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

webrtc/video/rtp_streams_synchronizer.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 105 matching lines @@
   int audio_jitter_buffer_delay_ms = 0;
   int playout_buffer_delay_ms = 0;
   if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
                                             &audio_jitter_buffer_delay_ms,
                                             &playout_buffer_delay_ms) != 0) {
     return;
   }
   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;
   }
 
   if (UpdateMeasurements(&audio_measurement_, audio_rtp_rtcp_,
                          audio_rtp_receiver_) != 0) {
     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_,
                                    &relative_delay_ms)) {
     return;
   }
 
   TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay", current_video_delay_ms);
   TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay", current_audio_delay_ms);
   TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
   int target_audio_delay_ms = 0;
   int target_video_delay_ms = current_video_delay_ms;
   // Calculate the necessary extra audio delay and desired total video
   // delay to get the streams in sync.
   if (!sync_->ComputeDelays(relative_delay_ms,
                             current_audio_delay_ms,
                             &target_audio_delay_ms,
                             &target_video_delay_ms)) {
     return;
   }
 
   if (voe_sync_interface_->SetMinimumPlayoutDelay(
       voe_channel_id_, target_audio_delay_ms) == -1) {
     LOG(LS_ERROR) << "Error setting voice delay.";
   }
   video_receiver_->SetMinimumPlayoutDelay(target_video_delay_ms);
 }
 
 bool RtpStreamsSynchronizer::GetStreamSyncOffsetInMs(

[mflodman, 2016/09/14 13:19:08]

Will this method also cause stats problems if it's being called when there are
no new frames? Mainly for the opposite problem I assume, when audio is muted and
video is received.

     const VideoFrame& frame, int64_t* stream_offset_ms) const {
   rtc::CritScope lock(&crit_);
   if (voe_channel_id_ == -1)
     return false;
 
   uint32_t playout_timestamp = 0;
   if (voe_sync_interface_->GetPlayoutTimestamp(voe_channel_id_,
                                                playout_timestamp) != 0) {
     return false;
   }
@@ skipping 13 matching lines @@
   int64_t time_to_render_ms =
       frame.render_time_ms() - clock_->TimeInMilliseconds();
   if (time_to_render_ms > 0)
     latest_video_ntp += time_to_render_ms;
 
   *stream_offset_ms = latest_audio_ntp - latest_video_ntp;
   return true;
 }
 
 }  // namespace webrtc