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

webrtc/media/engine/webrtcvideoengine2.cc

Index: webrtc/media/engine/webrtcvideoengine2.cc
diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc
index 0e9866064df142cf7d1529065a818560a674dea3..65cee806da048d450e286948f8b5465ed5edd973 100644
--- a/webrtc/media/engine/webrtcvideoengine2.cc
+++ b/webrtc/media/engine/webrtcvideoengine2.cc
@@ -625,6 +625,9 @@ WebRtcVideoChannel2::WebRtcVideoChannel2(
   default_send_ssrc_ = 0;
   RTC_DCHECK(ValidateCodecFormats(recv_codecs));
   recv_codecs_ = FilterSupportedCodecs(MapCodecs(recv_codecs));
+  call_->SignalChannelNetworkState(
+    webrtc::MediaType::VIDEO,
+    webrtc::ChannelNetworkState::CHANNEL_NETWORK_UP);

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

I'd think Call can already figure this out since it knows whether any
video/audio streams are available. Why do we need to signal it?

 }
 
 WebRtcVideoChannel2::~WebRtcVideoChannel2() {
@@ -632,6 +635,9 @@ WebRtcVideoChannel2::~WebRtcVideoChannel2() {
     delete kv.second;
   for (auto& kv : receive_streams_)
     delete kv.second;
+  call_->SignalChannelNetworkState(

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

Likewise.

+    webrtc::MediaType::VIDEO,
+    webrtc::ChannelNetworkState::CHANNEL_NOT_PRESENT);
 }
 
 bool WebRtcVideoChannel2::CodecIsExternallySupported(
@@ -1360,7 +1366,11 @@ void WebRtcVideoChannel2::OnRtcpReceived(
 
 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,
+    ready ?
+    webrtc::ChannelNetworkState::CHANNEL_NETWORK_UP :
+    webrtc::ChannelNetworkState::CHANNEL_NETWORK_DOWN);
 }
 
 bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) {