Fix #3167: WebRTC: Play stucked when republish. v4.0.260

trunk/doc/CHANGELOG.md

@@ -8,6 +8,7 @@ The changelog for SRS.
 
 ## SRS 4.0 Changelog
 
+* v4.0, 2022-09-03, For [#3167](https://github.com/ossrs/srs/issues/3167): WebRTC: Play stucked when republish. v4.0.260
 * v4.0, 2022-09-02, For [#307](https://github.com/ossrs/srs/issues/307): WebRTC: Support use domain name as CANDIDATE. v4.0.259
 * v4.0, 2022-08-29, Copy libxml2-dev for FFmpeg. v4.0.258
 * v4.0, 2022-08-24, STAT: Support config server_id and generate one if empty. v4.0.257

trunk/src/app/srs_app_rtc_conn.cpp

@@ -510,10 +510,12 @@ srs_error_t SrsRtcPlayStream::initialize(SrsRequest* req, std::map<uint32_t, Srs
 
 void SrsRtcPlayStream::on_stream_change(SrsRtcSourceDescription* desc)
 {
+ if (!desc) return;
+
  // Refresh the relation for audio.
  // TODO: FIMXE: Match by label?
  if (desc && desc->audio_track_desc_ && audio_tracks_.size() == 1) {
- if (! audio_tracks_.empty()) {
+ if (!audio_tracks_.empty()) {
  uint32_t ssrc = desc->audio_track_desc_->ssrc_;
  SrsRtcAudioSendTrack* track = audio_tracks_.begin()->second;
 
@@ -525,7 +527,7 @@ void SrsRtcPlayStream::on_stream_change(SrsRtcSourceDescription* desc)
  // Refresh the relation for video.
  // TODO: FIMXE: Match by label?
  if (desc && desc->video_track_descs_.size() == 1) {
- if (! video_tracks_.empty()) {
+ if (!video_tracks_.empty()) {
  SrsRtcTrackDescription* vdesc = desc->video_track_descs_.at(0);
  uint32_t ssrc = vdesc->ssrc_;
  SrsRtcVideoSendTrack* track = video_tracks_.begin()->second;
@@ -534,6 +536,15 @@ void SrsRtcPlayStream::on_stream_change(SrsRtcSourceDescription* desc)
  video_tracks_.insert(make_pair(ssrc, track));
  }
  }
+
+ // Request keyframe(PLI) when stream changed.
+ if (desc->audio_track_desc_) {
+ pli_worker_->request_keyframe(desc->audio_track_desc_->ssrc_, cid_);
+ }
+ for (vector<SrsRtcTrackDescription*>::iterator it = desc->video_track_descs_.begin(); it != desc->video_track_descs_.end(); ++it) {
+ SrsRtcTrackDescription* vdesc = *it;
+ pli_worker_->request_keyframe(vdesc->ssrc_, cid_);
+ }
 }
 
 srs_error_t SrsRtcPlayStream::on_reload_vhost_play(string vhost)

trunk/src/app/srs_app_rtc_source.cpp

@@ -2546,12 +2546,53 @@ srs_error_t SrsRtcVideoRecvTrack::check_send_nacks()
  return err;
 }
 
+SrsRtcJitter::SrsRtcJitter(uint32_t base)
+{
+ pkt_base_ = pkt_last_ = 0;
+ correct_last_ = correct_base_ = 0;
+ base_ = base;
+ init_ = false;
+}
+
+SrsRtcJitter::~SrsRtcJitter()
+{
+}
+
+uint32_t SrsRtcJitter::correct(uint32_t ts)
+{
+ if (!init_) {
+ init_ = true;
+ correct_base_ = base_;
+ srs_trace("RTC: Jitter init base=%u, ts=%u", base_, ts);
+ }
+ if (!pkt_base_) pkt_base_ = ts;
+
+ if (pkt_last_) {
+ int32_t distance = srs_rtp_ts_distance(ts, pkt_last_);
+ static int32_t max_deviation = 90 * 3 * 1000;
+ if (distance > max_deviation || distance < -1 * max_deviation) {
+ srs_trace("RTC: Jitter rebase ts=%u, last=%u, distance=%d, pkt-base=%u/%u, correct-base=%u/%u", ts, pkt_last_, distance, pkt_base_, ts, correct_base_, correct_last_);
+ pkt_base_ = ts;
+ correct_base_ = correct_last_;
+ }
+ }
+ pkt_last_ = ts;
+
+ correct_last_ = correct_base_ + ts - pkt_base_;
+ return correct_last_;
+}
+
 SrsRtcSendTrack::SrsRtcSendTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc, bool is_audio)
 {
  session_ = session;
  track_desc_ = track_desc->copy();
  nack_no_copy_ = false;
 
+ seqno_ = 0;
+ init_ = false;
+ // Make a different start of sequence number, for debugging.
+ jitter_ = new SrsRtcJitter(track_desc_->type_ == "audio" ? 10000 : 20000);
+
  if (is_audio) {
  rtp_queue_ = new SrsRtpRingBuffer(100);
  } else {
@@ -2566,6 +2607,7 @@ SrsRtcSendTrack::~SrsRtcSendTrack()
  srs_freep(rtp_queue_);
  srs_freep(track_desc_);
  srs_freep(nack_epp);
+ srs_freep(jitter_);
 }
 
 bool SrsRtcSendTrack::has_ssrc(uint32_t ssrc)
@@ -2616,6 +2658,23 @@ std::string SrsRtcSendTrack::get_track_id()
  return track_desc_->id_;
 }
 
+void SrsRtcSendTrack::rebuild_packet(SrsRtpPacket* pkt)
+{
+ // Rebuild the sequence number.
+ if (!init_) {
+ init_ = true;
+ // Make a different start of sequence number, for debugging.
+ seqno_ = track_desc_->type_ == "audio" ? 1000 : 2000;
+ srs_trace("RTC: Seqno rebuild %s track=%s, ssrc=%d, seqno=%d to %d", track_desc_->type_.c_str(), track_desc_->id_.c_str(),
+ pkt->header.get_ssrc(), pkt->header.get_sequence(), seqno_);
+ }
+ pkt->header.set_sequence(seqno_++);
+
+ // Rebuild the timestamp.
+ uint32_t ts = pkt->header.get_timestamp();
+ pkt->header.set_timestamp(jitter_->correct(ts));
+}
+
 srs_error_t SrsRtcSendTrack::on_nack(SrsRtpPacket** ppkt)
 {
  srs_error_t err = srs_success;
@@ -2693,10 +2752,16 @@ srs_error_t SrsRtcAudioSendTrack::on_rtp(SrsRtpPacket* pkt)
  // TODO: FIXME: Should update PT for RTX.
  }
 
+ // Rebuild the sequence number and timestamp of packet, see https://github.com/ossrs/srs/issues/3167
+ rebuild_packet(pkt);
+
  if ((err = session_->do_send_packet(pkt)) != srs_success) {
  return srs_error_wrap(err, "raw send");
  }
 
+ srs_info("RTC: Send audio ssrc=%d, seqno=%d, keyframe=%d, ts=%u", pkt->header.get_ssrc(),
+ pkt->header.get_sequence(), pkt->is_keyframe(), pkt->header.get_timestamp());
+
  return err;
 }
 
@@ -2737,10 +2802,16 @@ srs_error_t SrsRtcVideoSendTrack::on_rtp(SrsRtpPacket* pkt)
  // TODO: FIXME: Should update PT for RTX.
  }
 
+ // Rebuild the sequence number and timestamp of packet, see https://github.com/ossrs/srs/issues/3167
+ rebuild_packet(pkt);
+
  if ((err = session_->do_send_packet(pkt)) != srs_success) {
  return srs_error_wrap(err, "raw send");
  }
 
+ srs_info("RTC: Send video ssrc=%d, seqno=%d, keyframe=%d, ts=%u", pkt->header.get_ssrc(),
+ pkt->header.get_sequence(), pkt->is_keyframe(), pkt->header.get_timestamp());
+
  return err;
 }
 

trunk/src/app/srs_app_rtc_source.hpp

@@ -579,16 +579,43 @@ class SrsRtcVideoRecvTrack : public SrsRtcRecvTrack, public ISrsRtspPacketDecode
  virtual srs_error_t check_send_nacks();
 };
 
+class SrsRtcJitter
+{
+private:
+ // The ts about packet.
+ uint32_t pkt_base_;
+ uint32_t pkt_last_;
+ // The ts after corrected.
+ uint32_t correct_base_;
+ uint32_t correct_last_;
+ // The base timestamp by config, start from it.
+ uint32_t base_;
+ // Whether initialized. Note that we should not use correct_base_(0) as init state, because it might flip back.
+ bool init_;
+public:
+ SrsRtcJitter(uint32_t base);
+ virtual ~SrsRtcJitter();
+public:
+ uint32_t correct(uint32_t ts);
+};
+
 class SrsRtcSendTrack
 {
-protected:
+public:
  // send track description
  SrsRtcTrackDescription* track_desc_;
 protected:
  // The owner connection for this track.
  SrsRtcConnection* session_;
  // NACK ARQ ring buffer.
  SrsRtpRingBuffer* rtp_queue_;
+protected:
+ // Current sequence number.
+ uint64_t seqno_;
+ // Whether initialized. Note that we should not use seqno_(0) as init state, because it might flip back.
+ bool init_;
+ // The jitter to correct ts.
+ SrsRtcJitter* jitter_;
 private:
  // By config, whether no copy.
  bool nack_no_copy_;
@@ -605,6 +632,8 @@ class SrsRtcSendTrack
  bool set_track_status(bool active);
  bool get_track_status();
  std::string get_track_id();
+protected:
+ void rebuild_packet(SrsRtpPacket* pkt);
 public:
  // Note that we can set the pkt to NULL to avoid copy, for example, if the NACK cache the pkt and
  // set to NULL, nack nerver copy it but set the pkt to NULL.

trunk/src/core/srs_core_version4.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR 4
 #define VERSION_MINOR 0
-#define VERSION_REVISION 259
+#define VERSION_REVISION 260
 
 #endif

trunk/src/kernel/srs_kernel_rtc_rtp.hpp

@@ -61,6 +61,10 @@ inline int16_t srs_rtp_seq_distance(const uint16_t& prev_value, const uint16_t&
 {
  return (int16_t)(value - prev_value);
 }
+inline int32_t srs_rtp_ts_distance(const uint32_t& prev_value, const uint32_t& value)
+{
+ return (int32_t)(value - prev_value);
+}
 
 // For map to compare the sequence of RTP.
 struct SrsSeqCompareLess {