Chromium Code Reviews| Index: media/cast/audio_sender/audio_sender.cc |
| diff --git a/media/cast/audio_sender/audio_sender.cc b/media/cast/audio_sender/audio_sender.cc |
| index 27b42d0dc8c2ac08757cb561d208f849c5835baa..836fafd9f8484315c2c12c61ade457bbc02d90b6 100644 |
| --- a/media/cast/audio_sender/audio_sender.cc |
| +++ b/media/cast/audio_sender/audio_sender.cc |
| @@ -106,7 +106,17 @@ void AudioSender::SendEncodedAudioFrame( |
| void AudioSender::ResendPackets( |
| const MissingFramesAndPacketsMap& missing_frames_and_packets) { |
| DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); |
| - transport_sender_->ResendPackets(true, missing_frames_and_packets, false); |
| + |
| + base::TimeDelta rtt; |
| + base::TimeDelta avg_rtt; |
| + base::TimeDelta min_rtt; |
| + base::TimeDelta max_rtt; |
| + rtcp_.Rtt(&rtt, &avg_rtt, &min_rtt, &max_rtt); |
| + |
| + // It would probably be better to use the 10% percentile rtt |
| + // rather than the min. |
| + transport_sender_->ResendPackets( |
| + true, missing_frames_and_packets, false, min_rtt); |
|
Alpha Left Google
2014/06/18 01:43:32
In VideoSender we used rtt and avg_rtt while min_r
hubbe
2014/06/18 20:22:28
Done.
|
| } |
| void AudioSender::IncomingRtcpPacket(scoped_ptr<Packet> packet) { |