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

webrtc/call.h

Index: webrtc/call.h
diff --git a/webrtc/call.h b/webrtc/call.h
index 313c5e58c16f49446c8d6fbcdaa92950179628a8..57b8ef6cea4cb266045f235c06c37e5921043940 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -34,6 +34,17 @@ enum class MediaType {
   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 {
@@ -133,7 +144,9 @@ class Call {
   // 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;