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

webrtc/call.h

 /*
  *  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.
  */
 #ifndef WEBRTC_CALL_H_
@@ skipping 115 matching lines @@
   // pacing delay, etc.
   virtual Stats GetStats() const = 0;
 
   // TODO(pbos): Like BitrateConfig above this is currently per-stream instead
   // of maximum for entire Call. This should be fixed along with the above.
   // Specifying a start bitrate (>0) will currently reset the current bitrate
   // estimate. This is due to how the 'x-google-start-bitrate' flag is currently
   // implemented.
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
-  virtual void SignalNetworkState(NetworkState state) = 0;
+
+  // TODO(skvlad): When the unbundled case with multiple streams for the same
+  // media type going over different networks is supported, track the state
+  // for each stream separately. Right now it's global per media type.
+  virtual void SignalChannelNetworkState(MediaType media,
+                                         NetworkState state) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_