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

webrtc/media/engine/fakewebrtccall.cc

Index: webrtc/media/engine/fakewebrtccall.cc
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index 1af11afd24e63c553de34b8785f6762dbe3ac2e5..fb4c974001b404fc34c139289325465938d489f1 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -227,7 +227,8 @@ void FakeVideoReceiveStream::SetStats(
 
 FakeCall::FakeCall(const webrtc::Call::Config& config)
     : config_(config),
-      network_state_(webrtc::kNetworkUp),
+      audio_network_state_(webrtc::ChannelNetworkState::CHANNEL_NOT_PRESENT),
+      video_network_state_(webrtc::ChannelNetworkState::CHANNEL_NOT_PRESENT),
       num_created_send_streams_(0),
       num_created_receive_streams_(0) {}
 
@@ -276,8 +277,16 @@ const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
   return nullptr;
 }
 
-webrtc::NetworkState FakeCall::GetNetworkState() const {
-  return network_state_;
+webrtc::ChannelNetworkState FakeCall::GetNetworkState(
+      webrtc::MediaType media) const {
+  switch (media) {
+    case webrtc::MediaType::AUDIO:
+      return audio_network_state_;
+    case webrtc::MediaType::VIDEO:
+      return video_network_state_;
+    default:
+      return webrtc::ChannelNetworkState::CHANNEL_NOT_PRESENT;
+  }
 }
 
 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
@@ -293,7 +302,7 @@ void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
                       audio_send_streams_.end(),
                       static_cast<FakeAudioSendStream*>(send_stream));
   if (it == audio_send_streams_.end()) {
-    ADD_FAILURE() << "DestroyAudioSendStream called with unknown paramter.";
+    ADD_FAILURE() << "DestroyAudioSendStream called with unknown parameter.";
   } else {
     delete *it;
     audio_send_streams_.erase(it);
@@ -313,7 +322,7 @@ void FakeCall::DestroyAudioReceiveStream(
                       audio_receive_streams_.end(),
                       static_cast<FakeAudioReceiveStream*>(receive_stream));
   if (it == audio_receive_streams_.end()) {
-    ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown paramter.";
+    ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown parameter.";
   } else {
     delete *it;
     audio_receive_streams_.erase(it);
@@ -335,7 +344,7 @@ void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
                       video_send_streams_.end(),
                       static_cast<FakeVideoSendStream*>(send_stream));
   if (it == video_send_streams_.end()) {
-    ADD_FAILURE() << "DestroyVideoSendStream called with unknown paramter.";
+    ADD_FAILURE() << "DestroyVideoSendStream called with unknown parameter.";
   } else {
     delete *it;
     video_send_streams_.erase(it);
@@ -355,7 +364,7 @@ void FakeCall::DestroyVideoReceiveStream(
                       video_receive_streams_.end(),
                       static_cast<FakeVideoReceiveStream*>(receive_stream));
   if (it == video_receive_streams_.end()) {
-    ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown paramter.";
+    ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown parameter.";
   } else {
     delete *it;
     video_receive_streams_.erase(it);
@@ -416,8 +425,18 @@ void FakeCall::SetBitrateConfig(
   config_.bitrate_config = bitrate_config;
 }
 
-void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
-  network_state_ = state;
+void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
+     webrtc::ChannelNetworkState state) {
+  switch (media) {
+    case webrtc::MediaType::AUDIO:
+      audio_network_state_ = state;
+      break;
+    case webrtc::MediaType::VIDEO:
+      video_network_state_ = state;
+      break;
+    default:
+    ADD_FAILURE() << "SignalChannelNetworkState called with unknown parameter.";
+  }
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {