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WebRTC video frames count

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This commit is contained in:
Artem Smorodin 2026-02-11 05:00:50 +03:00
parent 6e2392f366
commit 9e9b5f15ee
2 changed files with 14 additions and 8 deletions
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20
trunk/src/app/srs_app_rtc_conn.cpp
View File

@ -1687,7 +1687,7 @@ srs_error_t SrsRtcPublishStream::do_on_rtp_plaintext(SrsRtpPacket *&pkt, SrsBuff
} }
// Update RTP packet statistics. // Update RTP packet statistics.
update_rtp_packet_stats(is_audio); update_rtp_packet_stats(pkt, track, is_audio);
// Consume packet by track. // Consume packet by track.
if ((err = track->on_rtp(source_, pkt)) != srs_success) { if ((err = track->on_rtp(source_, pkt)) != srs_success) {
@ -1709,19 +1709,25 @@ srs_error_t SrsRtcPublishStream::do_on_rtp_plaintext(SrsRtpPacket *&pkt, SrsBuff
return err; 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; 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) { if (is_audio) {
++nn_audio_frames_; ++nn_audio_frames_;
} else { } else if (pkt->header_.get_marker()) {
++nn_video_frames_; ++nn_video_frames_;
} }
// Update the stat for video frames, counting RTP packets as frames. // Update the stat for video frames. Note that for WebRTC, a video frame may be
if (nn_video_frames_ > 288) { // 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) { 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_warn("RTC: stat video frames err %s", srs_error_desc(err).c_str());
srs_freep(err); srs_freep(err);
@ -1730,7 +1736,7 @@ void SrsRtcPublishStream::update_rtp_packet_stats(bool is_audio)
} }
// Update the stat for audio frames periodically. // 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) { 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_warn("RTC: stat audio frames err %s", srs_error_desc(err).c_str());
srs_freep(err); srs_freep(err);

2
trunk/src/app/srs_app_rtc_conn.hpp
View File

@ -622,7 +622,7 @@ public:
// clang-format off // clang-format off
SRS_DECLARE_PRIVATE: // clang-format on SRS_DECLARE_PRIVATE: // clang-format on
srs_error_t do_on_rtp_plaintext(SrsRtpPacket *&pkt, SrsBuffer *buf); 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: public:
srs_error_t check_send_nacks(); srs_error_t check_send_nacks();