Add Audio support in Chromoting host when using WebRTC.

remoting/protocol/webrtc_connection_to_client.cc

Index: remoting/protocol/webrtc_connection_to_client.cc
diff --git a/remoting/protocol/webrtc_connection_to_client.cc b/remoting/protocol/webrtc_connection_to_client.cc
index 895ee3403597fbd17ad7310dee0e2e3e36df4297..be663ab88605cff06a7995bf4fb80307fb0e0228 100644
--- a/remoting/protocol/webrtc_connection_to_client.cc
+++ b/remoting/protocol/webrtc_connection_to_client.cc
@@ -21,6 +21,7 @@
 #include "remoting/protocol/input_stub.h"
 #include "remoting/protocol/message_pipe.h"
 #include "remoting/protocol/transport_context.h"
+#include "remoting/protocol/webrtc_audio_stream.h"
 #include "remoting/protocol/webrtc_transport.h"
 #include "remoting/protocol/webrtc_video_stream.h"
 #include "third_party/webrtc/api/mediastreaminterface.h"
@@ -37,13 +38,15 @@ namespace protocol {
 WebrtcConnectionToClient::WebrtcConnectionToClient(
     std::unique_ptr<protocol::Session> session,
     scoped_refptr<protocol::TransportContext> transport_context,
-    scoped_refptr<base::SingleThreadTaskRunner> video_encode_task_runner)
+    scoped_refptr<base::SingleThreadTaskRunner> video_encode_task_runner,
+    scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner)
     : transport_(
           new WebrtcTransport(jingle_glue::JingleThreadWrapper::current(),
                               transport_context,
                               this)),
       session_(std::move(session)),
       video_encode_task_runner_(video_encode_task_runner),
+      audio_task_runner_(audio_task_runner),
       control_dispatcher_(new HostControlDispatcher()),
       event_dispatcher_(new HostEventDispatcher()),
       weak_factory_(this) {
@@ -84,8 +87,12 @@ std::unique_ptr<VideoStream> WebrtcConnectionToClient::StartVideoStream(
 
 std::unique_ptr<AudioStream> WebrtcConnectionToClient::StartAudioStream(
     std::unique_ptr<AudioSource> audio_source) {
-  NOTIMPLEMENTED();
-  return nullptr;
+  std::unique_ptr<WebrtcAudioStream> stream(new WebrtcAudioStream());
+  if (!stream->Start(audio_task_runner_, std::move(audio_source),
+                     transport_.get())) {
+    return nullptr;

[Jamie, 2016/10/04 23:36:06]

Do we get an error log in Start()? If not then please add one here.

[Sergey Ulanov, 2016/10/05 21:52:23]

Actually Start() is not expected to fail. Updated it to return void and added
DCHECK in Start().
Also updated WebrtcVideoStream for consistency

+  }
+  return std::move(stream);
 }
 
 // Return pointer to ClientStub.