Ensures that WebRTC works for device selection using a different sample rate than default

content/renderer/media/media_stream_dependency_factory.cc

Index: content/renderer/media/media_stream_dependency_factory.cc
diff --git a/content/renderer/media/media_stream_dependency_factory.cc b/content/renderer/media/media_stream_dependency_factory.cc
index 31072a2149b68cdeb0b7153c56702710823ad30a..14cbf4387dd4919c1d15ff27ac8f23f81f8cf000 100644
--- a/content/renderer/media/media_stream_dependency_factory.cc
+++ b/content/renderer/media/media_stream_dependency_factory.cc
@@ -190,6 +190,7 @@ void MediaStreamDependencyFactory::CreateNativeMediaSources(
     const WebKit::WebMediaConstraints& video_constraints,
     WebKit::WebMediaStreamDescriptor* description,
     const MediaSourcesCreatedCallback& sources_created) {
+  DVLOG(1) << "MediaStreamDependencyFactory::CreateNativeMediaSources()";
   if (!EnsurePeerConnectionFactory()) {
     sources_created.Run(description, false);
     return;
@@ -200,8 +201,6 @@ void MediaStreamDependencyFactory::CreateNativeMediaSources(
   SourceStateObserver* source_observer =
       new SourceStateObserver(description, sources_created);
 
-  // TODO(perkj): Implement local audio sources.
-
   // Create local video sources.
   RTCMediaConstraints native_video_constraints(video_constraints);
   WebKit::WebVector<WebKit::WebMediaStreamComponent> video_components;
@@ -223,11 +222,37 @@ void MediaStreamDependencyFactory::CreateNativeMediaSources(
                           &native_video_constraints));
     source_observer->AddSource(source_data->video_source());
   }
+
+  // Do additional source initialization if the audio source is a valid
+  // microphone.
+  WebKit::WebVector<WebKit::WebMediaStreamComponent> audio_components;
+  description->audioSources(audio_components);
+  for (size_t i = 0; i < audio_components.size(); ++i) {
+    const WebKit::WebMediaStreamSource& source = audio_components[i].source();
+    MediaStreamSourceExtraData* source_data =
+        static_cast<MediaStreamSourceExtraData*>(source.extraData());
+    if (!source_data) {
+      // TODO(henrika): Implement support for sources from remote MediaStreams.
+      NOTIMPLEMENTED();
+      continue;
+    }
+
+    const StreamDeviceInfo device_info = source_data->device_info();
+    if (device_info.device.type == content::MEDIA_DEVICE_AUDIO_CAPTURE) {
+      if (!InitializeAudioSource(device_info)) {
+        DLOG(WARNING) << "Unsupported audio source";
+        sources_created.Run(description, false);
+        return;
+      }
+    }
+  }
+
   source_observer->StartObservering();
 }
 
 void MediaStreamDependencyFactory::CreateNativeLocalMediaStream(
     WebKit::WebMediaStreamDescriptor* description) {
+  DVLOG(1) << "MediaStreamDependencyFactory::CreateNativeLocalMediaStream()";
   DCHECK(PeerConnectionFactoryCreated());
 
   std::string label = UTF16ToUTF8(description->label());
@@ -253,10 +278,6 @@ void MediaStreamDependencyFactory::CreateNativeLocalMediaStream(
         CreateLocalAudioTrack(UTF16ToUTF8(source.id()), NULL));
     native_stream->AddTrack(audio_track);
     audio_track->set_enabled(audio_components[i].isEnabled());
-    // TODO(xians): This set the source of all audio tracks to the same
-    // microphone. Implement support for setting the source per audio track
-    // instead.
-    SetAudioDeviceSessionId(source_data->device_info().session_id);
   }
 
   // Add video tracks.
@@ -295,6 +316,7 @@ void MediaStreamDependencyFactory::CreateNativeLocalMediaStream(
 }
 
 bool MediaStreamDependencyFactory::CreatePeerConnectionFactory() {
+  DVLOG(1) << "MediaStreamDependencyFactory::CreatePeerConnectionFactory()";
   if (!pc_factory_.get()) {
     DCHECK(!audio_device_);
     audio_device_ = new WebRtcAudioDeviceImpl();
@@ -352,6 +374,32 @@ MediaStreamDependencyFactory::CreateVideoSource(
   return source;
 }
 
+bool MediaStreamDependencyFactory::InitializeAudioSource(
+  const StreamDeviceInfo& device_info) {
+  DVLOG(1) << "MediaStreamDependencyFactory::InitializeAudioSource()";
+  const MediaStreamDevice device = device_info.device;
+
+  // Initialize the source using audio parameters for the selected
+  // capture device.
+  WebRtcAudioCapturer* capturer = GetWebRtcAudioDevice()->capturer();
+  // TODO(henrika): refactor \content\public\common\media_stream_request.h
+  // to allow dependency of media::ChannelLayout and avoid static_cast.
+  if (!capturer->Initialize(
+          static_cast<media::ChannelLayout>(device.channel_layout),
+          device.sample_rate))
+    return false;
+
+  // Specify which capture device to use. The acquired session id is used
+  // for identification.
+  // TODO(henrika): the current design does not support a uniqe source
+  // for each audio track.
+  if (device_info.session_id <= 0)
+    return false;
+
+  capturer->SetDevice(device_info.session_id);
+  return true;
+}
+
 scoped_refptr<webrtc::VideoTrackInterface>
 MediaStreamDependencyFactory::CreateLocalVideoTrack(
     const std::string& label,
@@ -384,10 +432,6 @@ MediaStreamDependencyFactory::GetWebRtcAudioDevice() {
   return audio_device_;
 }
 
-void MediaStreamDependencyFactory::SetAudioDeviceSessionId(int session_id) {
-  audio_device_->SetSessionId(session_id);
-}
-
 void MediaStreamDependencyFactory::InitializeWorkerThread(
     talk_base::Thread** thread,
     base::WaitableEvent* event) {