Make the audio channel communicate network state changes to the call.

webrtc/media/engine/webrtcvideoengine2.cc

 /*
  *  Copyright (c) 2014 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 1342 matching lines @@
   // filter RTCP anymore incoming RTCP packets could've been going to audio (so
   // logging failures spam the log).
   call_->Receiver()->DeliverPacket(
       webrtc::MediaType::VIDEO,
       reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
       webrtc_packet_time);
 }
 
 void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
   LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
-  call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
+  call_->SignalChannelNetworkState(
+    webrtc::MediaType::VIDEO,

[the sun, 2016/03/07 14:16:43]

nit: indent, plus make a single line of the ternary below

[skvlad, 2016/03/07 19:20:56]

Done.

+    ready ?
+    webrtc::kNetworkUp : webrtc::kNetworkDown);
 }
 
 bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) {
   LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
                   << (mute ? "mute" : "unmute");
   RTC_DCHECK(ssrc != 0);
   rtc::CritScope stream_lock(&stream_crit_);
   const auto& kv = send_streams_.find(ssrc);
   if (kv == send_streams_.end()) {
     LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
@@ skipping 1160 matching lines @@
         rtx_mapping[video_codecs[i].codec.id] !=
             fec_settings.red_payload_type) {
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket