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

content/renderer/media/media_stream_dependency_factory.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/media_stream_dependency_factory.h"
 
 #include <vector>
 
 #include "base/synchronization/waitable_event.h"
 #include "base/utf_string_conversions.h"
@@ skipping 172 matching lines @@
     return NULL;
 
   return new RTCPeerConnectionHandler(client, this);
 }
 
 void MediaStreamDependencyFactory::CreateNativeMediaSources(
     const WebKit::WebMediaConstraints& audio_constraints,
     const WebKit::WebMediaConstraints& video_constraints,
     WebKit::WebMediaStreamDescriptor* description,
     const MediaSourcesCreatedCallback& sources_created) {
+  DVLOG(1) << "MediaStreamDependencyFactory::CreateNativeMediaSources()";
   if (!EnsurePeerConnectionFactory()) {
     sources_created.Run(description, false);
     return;
   }
 
   // |source_observer| clean up itself when it has completed
   // source_observer->StartObservering.
   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;
   description->videoSources(video_components);
   for (size_t i = 0; i < video_components.size(); ++i) {
     const WebKit::WebMediaStreamSource& source = video_components[i].source();
     MediaStreamSourceExtraData* source_data =
         static_cast<MediaStreamSourceExtraData*>(source.extraData());
     if (!source_data) {
       // TODO(perkj): Implement support for sources from remote MediaStreams.
       NOTIMPLEMENTED();
       continue;
     }
     const bool is_screencast = (source_data->device_info().device.type ==
         content::MEDIA_TAB_VIDEO_CAPTURE);
     source_data->SetVideoSource(
         CreateVideoSource(source_data->device_info().session_id,
                           is_screencast,
                           &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());
   scoped_refptr<webrtc::LocalMediaStreamInterface> native_stream =
       CreateLocalMediaStream(label);
 
   // Add audio tracks.
   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(perkj): Implement support for sources from remote MediaStreams.
       NOTIMPLEMENTED();
       continue;
     }
     // TODO(perkj): Refactor the creation of audio tracks to use a proper
     // interface for receiving audio input data. Currently NULL is passed since
     // the |audio_device| is the wrong class and is unused.
     scoped_refptr<webrtc::LocalAudioTrackInterface> audio_track(
         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.
   WebKit::WebVector<WebKit::WebMediaStreamComponent> video_components;
   description->videoSources(video_components);
   for (size_t i = 0; i < video_components.size(); ++i) {
     const WebKit::WebMediaStreamSource& source = video_components[i].source();
     MediaStreamSourceExtraData* source_data =
         static_cast<MediaStreamSourceExtraData*>(source.extraData());
     if (!source_data || !source_data->video_source()) {
@@ skipping 18 matching lines @@
     WebKit::WebMediaStreamDescriptor* description,
     const MediaStreamExtraData::StreamStopCallback& stream_stop) {
   CreateNativeLocalMediaStream(description);
 
   MediaStreamExtraData* extra_data =
      static_cast<MediaStreamExtraData*>(description->extraData());
   extra_data->SetLocalStreamStopCallback(stream_stop);
 }
 
 bool MediaStreamDependencyFactory::CreatePeerConnectionFactory() {
+  DVLOG(1) << "MediaStreamDependencyFactory::CreatePeerConnectionFactory()";
   if (!pc_factory_.get()) {
     DCHECK(!audio_device_);
     audio_device_ = new WebRtcAudioDeviceImpl();
     scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory(
         webrtc::CreatePeerConnectionFactory(worker_thread_,
                                             signaling_thread_,
                                             audio_device_));
     if (factory.get())
       pc_factory_ = factory;
     else
@@ skipping 37 matching lines @@
     const webrtc::MediaConstraintsInterface* constraints) {
   RtcVideoCapturer* capturer = new RtcVideoCapturer(
       video_session_id, vc_manager_.get(), is_screencast);
 
   // The video source takes ownership of |capturer|.
   scoped_refptr<webrtc::VideoSourceInterface> source =
       pc_factory_->CreateVideoSource(capturer, constraints).get();
   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,
     webrtc::VideoSourceInterface* source) {
   return pc_factory_->CreateVideoTrack(label, source).get();
 }
 
 scoped_refptr<webrtc::LocalAudioTrackInterface>
 MediaStreamDependencyFactory::CreateLocalAudioTrack(
     const std::string& label,
@@ skipping 12 matching lines @@
     int sdp_mline_index,
     const std::string& sdp) {
   return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp);
 }
 
 WebRtcAudioDeviceImpl*
 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) {
   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
   *thread = jingle_glue::JingleThreadWrapper::current();
   event->Signal();
 }
 
 void MediaStreamDependencyFactory::CreateIpcNetworkManagerOnWorkerThread(
@@ skipping 78 matching lines @@
       // processed before returning. We wait for the above task to finish before
       // letting the the function continue to avoid any potential race issues.
       chrome_worker_thread_.Stop();
     } else {
       NOTREACHED() << "Worker thread not running.";
     }
   }
 }
 
 }  // namespace content