Do not update stream synchronization if no new video packet has been received since last u…

webrtc/video/rtp_streams_synchronizer.cc

Index: webrtc/video/rtp_streams_synchronizer.cc
diff --git a/webrtc/video/rtp_streams_synchronizer.cc b/webrtc/video/rtp_streams_synchronizer.cc
index 75dd8cd1b3ad4571cc36b2608c754bdb7104f941..d33e8733d45ce05828ca3ff0045daefae8f9f85c 100644
--- a/webrtc/video/rtp_streams_synchronizer.cc
+++ b/webrtc/video/rtp_streams_synchronizer.cc
@@ -123,6 +123,9 @@ void RtpStreamsSynchronizer::Process() {
   const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
       playout_buffer_delay_ms;
 
+  int64_t last_audio_receive_ms = audio_measurement_.latest_receive_time_ms;
+  int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
+
   if (UpdateMeasurements(&video_measurement_, video_rtp_rtcp_,
                          video_rtp_receiver_) != 0) {
     return;
@@ -133,6 +136,12 @@ void RtpStreamsSynchronizer::Process() {
     return;
   }
 
+  if (last_audio_receive_ms == audio_measurement_.latest_receive_time_ms ||
+      last_video_receive_ms == video_measurement_.latest_receive_time_ms) {
+    // No new audio or video packet has been received since last update.
+    return;
+  }
+
   int relative_delay_ms;
   // Calculate how much later or earlier the audio stream is compared to video.
   if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_,