Adding an option to disable audio PTS adjustment on sender reports

pkg/synchronizer/synchronizer.go

@@ -29,8 +29,9 @@ type SynchronizerOption func(*SynchronizerConfig)
 
 // SynchronizerConfig holds configuration for the Synchronizer
 type SynchronizerConfig struct {
-	MaxTsDiff time.Duration
-	OnStarted func()
+	MaxTsDiff                  time.Duration
+	OnStarted                  func()
+	AudioPTSAdjustmentDisabled bool
 }
 
 // WithMaxTsDiff sets the maximum acceptable difference between RTP packets
@@ -48,6 +49,17 @@ func WithOnStarted(onStarted func()) SynchronizerOption {
 	}
 }
 
+// WithAudioPTSAdjustmentDisabled - disables auto PTS adjustments after sender reports
+// Use case: when media processing pipeline needs stable - monotonically increasing
+// PTS sequence - small adjustments coming from RTCP sender reports could cause gaps in the audio
+// Media processing pipeline could opt out of auto PTS adjustments and handle the gap
+// by e.g modifying tempo to compensate instead
+func WithAudioPTSAdjustmentDisabled() SynchronizerOption {
+	return func(config *SynchronizerConfig) {
+		config.AudioPTSAdjustmentDisabled = true
+	}
+}
+
 // a single Synchronizer is shared between all audio and video writers
 type Synchronizer struct {
 	sync.RWMutex

pkg/synchronizer/track.go

@@ -46,7 +46,8 @@ type TrackSynchronizer struct {
 	*rtpConverter
 
 	// config
-	maxTsDiff time.Duration // maximum acceptable difference between RTP packets
+	maxTsDiff                   time.Duration // maximum acceptable difference between RTP packets
+	audioPTSAdjustmentsDisabled bool          // disable audio packets PTS adjustments on SRs
 
 	// timing info
 	startTime       time.Time     // time first packet was pushed
@@ -67,11 +68,12 @@ type TrackSynchronizer struct {
 
 func newTrackSynchronizer(s *Synchronizer, track TrackRemote) *TrackSynchronizer {
 	t := &TrackSynchronizer{
-		sync:         s,
-		track:        track,
-		logger:       logger.GetLogger().WithValues("trackID", track.ID(), "codec", track.Codec().MimeType),
-		rtpConverter: newRTPConverter(int64(track.Codec().ClockRate)),
-		maxTsDiff:    s.config.MaxTsDiff,
+		sync:                        s,
+		track:                       track,
+		logger:                      logger.GetLogger().WithValues("trackID", track.ID(), "codec", track.Codec().MimeType),
+		rtpConverter:                newRTPConverter(int64(track.Codec().ClockRate)),
+		maxTsDiff:                   s.config.MaxTsDiff,
+		audioPTSAdjustmentsDisabled: s.config.AudioPTSAdjustmentDisabled,
 	}
 
 	return t
@@ -123,11 +125,14 @@ func (t *TrackSynchronizer) GetPTS(pkt *rtp.Packet) (time.Duration, error) {
 		t.startRTP = ts - t.toRTP(pts)
 	}
 
-	if t.currentPTSOffset > t.desiredPTSOffset {
-		t.currentPTSOffset = max(t.currentPTSOffset-maxAdjustment, t.desiredPTSOffset)
-	} else if t.currentPTSOffset < t.desiredPTSOffset {
-		t.currentPTSOffset = min(t.currentPTSOffset+maxAdjustment, t.desiredPTSOffset)
+	if t.shouldAdjustPTS() {
+		if t.currentPTSOffset > t.desiredPTSOffset {
+			t.currentPTSOffset = max(t.currentPTSOffset-maxAdjustment, t.desiredPTSOffset)
+		} else if t.currentPTSOffset < t.desiredPTSOffset {
+			t.currentPTSOffset = min(t.currentPTSOffset+maxAdjustment, t.desiredPTSOffset)
+		}
 	}
+
 	adjusted := pts + t.currentPTSOffset
 
 	// if past end time, return EOF
@@ -179,6 +184,14 @@ func (t *TrackSynchronizer) acceptable(d time.Duration) bool {
 	return d > -t.maxTsDiff && d < t.maxTsDiff
 }
 
+func (t *TrackSynchronizer) shouldAdjustPTS() bool {
+	adjustmentEnabled := true
+	if t.track.Kind() == webrtc.RTPCodecTypeAudio {
+		adjustmentEnabled = !t.audioPTSAdjustmentsDisabled
+	}
+	return adjustmentEnabled && (t.currentPTSOffset != t.desiredPTSOffset)
+}
+
 type rtpConverter struct {
 	ts  uint64
 	rtp uint64