Make the audio channel communicate network state changes to the call.
This change enables voice-only calls to keep track of the network state.
This is only a partial fix - the last modality to change state controls
the state for the entire call, so a call with a failed video transport
will also stop sending audio packets. Handling this condition correctly
would require the call to keep track of network state for each media
type separately, and take care of conditions such as a failed video
channel getting removed, while a functioning audio channel remains.
BUG=webrtc:5307
Review URL: https://codereview.webrtc.org/1757683002
Cr-Commit-Position: refs/heads/master@{#12093}
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index a16d042..056b078 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -183,7 +183,7 @@
const rtc::PacketTime& packet_time) override;
void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) override;
- void OnReadyToSend(bool ready) override {}
+ void OnReadyToSend(bool ready) override;
bool GetStats(VoiceMediaInfo* info) override;
void SetRawAudioSink(