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 16 matching lines @@
 
 const char* Version();
 
 enum class MediaType {
   ANY,
   AUDIO,
   VIDEO,
   DATA
 };
 
+enum class ChannelNetworkState

[the sun, 2016/03/04 12:40:48]

I don't think you need this enum. The CHANNEL_NOT_PRESENT state is implicit when
we don't have streams of a certain kind.

You can figure out in Call::SignalChannelNetworkState() what the aggregate state
is, by just keeping track of the audio/video network states separately and
looking at the stream counts. Note that when adding/removing streams, you will
potentially have to resignal the aggregate state to the congestion controller.

+{
+  CHANNEL_NOT_PRESENT,
+  CHANNEL_NETWORK_DOWN,
+  CHANNEL_NETWORK_UP
+};
+
+NetworkState ChannelStateToNetworkState(ChannelNetworkState channelState);
+NetworkState AggregateNetworkState(ChannelNetworkState channelA,
+                                   ChannelNetworkState channelB);
+
 class PacketReceiver {
  public:
   enum DeliveryStatus {
     DELIVERY_OK,
     DELIVERY_UNKNOWN_SSRC,
     DELIVERY_PACKET_ERROR,
   };
 
   virtual DeliveryStatus DeliverPacket(MediaType media_type,
                                        const uint8_t* packet,
@@ skipping 79 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;
+
+  virtual void SignalChannelNetworkState(MediaType media,
+      ChannelNetworkState state) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_