changed files with 770 additions and 400 deletions
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@ -52,27 +52,23 @@ SrsPps* _srs_pps_rmnack = NULL;
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extern SrsPps* _srs_pps_aloss2;
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// Firefox defaults as 109, Chrome is 111.
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const int kAudioPayloadType = 111;
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const int kAudioChannel = 2;
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const int kAudioSamplerate = 48000;
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// Firefox defaults as 126, Chrome is 102.
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const int kVideoPayloadType = 102;
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const int kVideoSamplerate = 90000;
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using namespace std;
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// The RTP payload max size, reserved some paddings for SRTP as such:
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// kRtpPacketSize = kRtpMaxPayloadSize + paddings
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// For example, if kRtpPacketSize is 1500, recommend to set kRtpMaxPayloadSize to 1400,
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// which reserves 100 bytes for SRTP or paddings.
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// otherwise, the kRtpPacketSize must less than MTU, in webrtc source code,
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// otherwise, the kRtpPacketSize must less than MTU, in webrtc source code,
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// the rtp max size is assigned by kVideoMtu = 1200.
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// so we set kRtpMaxPayloadSize = 1200.
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// see @doc https://groups.google.com/g/discuss-webrtc/c/gH5ysR3SoZI
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const int kRtpMaxPayloadSize = kRtpPacketSize - 300;
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using namespace std;
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// TODO: Add this function into SrsRtpMux class.
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srs_error_t aac_raw_append_adts_header(SrsSharedPtrMessage* shared_audio, SrsFormat* format, char** pbuf, int* pnn_buf)
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{
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@ -323,14 +319,6 @@ ISrsRtcSourceEventHandler::~ISrsRtcSourceEventHandler()
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{
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}
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ISrsRtcSourceBridge::ISrsRtcSourceBridge()
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{
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}
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ISrsRtcSourceBridge::~ISrsRtcSourceBridge()
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{
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}
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SrsRtcSource::SrsRtcSource()
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{
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is_created_ = false;
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@ -341,6 +329,9 @@ SrsRtcSource::SrsRtcSource()
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req = NULL;
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bridge_ = NULL;
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#ifdef SRS_FFMPEG_FIT
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frame_builder_ = NULL;
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#endif
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pli_for_rtmp_ = pli_elapsed_ = 0;
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}
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@ -351,6 +342,9 @@ SrsRtcSource::~SrsRtcSource()
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// for all consumers are auto free.
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consumers.clear();
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#ifdef SRS_FFMPEG_FIT
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srs_freep(frame_builder_);
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#endif
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srs_freep(bridge_);
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srs_freep(req);
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srs_freep(stream_desc_);
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@ -464,10 +458,15 @@ SrsContextId SrsRtcSource::pre_source_id()
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return _pre_source_id;
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}
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void SrsRtcSource::set_bridge(ISrsRtcSourceBridge *bridge)
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void SrsRtcSource::set_bridge(ISrsStreamBridge* bridge)
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{
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srs_freep(bridge_);
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bridge_ = bridge;
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#ifdef SRS_FFMPEG_FIT
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srs_freep(frame_builder_);
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frame_builder_ = new SrsRtcFrameBuilder(bridge);
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#endif
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}
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srs_error_t SrsRtcSource::create_consumer(SrsRtcConsumer*& consumer)
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@ -541,6 +540,16 @@ srs_error_t SrsRtcSource::on_publish()
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// If bridge to other source, handle event and start timer to request PLI.
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if (bridge_) {
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#ifdef SRS_FFMPEG_FIT
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if ((err = frame_builder_->initialize(req)) != srs_success) {
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return srs_error_wrap(err, "frame builder initialize");
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}
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if ((err = frame_builder_->on_publish()) != srs_success) {
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return srs_error_wrap(err, "frame builder on publish");
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}
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#endif
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if ((err = bridge_->on_publish()) != srs_success) {
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return srs_error_wrap(err, "bridge on publish");
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}
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@ -585,6 +594,11 @@ void SrsRtcSource::on_unpublish()
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// For SrsRtcSource::on_timer()
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_srs_hybrid->timer100ms()->unsubscribe(this);
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#ifdef SRS_FFMPEG_FIT
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frame_builder_->on_unpublish();
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srs_freep(frame_builder_);
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#endif
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bridge_->on_unpublish();
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srs_freep(bridge_);
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}
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@ -636,9 +650,11 @@ srs_error_t SrsRtcSource::on_rtp(SrsRtpPacket* pkt)
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}
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}
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if (bridge_ && (err = bridge_->on_rtp(pkt)) != srs_success) {
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return srs_error_wrap(err, "bridge consume message");
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#ifdef SRS_FFMPEG_FIT
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if (frame_builder_ && (err = frame_builder_->on_rtp(pkt)) != srs_success) {
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return srs_error_wrap(err, "frame builder consume packet");
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}
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#endif
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return err;
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}
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@ -712,85 +728,56 @@ srs_error_t SrsRtcSource::on_timer(srs_utime_t interval)
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#ifdef SRS_FFMPEG_FIT
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SrsRtcFromRtmpBridge::SrsRtcFromRtmpBridge(SrsRtcSource* source)
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SrsRtcRtpBuilder::SrsRtcRtpBuilder(SrsFrameToRtcBridge* bridge, uint32_t assrc, uint8_t apt, uint32_t vssrc, uint8_t vpt)
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{
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req = NULL;
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source_ = source;
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bridge_ = bridge;
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format = new SrsRtmpFormat();
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codec_ = new SrsAudioTranscoder();
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latest_codec_ = SrsAudioCodecIdForbidden;
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rtmp_to_rtc = false;
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keep_bframe = false;
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merge_nalus = false;
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meta = new SrsMetaCache();
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audio_sequence = 0;
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video_sequence = 0;
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// audio track ssrc
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if (true) {
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std::vector<SrsRtcTrackDescription*> descs = source->get_track_desc("audio", "opus");
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if (!descs.empty()) {
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audio_ssrc = descs.at(0)->ssrc_;
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}
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// Note we must use the PT of source, see https://github.com/ossrs/srs/pull/3079
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audio_payload_type_ = descs.empty() ? kAudioPayloadType : descs.front()->media_->pt_;
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}
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// video track ssrc
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if (true) {
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std::vector<SrsRtcTrackDescription*> descs = source->get_track_desc("video", "H264");
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if (!descs.empty()) {
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video_ssrc = descs.at(0)->ssrc_;
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}
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// Note we must use the PT of source, see https://github.com/ossrs/srs/pull/3079
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video_payload_type_ = descs.empty() ? kVideoPayloadType : descs.front()->media_->pt_;
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}
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audio_ssrc_ = assrc;
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audio_payload_type_ = apt;
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video_ssrc_ = vssrc;
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video_payload_type_ = vpt;
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}
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SrsRtcFromRtmpBridge::~SrsRtcFromRtmpBridge()
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SrsRtcRtpBuilder::~SrsRtcRtpBuilder()
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{
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srs_freep(format);
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srs_freep(codec_);
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srs_freep(meta);
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}
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srs_error_t SrsRtcFromRtmpBridge::initialize(SrsRequest* r)
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srs_error_t SrsRtcRtpBuilder::initialize(SrsRequest* r)
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{
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srs_error_t err = srs_success;
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req = r;
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rtmp_to_rtc = _srs_config->get_rtc_from_rtmp(req->vhost);
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if (rtmp_to_rtc) {
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if ((err = format->initialize()) != srs_success) {
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return srs_error_wrap(err, "format initialize");
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}
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// Setup the SPS/PPS parsing strategy.
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format->try_annexb_first = _srs_config->try_annexb_first(r->vhost);
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if ((err = format->initialize()) != srs_success) {
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return srs_error_wrap(err, "format initialize");
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}
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// Setup the SPS/PPS parsing strategy.
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format->try_annexb_first = _srs_config->try_annexb_first(r->vhost);
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keep_bframe = _srs_config->get_rtc_keep_bframe(req->vhost);
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merge_nalus = _srs_config->get_rtc_server_merge_nalus();
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srs_trace("RTC bridge from RTMP, rtmp2rtc=%d, keep_bframe=%d, merge_nalus=%d",
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rtmp_to_rtc, keep_bframe, merge_nalus);
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srs_trace("RTC bridge from RTMP, keep_bframe=%d, merge_nalus=%d", keep_bframe, merge_nalus);
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return err;
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}
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srs_error_t SrsRtcFromRtmpBridge::on_publish()
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srs_error_t SrsRtcRtpBuilder::on_publish()
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{
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srs_error_t err = srs_success;
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if (!rtmp_to_rtc) {
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return err;
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}
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// TODO: FIXME: Should sync with bridge?
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if ((err = source_->on_publish()) != srs_success) {
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return srs_error_wrap(err, "source publish");
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}
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// Reset the metadata cache, to make VLC happy when disable/enable stream.
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// @see https://github.com/ossrs/srs/issues/1630#issuecomment-597979448
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meta->clear();
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@ -798,30 +785,28 @@ srs_error_t SrsRtcFromRtmpBridge::on_publish()
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return err;
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}
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void SrsRtcFromRtmpBridge::on_unpublish()
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void SrsRtcRtpBuilder::on_unpublish()
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{
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if (!rtmp_to_rtc) {
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return;
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}
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// Reset the metadata cache, to make VLC happy when disable/enable stream.
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// @see https://github.com/ossrs/srs/issues/1630#issuecomment-597979448
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meta->update_previous_vsh();
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meta->update_previous_ash();
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// @remark This bridge might be disposed here, so never use it.
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// TODO: FIXME: Should sync with bridge?
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source_->on_unpublish();
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}
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srs_error_t SrsRtcFromRtmpBridge::on_audio(SrsSharedPtrMessage* msg)
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srs_error_t SrsRtcRtpBuilder::on_frame(SrsSharedPtrMessage* frame)
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{
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if (frame->is_audio()) {
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return on_audio(frame);
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} else if (frame->is_video()) {
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return on_video(frame);
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}
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return srs_success;
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}
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srs_error_t SrsRtcRtpBuilder::on_audio(SrsSharedPtrMessage* msg)
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{
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srs_error_t err = srs_success;
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if (!rtmp_to_rtc) {
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return err;
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}
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// TODO: FIXME: Support parsing OPUS for RTC.
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if ((err = format->on_audio(msg)) != srs_success) {
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return srs_error_wrap(err, "format consume audio");
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@ -880,7 +865,7 @@ srs_error_t SrsRtcFromRtmpBridge::on_audio(SrsSharedPtrMessage* msg)
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return err;
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}
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srs_error_t SrsRtcFromRtmpBridge::init_codec(SrsAudioCodecId codec)
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srs_error_t SrsRtcRtpBuilder::init_codec(SrsAudioCodecId codec)
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{
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srs_error_t err = srs_success;
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@ -902,18 +887,18 @@ srs_error_t SrsRtcFromRtmpBridge::init_codec(SrsAudioCodecId codec)
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srs_trace("RTMP2RTC: Init audio codec to %d(%s)", codec, srs_audio_codec_id2str(codec).c_str());
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} else {
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srs_trace("RTMP2RTC: Switch audio codec %d(%s) to %d(%s)", latest_codec_, srs_audio_codec_id2str(latest_codec_).c_str(),
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codec, srs_audio_codec_id2str(codec).c_str());
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codec, srs_audio_codec_id2str(codec).c_str());
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}
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latest_codec_ = codec;
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return err;
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}
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srs_error_t SrsRtcFromRtmpBridge::transcode(SrsAudioFrame* audio)
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srs_error_t SrsRtcRtpBuilder::transcode(SrsAudioFrame* audio)
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{
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srs_error_t err = srs_success;
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std::vector<SrsAudioFrame *> out_audios;
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std::vector<SrsAudioFrame*> out_audios;
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if ((err = codec_->transcode(audio, out_audios)) != srs_success) {
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return srs_error_wrap(err, "recode error");
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}
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@ -934,7 +919,7 @@ srs_error_t SrsRtcFromRtmpBridge::transcode(SrsAudioFrame* audio)
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break;
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}
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if ((err = source_->on_rtp(pkt)) != srs_success) {
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if ((err = bridge_->on_rtp(pkt)) != srs_success) {
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err = srs_error_wrap(err, "consume opus");
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break;
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}
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@ -945,12 +930,12 @@ srs_error_t SrsRtcFromRtmpBridge::transcode(SrsAudioFrame* audio)
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return err;
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}
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srs_error_t SrsRtcFromRtmpBridge::package_opus(SrsAudioFrame* audio, SrsRtpPacket* pkt)
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srs_error_t SrsRtcRtpBuilder::package_opus(SrsAudioFrame* audio, SrsRtpPacket* pkt)
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{
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srs_error_t err = srs_success;
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pkt->header.set_payload_type(audio_payload_type_);
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pkt->header.set_ssrc(audio_ssrc);
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pkt->header.set_ssrc(audio_ssrc_);
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pkt->frame_type = SrsFrameTypeAudio;
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pkt->header.set_marker(true);
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pkt->header.set_sequence(audio_sequence++);
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@ -966,21 +951,10 @@ srs_error_t SrsRtcFromRtmpBridge::package_opus(SrsAudioFrame* audio, SrsRtpPacke
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return err;
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}
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srs_error_t SrsRtcFromRtmpBridge::on_video(SrsSharedPtrMessage* msg)
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srs_error_t SrsRtcRtpBuilder::on_video(SrsSharedPtrMessage* msg)
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{
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srs_error_t err = srs_success;
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if (!rtmp_to_rtc) {
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return err;
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}
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// WebRTC NOT support HEVC.
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#ifdef SRS_H265
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if (format->vcodec->id == SrsVideoCodecIdHEVC) {
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return err;
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}
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#endif
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// cache the sequence header if h264
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bool is_sequence_header = SrsFlvVideo::sh(msg->payload, msg->size);
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if (is_sequence_header && (err = meta->update_vsh(msg)) != srs_success) {
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@ -997,6 +971,13 @@ srs_error_t SrsRtcFromRtmpBridge::on_video(SrsSharedPtrMessage* msg)
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return err;
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}
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// WebRTC does NOT support HEVC.
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#ifdef SRS_H265
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if (format->vcodec->id == SrsVideoCodecIdHEVC) {
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return err;
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}
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#endif
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bool has_idr = false;
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vector<SrsSample*> samples;
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if ((err = filter(msg, format, has_idr, samples)) != srs_success) {
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@ -1009,11 +990,11 @@ srs_error_t SrsRtcFromRtmpBridge::on_video(SrsSharedPtrMessage* msg)
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SrsRtpPacket* pkt = new SrsRtpPacket();
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SrsAutoFree(SrsRtpPacket, pkt);
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if ((err = package_stap_a(source_, msg, pkt)) != srs_success) {
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if ((err = package_stap_a(msg, pkt)) != srs_success) {
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return srs_error_wrap(err, "package stap-a");
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}
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if ((err = source_->on_rtp(pkt)) != srs_success) {
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if ((err = bridge_->on_rtp(pkt)) != srs_success) {
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return srs_error_wrap(err, "consume sps/pps");
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}
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}
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@ -1054,7 +1035,7 @@ srs_error_t SrsRtcFromRtmpBridge::on_video(SrsSharedPtrMessage* msg)
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return consume_packets(pkts);
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}
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srs_error_t SrsRtcFromRtmpBridge::filter(SrsSharedPtrMessage* msg, SrsFormat* format, bool& has_idr, vector<SrsSample*>& samples)
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srs_error_t SrsRtcRtpBuilder::filter(SrsSharedPtrMessage* msg, SrsFormat* format, bool& has_idr, vector<SrsSample*>& samples)
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{
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srs_error_t err = srs_success;
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@ -1084,7 +1065,7 @@ srs_error_t SrsRtcFromRtmpBridge::filter(SrsSharedPtrMessage* msg, SrsFormat* fo
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return err;
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}
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srs_error_t SrsRtcFromRtmpBridge::package_stap_a(SrsRtcSource* source, SrsSharedPtrMessage* msg, SrsRtpPacket* pkt)
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srs_error_t SrsRtcRtpBuilder::package_stap_a(SrsSharedPtrMessage* msg, SrsRtpPacket* pkt)
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{
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srs_error_t err = srs_success;
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@ -1101,7 +1082,7 @@ srs_error_t SrsRtcFromRtmpBridge::package_stap_a(SrsRtcSource* source, SrsShared
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}
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pkt->header.set_payload_type(video_payload_type_);
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pkt->header.set_ssrc(video_ssrc);
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pkt->header.set_ssrc(video_ssrc_);
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pkt->frame_type = SrsFrameTypeVideo;
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pkt->nalu_type = (SrsAvcNaluType)kStapA;
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pkt->header.set_marker(false);
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@ -1143,7 +1124,7 @@ srs_error_t SrsRtcFromRtmpBridge::package_stap_a(SrsRtcSource* source, SrsShared
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return err;
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}
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||||
|
||||
srs_error_t SrsRtcFromRtmpBridge::package_nalus(SrsSharedPtrMessage* msg, const vector<SrsSample*>& samples, vector<SrsRtpPacket*>& pkts)
|
||||
srs_error_t SrsRtcRtpBuilder::package_nalus(SrsSharedPtrMessage* msg, const vector<SrsSample*>& samples, vector<SrsRtpPacket*>& pkts)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
|
|
@ -1183,7 +1164,7 @@ srs_error_t SrsRtcFromRtmpBridge::package_nalus(SrsSharedPtrMessage* msg, const
|
|||
pkts.push_back(pkt);
|
||||
|
||||
pkt->header.set_payload_type(video_payload_type_);
|
||||
pkt->header.set_ssrc(video_ssrc);
|
||||
pkt->header.set_ssrc(video_ssrc_);
|
||||
pkt->frame_type = SrsFrameTypeVideo;
|
||||
pkt->nalu_type = (SrsAvcNaluType)first_nalu_type;
|
||||
pkt->header.set_sequence(video_sequence++);
|
||||
|
|
@ -1217,7 +1198,7 @@ srs_error_t SrsRtcFromRtmpBridge::package_nalus(SrsSharedPtrMessage* msg, const
|
|||
pkts.push_back(pkt);
|
||||
|
||||
pkt->header.set_payload_type(video_payload_type_);
|
||||
pkt->header.set_ssrc(video_ssrc);
|
||||
pkt->header.set_ssrc(video_ssrc_);
|
||||
pkt->frame_type = SrsFrameTypeVideo;
|
||||
pkt->nalu_type = (SrsAvcNaluType)kFuA;
|
||||
pkt->header.set_sequence(video_sequence++);
|
||||
|
|
@ -1239,7 +1220,7 @@ srs_error_t SrsRtcFromRtmpBridge::package_nalus(SrsSharedPtrMessage* msg, const
|
|||
}
|
||||
|
||||
// Single NAL Unit Packet @see https://tools.ietf.org/html/rfc6184#section-5.6
|
||||
srs_error_t SrsRtcFromRtmpBridge::package_single_nalu(SrsSharedPtrMessage* msg, SrsSample* sample, vector<SrsRtpPacket*>& pkts)
|
||||
srs_error_t SrsRtcRtpBuilder::package_single_nalu(SrsSharedPtrMessage* msg, SrsSample* sample, vector<SrsRtpPacket*>& pkts)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
|
|
@ -1247,7 +1228,7 @@ srs_error_t SrsRtcFromRtmpBridge::package_single_nalu(SrsSharedPtrMessage* msg,
|
|||
pkts.push_back(pkt);
|
||||
|
||||
pkt->header.set_payload_type(video_payload_type_);
|
||||
pkt->header.set_ssrc(video_ssrc);
|
||||
pkt->header.set_ssrc(video_ssrc_);
|
||||
pkt->frame_type = SrsFrameTypeVideo;
|
||||
pkt->header.set_sequence(video_sequence++);
|
||||
pkt->header.set_timestamp(msg->timestamp * 90);
|
||||
|
|
@ -1263,7 +1244,7 @@ srs_error_t SrsRtcFromRtmpBridge::package_single_nalu(SrsSharedPtrMessage* msg,
|
|||
return err;
|
||||
}
|
||||
|
||||
srs_error_t SrsRtcFromRtmpBridge::package_fu_a(SrsSharedPtrMessage* msg, SrsSample* sample, int fu_payload_size, vector<SrsRtpPacket*>& pkts)
|
||||
srs_error_t SrsRtcRtpBuilder::package_fu_a(SrsSharedPtrMessage* msg, SrsSample* sample, int fu_payload_size, vector<SrsRtpPacket*>& pkts)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
|
|
@ -1280,7 +1261,7 @@ srs_error_t SrsRtcFromRtmpBridge::package_fu_a(SrsSharedPtrMessage* msg, SrsSamp
|
|||
pkts.push_back(pkt);
|
||||
|
||||
pkt->header.set_payload_type(video_payload_type_);
|
||||
pkt->header.set_ssrc(video_ssrc);
|
||||
pkt->header.set_ssrc(video_ssrc_);
|
||||
pkt->frame_type = SrsFrameTypeVideo;
|
||||
pkt->header.set_sequence(video_sequence++);
|
||||
pkt->header.set_timestamp(msg->timestamp * 90);
|
||||
|
|
@ -1305,14 +1286,14 @@ srs_error_t SrsRtcFromRtmpBridge::package_fu_a(SrsSharedPtrMessage* msg, SrsSamp
|
|||
return err;
|
||||
}
|
||||
|
||||
srs_error_t SrsRtcFromRtmpBridge::consume_packets(vector<SrsRtpPacket*>& pkts)
|
||||
srs_error_t SrsRtcRtpBuilder::consume_packets(vector<SrsRtpPacket*>& pkts)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
// TODO: FIXME: Consume a range of packets.
|
||||
for (int i = 0; i < (int)pkts.size(); i++) {
|
||||
SrsRtpPacket* pkt = pkts[i];
|
||||
if ((err = source_->on_rtp(pkt)) != srs_success) {
|
||||
if ((err = bridge_->on_rtp(pkt)) != srs_success) {
|
||||
err = srs_error_wrap(err, "consume sps/pps");
|
||||
break;
|
||||
}
|
||||
|
|
@ -1326,31 +1307,28 @@ srs_error_t SrsRtcFromRtmpBridge::consume_packets(vector<SrsRtpPacket*>& pkts)
|
|||
return err;
|
||||
}
|
||||
|
||||
SrsRtmpFromRtcBridge::SrsRtmpFromRtcBridge(SrsLiveSource *src)
|
||||
SrsRtcFrameBuilder::SrsRtcFrameBuilder(ISrsStreamBridge* bridge)
|
||||
{
|
||||
source_ = src;
|
||||
bridge_ = bridge;
|
||||
is_first_audio_ = true;
|
||||
codec_ = NULL;
|
||||
is_first_audio = true;
|
||||
is_first_video = true;
|
||||
format = NULL;
|
||||
rtp_key_frame_ts_ = -1;
|
||||
header_sn_ = 0;
|
||||
memset(cache_video_pkts_, 0, sizeof(cache_video_pkts_));
|
||||
rtp_key_frame_ts_ = -1;
|
||||
}
|
||||
|
||||
SrsRtmpFromRtcBridge::~SrsRtmpFromRtcBridge()
|
||||
SrsRtcFrameBuilder::~SrsRtcFrameBuilder()
|
||||
{
|
||||
srs_freep(codec_);
|
||||
srs_freep(format);
|
||||
clear_cached_video();
|
||||
}
|
||||
|
||||
srs_error_t SrsRtmpFromRtcBridge::initialize(SrsRequest* r)
|
||||
srs_error_t SrsRtcFrameBuilder::initialize(SrsRequest* r)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
srs_freep(codec_);
|
||||
codec_ = new SrsAudioTranscoder();
|
||||
format = new SrsRtmpFormat();
|
||||
|
||||
SrsAudioCodecId from = SrsAudioCodecIdOpus; // TODO: From SDP?
|
||||
SrsAudioCodecId to = SrsAudioCodecIdAAC; // The output audio codec.
|
||||
|
|
@ -1361,32 +1339,21 @@ srs_error_t SrsRtmpFromRtcBridge::initialize(SrsRequest* r)
|
|||
return srs_error_wrap(err, "bridge initialize");
|
||||
}
|
||||
|
||||
if ((err = format->initialize()) != srs_success) {
|
||||
return srs_error_wrap(err, "format initialize");
|
||||
}
|
||||
|
||||
// Setup the SPS/PPS parsing strategy.
|
||||
format->try_annexb_first = _srs_config->try_annexb_first(r->vhost);
|
||||
|
||||
return err;
|
||||
}
|
||||
|
||||
srs_error_t SrsRtmpFromRtcBridge::on_publish()
|
||||
srs_error_t SrsRtcFrameBuilder::on_publish()
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
is_first_audio_ = true;
|
||||
|
||||
is_first_audio = true;
|
||||
is_first_video = true;
|
||||
|
||||
// TODO: FIXME: Should sync with bridge?
|
||||
if ((err = source_->on_publish()) != srs_success) {
|
||||
return srs_error_wrap(err, "source publish");
|
||||
}
|
||||
|
||||
return err;
|
||||
return srs_success;
|
||||
}
|
||||
|
||||
srs_error_t SrsRtmpFromRtcBridge::on_rtp(SrsRtpPacket *pkt)
|
||||
void SrsRtcFrameBuilder::on_unpublish()
|
||||
{
|
||||
}
|
||||
|
||||
srs_error_t SrsRtcFrameBuilder::on_rtp(SrsRtpPacket *pkt)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
|
|
@ -1394,7 +1361,7 @@ srs_error_t SrsRtmpFromRtcBridge::on_rtp(SrsRtpPacket *pkt)
|
|||
return err;
|
||||
}
|
||||
|
||||
// Have no received any sender report, can't calculate avsync_time,
|
||||
// Have no received any sender report, can't calculate avsync_time,
|
||||
// discard it to avoid timestamp problem in live source
|
||||
if (pkt->get_avsync_time() <= 0) {
|
||||
return err;
|
||||
|
|
@ -1409,35 +1376,34 @@ srs_error_t SrsRtmpFromRtcBridge::on_rtp(SrsRtpPacket *pkt)
|
|||
return err;
|
||||
}
|
||||
|
||||
void SrsRtmpFromRtcBridge::on_unpublish()
|
||||
{
|
||||
// TODO: FIXME: Should sync with bridge?
|
||||
source_->on_unpublish();
|
||||
}
|
||||
|
||||
srs_error_t SrsRtmpFromRtcBridge::transcode_audio(SrsRtpPacket *pkt)
|
||||
srs_error_t SrsRtcFrameBuilder::transcode_audio(SrsRtpPacket *pkt)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
// to common message.
|
||||
uint32_t ts = pkt->get_avsync_time();
|
||||
if (is_first_audio) {
|
||||
if (is_first_audio_) {
|
||||
int header_len = 0;
|
||||
uint8_t* header = NULL;
|
||||
codec_->aac_codec_header(&header, &header_len);
|
||||
|
||||
SrsCommonMessage out_rtmp;
|
||||
packet_aac(&out_rtmp, (char *)header, header_len, ts, is_first_audio);
|
||||
packet_aac(&out_rtmp, (char *)header, header_len, ts, is_first_audio_);
|
||||
|
||||
if ((err = source_->on_audio(&out_rtmp)) != srs_success) {
|
||||
SrsSharedPtrMessage msg;
|
||||
if ((err = msg.create(&out_rtmp)) != srs_success) {
|
||||
return srs_error_wrap(err, "create message");
|
||||
}
|
||||
|
||||
if ((err = bridge_->on_frame(&msg)) != srs_success) {
|
||||
return srs_error_wrap(err, "source on audio");
|
||||
}
|
||||
|
||||
is_first_audio = false;
|
||||
is_first_audio_ = false;
|
||||
}
|
||||
|
||||
std::vector<SrsAudioFrame *> out_pkts;
|
||||
SrsRtpRawPayload *payload = dynamic_cast<SrsRtpRawPayload *>(pkt->payload());
|
||||
std::vector<SrsAudioFrame*> out_pkts;
|
||||
SrsRtpRawPayload *payload = dynamic_cast<SrsRtpRawPayload*>(pkt->payload());
|
||||
|
||||
SrsAudioFrame frame;
|
||||
frame.add_sample(payload->payload, payload->nn_payload);
|
||||
|
|
@ -1449,12 +1415,17 @@ srs_error_t SrsRtmpFromRtcBridge::transcode_audio(SrsRtpPacket *pkt)
|
|||
return err;
|
||||
}
|
||||
|
||||
for (std::vector<SrsAudioFrame *>::iterator it = out_pkts.begin(); it != out_pkts.end(); ++it) {
|
||||
for (std::vector<SrsAudioFrame*>::iterator it = out_pkts.begin(); it != out_pkts.end(); ++it) {
|
||||
SrsCommonMessage out_rtmp;
|
||||
out_rtmp.header.timestamp = (*it)->dts;
|
||||
packet_aac(&out_rtmp, (*it)->samples[0].bytes, (*it)->samples[0].size, ts, is_first_audio);
|
||||
packet_aac(&out_rtmp, (*it)->samples[0].bytes, (*it)->samples[0].size, ts, is_first_audio_);
|
||||
|
||||
if ((err = source_->on_audio(&out_rtmp)) != srs_success) {
|
||||
SrsSharedPtrMessage msg;
|
||||
if ((err = msg.create(&out_rtmp)) != srs_success) {
|
||||
return srs_error_wrap(err, "create message");
|
||||
}
|
||||
|
||||
if ((err = bridge_->on_frame(&msg)) != srs_success) {
|
||||
err = srs_error_wrap(err, "source on audio");
|
||||
break;
|
||||
}
|
||||
|
|
@ -1464,7 +1435,7 @@ srs_error_t SrsRtmpFromRtcBridge::transcode_audio(SrsRtpPacket *pkt)
|
|||
return err;
|
||||
}
|
||||
|
||||
void SrsRtmpFromRtcBridge::packet_aac(SrsCommonMessage* audio, char* data, int len, uint32_t pts, bool is_header)
|
||||
void SrsRtcFrameBuilder::packet_aac(SrsCommonMessage* audio, char* data, int len, uint32_t pts, bool is_header)
|
||||
{
|
||||
int rtmp_len = len + 2;
|
||||
audio->header.initialize_audio(rtmp_len, pts, 1);
|
||||
|
|
@ -1481,7 +1452,7 @@ void SrsRtmpFromRtcBridge::packet_aac(SrsCommonMessage* audio, char* data, int l
|
|||
audio->size = rtmp_len;
|
||||
}
|
||||
|
||||
srs_error_t SrsRtmpFromRtcBridge::packet_video(SrsRtpPacket* src)
|
||||
srs_error_t SrsRtcFrameBuilder::packet_video(SrsRtpPacket* src)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
|
|
@ -1521,7 +1492,7 @@ srs_error_t SrsRtmpFromRtcBridge::packet_video(SrsRtpPacket* src)
|
|||
return err;
|
||||
}
|
||||
|
||||
srs_error_t SrsRtmpFromRtcBridge::packet_video_key_frame(SrsRtpPacket* pkt)
|
||||
srs_error_t SrsRtcFrameBuilder::packet_video_key_frame(SrsRtpPacket* pkt)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
|
|
@ -1546,7 +1517,7 @@ srs_error_t SrsRtmpFromRtcBridge::packet_video_key_frame(SrsRtpPacket* pkt)
|
|||
char* flv = NULL;
|
||||
int nb_flv = 0;
|
||||
if ((err = avc->mux_avc2flv(sh, SrsVideoAvcFrameTypeKeyFrame, SrsVideoAvcFrameTraitSequenceHeader, pkt->get_avsync_time(),
|
||||
pkt->get_avsync_time(), &flv, &nb_flv)) != srs_success) {
|
||||
pkt->get_avsync_time(), &flv, &nb_flv)) != srs_success) {
|
||||
return srs_error_wrap(err, "avc to flv");
|
||||
}
|
||||
|
||||
|
|
@ -1557,7 +1528,12 @@ srs_error_t SrsRtmpFromRtcBridge::packet_video_key_frame(SrsRtpPacket* pkt)
|
|||
return srs_error_wrap(err, "create rtmp");
|
||||
}
|
||||
|
||||
if ((err = source_->on_video(&rtmp)) != srs_success) {
|
||||
SrsSharedPtrMessage msg;
|
||||
if ((err = msg.create(&rtmp)) != srs_success) {
|
||||
return srs_error_wrap(err, "create message");
|
||||
}
|
||||
|
||||
if ((err = bridge_->on_frame(&msg)) != srs_success) {
|
||||
return err;
|
||||
}
|
||||
}
|
||||
|
|
@ -1580,7 +1556,7 @@ srs_error_t SrsRtmpFromRtcBridge::packet_video_key_frame(SrsRtpPacket* pkt)
|
|||
lost_sn_ = header_sn_ + 1;
|
||||
clear_cached_video();
|
||||
srs_warn("drop old ts=%u, header=%hu, lost=%hu, set new ts=%u, header=%hu, lost=%hu",
|
||||
(uint32_t)old_ts, old_header_sn, old_lost_sn, (uint32_t)rtp_key_frame_ts_, header_sn_, lost_sn_);
|
||||
(uint32_t)old_ts, old_header_sn, old_lost_sn, (uint32_t)rtp_key_frame_ts_, header_sn_, lost_sn_);
|
||||
}
|
||||
|
||||
uint16_t index = cache_index(pkt->header.get_sequence());
|
||||
|
|
@ -1616,7 +1592,7 @@ srs_error_t SrsRtmpFromRtcBridge::packet_video_key_frame(SrsRtpPacket* pkt)
|
|||
return err;
|
||||
}
|
||||
|
||||
srs_error_t SrsRtmpFromRtcBridge::packet_video_rtmp(const uint16_t start, const uint16_t end)
|
||||
srs_error_t SrsRtcFrameBuilder::packet_video_rtmp(const uint16_t start, const uint16_t end)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
|
|
@ -1646,7 +1622,7 @@ srs_error_t SrsRtmpFromRtcBridge::packet_video_rtmp(const uint16_t start, const
|
|||
if (stap_payload) {
|
||||
for (int j = 0; j < (int)stap_payload->nalus.size(); ++j) {
|
||||
SrsSample* sample = stap_payload->nalus.at(j);
|
||||
if (sample->size > 0) {
|
||||
if (sample->size > 0) {
|
||||
nb_payload += 4 + sample->size;
|
||||
}
|
||||
}
|
||||
|
|
@ -1664,7 +1640,7 @@ srs_error_t SrsRtmpFromRtcBridge::packet_video_rtmp(const uint16_t start, const
|
|||
srs_warn("empty nalu");
|
||||
return err;
|
||||
}
|
||||
|
||||
|
||||
//type_codec1 + avc_type + composition time + nalu size + nalu
|
||||
nb_payload += 1 + 1 + 3;
|
||||
|
||||
|
|
@ -1725,7 +1701,7 @@ srs_error_t SrsRtmpFromRtcBridge::packet_video_rtmp(const uint16_t start, const
|
|||
if (stap_payload) {
|
||||
for (int j = 0; j < (int)stap_payload->nalus.size(); ++j) {
|
||||
SrsSample* sample = stap_payload->nalus.at(j);
|
||||
if (sample->size > 0) {
|
||||
if (sample->size > 0) {
|
||||
payload.write_4bytes(sample->size);
|
||||
payload.write_bytes(sample->bytes, sample->size);
|
||||
}
|
||||
|
|
@ -1745,7 +1721,12 @@ srs_error_t SrsRtmpFromRtcBridge::packet_video_rtmp(const uint16_t start, const
|
|||
srs_freep(pkt);
|
||||
}
|
||||
|
||||
if ((err = source_->on_video(&rtmp)) != srs_success) {
|
||||
SrsSharedPtrMessage msg;
|
||||
if ((err = msg.create(&rtmp)) != srs_success) {
|
||||
return srs_error_wrap(err, "create message");
|
||||
}
|
||||
|
||||
if ((err = bridge_->on_frame(&msg)) != srs_success) {
|
||||
srs_warn("fail to pack video frame");
|
||||
}
|
||||
|
||||
|
|
@ -1765,7 +1746,7 @@ srs_error_t SrsRtmpFromRtcBridge::packet_video_rtmp(const uint16_t start, const
|
|||
return err;
|
||||
}
|
||||
|
||||
int32_t SrsRtmpFromRtcBridge::find_next_lost_sn(uint16_t current_sn, uint16_t& end_sn)
|
||||
int32_t SrsRtcFrameBuilder::find_next_lost_sn(uint16_t current_sn, uint16_t& end_sn)
|
||||
{
|
||||
uint32_t last_rtp_ts = cache_video_pkts_[cache_index(header_sn_)].rtp_ts;
|
||||
for (int i = 0; i < s_cache_size; ++i) {
|
||||
|
|
@ -1791,7 +1772,7 @@ int32_t SrsRtmpFromRtcBridge::find_next_lost_sn(uint16_t current_sn, uint16_t& e
|
|||
return -2;
|
||||
}
|
||||
|
||||
void SrsRtmpFromRtcBridge::clear_cached_video()
|
||||
void SrsRtcFrameBuilder::clear_cached_video()
|
||||
{
|
||||
for (size_t i = 0; i < s_cache_size; i++)
|
||||
{
|
||||
|
|
@ -1805,7 +1786,7 @@ void SrsRtmpFromRtcBridge::clear_cached_video()
|
|||
}
|
||||
}
|
||||
|
||||
bool SrsRtmpFromRtcBridge::check_frame_complete(const uint16_t start, const uint16_t end)
|
||||
bool SrsRtcFrameBuilder::check_frame_complete(const uint16_t start, const uint16_t end)
|
||||
{
|
||||
int16_t cnt = srs_rtp_seq_distance(start, end) + 1;
|
||||
srs_assert(cnt >= 1);
|
||||
|
|
@ -1833,6 +1814,7 @@ bool SrsRtmpFromRtcBridge::check_frame_complete(const uint16_t start, const uint
|
|||
|
||||
return fu_s_c == fu_e_c;
|
||||
}
|
||||
|
||||
#endif
|
||||
|
||||
SrsCodecPayload::SrsCodecPayload()
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue