Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1001)

Unified Diff: media/cast/audio_sender/audio_sender.cc

Issue 343523005: Cast: Avoid retransmit if we sent the same packet recently (less than RTT) (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: store time before send, not after Created 6 years, 6 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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) {

Powered by Google App Engine
This is Rietveld 408576698