Jitter delay now depend on protection mode (FEC/NACK).

webrtc/video/video_receive_stream.cc

 /*
  *  Copyright (c) 2013 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 170 matching lines @@
   RTC_DCHECK(process_thread_);
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(call_stats_);
   RTC_DCHECK(remb_);
   RTC_CHECK(vie_channel_.Init() == 0);
 
   // Register the channel to receive stats updates.
   call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
 
   // TODO(pbos): This is not fine grained enough...
-  vie_channel_.SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false, -1,
-                                 -1);
+  vie_channel_.SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0,

[stefan-webrtc, 2016/05/04 09:30:48]

I think you may have to rebase this as mflodman changed this code recently (and
moved it into video_stream_decoder.cc, I think).

[philipel, 2016/05/04 10:56:05]

Talked to Magnus and there is still some refactoring in progress. I will wait
with rebasing until he is done.

+                                 config_.rtp.fec.ulpfec_payload_type != -1,
+                                 config_.rtp.fec.red_payload_type,
+                                 config_.rtp.fec.ulpfec_payload_type);
   RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
       << "A stream should not be configured with RTCP disabled. This value is "
          "reserved for internal usage.";
   rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
 
   RTC_DCHECK(config_.rtp.remote_ssrc != 0);
   // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
   RTC_DCHECK(config_.rtp.local_ssrc != 0);
   RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
   rtp_rtcp_->SetSSRC(config_.rtp.local_ssrc);
@@ skipping 237 matching lines @@
     const std::vector<uint16_t>& sequence_numbers) {
   rtp_rtcp_->SendNack(sequence_numbers);
 }
 
 void VideoReceiveStream::RequestKeyFrame() {
   rtp_rtcp_->RequestKeyFrame();
 }
 
 }  // namespace internal
 }  // namespace webrtc