Cast: Avoid retransmit if we sent the same packet recently (less than RTT)

media/cast/audio_sender/audio_sender.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/cast/audio_sender/audio_sender.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop.h"
 #include "media/cast/audio_sender/audio_encoder.h"
@@ skipping 88 matching lines @@
   }
 
   frame_id_to_rtp_timestamp_[audio_frame->frame_id & 0xff] =
       audio_frame->rtp_timestamp;
   transport_sender_->InsertCodedAudioFrame(*audio_frame);
 }
 
 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.

[palmer, 2014/06/18 21:18:02]

File a bug to study this, so it doesn't get forgotten. (Maybe you already have
done so.) Then reference the bug # here.

[hubbe, 2014/06/18 21:20:35]

No real need, alpha is deep in that study right now..

+  transport_sender_->ResendPackets(
+      true, missing_frames_and_packets, false, rtt);

[Alpha Left Google, 2014/06/18 21:07:37]

Sorry one last thing I missed in the last review. I think it is best to leave
audio RTX untouched. Audio packets don't degrade performance as bad as video RTX
packets. And receiver currently only NACK once that means audio packet has at
most 2 chance to send. Removing the resend can lead to only once chance to send.

[hubbe, 2014/06/18 21:20:35]

Changed back to min_rtt

 }
 
 void AudioSender::IncomingRtcpPacket(scoped_ptr<Packet> packet) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   rtcp_.IncomingRtcpPacket(&packet->front(), packet->size());
 }
 
 void AudioSender::ScheduleNextRtcpReport() {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
   base::TimeDelta time_to_next =
@@ skipping 47 matching lines @@
   uint32 acked_frame_id = static_cast<uint32>(cast_feedback.ack_frame_id_);
   VLOG(2) << "Received audio ACK: " << acked_frame_id;
   cast_environment_->Logging()->InsertFrameEvent(
         cast_environment_->Clock()->NowTicks(),
         FRAME_ACK_RECEIVED, AUDIO_EVENT,
         frame_id_to_rtp_timestamp_[acked_frame_id & 0xff], acked_frame_id);
 }
 
 }  // namespace cast
 }  // namespace media