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WebRTC: counting video frames for stats instead of packets #4635

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20 changes: 13 additions & 7 deletions trunk/src/app/srs_app_rtc_conn.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -1687,7 +1687,7 @@ srs_error_t SrsRtcPublishStream::do_on_rtp_plaintext(SrsRtpPacket *&pkt, SrsBuff
}

// Update RTP packet statistics.
update_rtp_packet_stats(is_audio);
update_rtp_packet_stats(pkt, track, is_audio);

// Consume packet by track.
if ((err = track->on_rtp(source_, pkt)) != srs_success) {
Expand All @@ -1709,19 +1709,25 @@ srs_error_t SrsRtcPublishStream::do_on_rtp_plaintext(SrsRtpPacket *&pkt, SrsBuff
return err;
}

void SrsRtcPublishStream::update_rtp_packet_stats(bool is_audio)
void SrsRtcPublishStream::update_rtp_packet_stats(SrsRtpPacket *pkt, SrsRtcRecvTrack *track, bool is_audio)
{
srs_error_t err = srs_success;

// Count RTP packets for statistics.
// Count audio packets and video frames for statistics.
bool is_primary_ssrc = track && pkt->header_.get_ssrc() == track->get_ssrc();
if (!is_primary_ssrc) {
return;
}

if (is_audio) {
++nn_audio_frames_;
} else {
} else if (pkt->header_.get_marker()) {
++nn_video_frames_;
}

// Update the stat for video frames, counting RTP packets as frames.
if (nn_video_frames_ > 288) {
// Update the stat for video frames. Note that for WebRTC, a video frame may be
// fragmented into multiple RTP packets, so we count frames by RTP marker bit.
if (nn_video_frames_ > 30) {
if ((err = stat_->on_video_frames(req_, nn_video_frames_)) != srs_success) {
srs_warn("RTC: stat video frames err %s", srs_error_desc(err).c_str());
srs_freep(err);
Expand All @@ -1730,7 +1736,7 @@ void SrsRtcPublishStream::update_rtp_packet_stats(bool is_audio)
}

// Update the stat for audio frames periodically.
if (nn_audio_frames_ > 288) {
if (nn_audio_frames_ > 50) {
if ((err = stat_->on_audio_frames(req_, nn_audio_frames_)) != srs_success) {
srs_warn("RTC: stat audio frames err %s", srs_error_desc(err).c_str());
srs_freep(err);
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2 changes: 1 addition & 1 deletion trunk/src/app/srs_app_rtc_conn.hpp
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Original file line number Diff line number Diff line change
Expand Up @@ -622,7 +622,7 @@ class SrsRtcPublishStream : public ISrsRtcPublishStream
// clang-format off
SRS_DECLARE_PRIVATE: // clang-format on
srs_error_t do_on_rtp_plaintext(SrsRtpPacket *&pkt, SrsBuffer *buf);
void update_rtp_packet_stats(bool is_audio);
void update_rtp_packet_stats(SrsRtpPacket *pkt, SrsRtcRecvTrack *track, bool is_audio);

public:
srs_error_t check_send_nacks();
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