Cleaned up the WebRtcAudioCapturer a bit.

content/renderer/media/rtc_peer_connection_handler.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/rtc_peer_connection_handler.h"
 
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/logging.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/stl_util.h"
 #include "base/strings/utf_string_conversions.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_dependency_factory.h"
+#include "content/renderer/media/media_stream_source_extra_data.h"
 #include "content/renderer/media/peer_connection_tracker.h"
 #include "content/renderer/media/remote_media_stream_impl.h"
 #include "content/renderer/media/rtc_data_channel_handler.h"
 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/render_thread_impl.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 // TODO(hta): Move the following include to WebRTCStatsRequest.h file.
@@ skipping 514 matching lines @@
 bool RTCPeerConnectionHandler::addStream(
     const blink::WebMediaStream& stream,
     const blink::WebMediaConstraints& options) {
   RTCMediaConstraints constraints(options);
 
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackAddStream(
         this, stream, PeerConnectionTracker::SOURCE_LOCAL);
 
   // A media stream is connected to a peer connection, enable the
-  // peer connection mode for the capturer.
-  WebRtcAudioDeviceImpl* audio_device =
-      dependency_factory_->GetWebRtcAudioDevice();
-  if (audio_device) {
-    WebRtcAudioCapturer* capturer = audio_device->GetDefaultCapturer();
-    if (capturer)
-      capturer->EnablePeerConnectionMode();
+  // peer connection mode for the sources.
+  blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
+  stream.audioTracks(audio_tracks);
+  for (size_t i = 0; i < audio_tracks.size(); ++i) {
+    const blink::WebMediaStreamSource& source = audio_tracks[i].source();
+    MediaStreamSourceExtraData* extra_data =
+        static_cast<MediaStreamSourceExtraData*>(source.extraData());
+    if (extra_data->GetAudioCapturer())
+      extra_data->GetAudioCapturer()->EnablePeerConnectionMode();
   }
 
   return AddStream(stream, &constraints);
 }
 
 void RTCPeerConnectionHandler::removeStream(
     const blink::WebMediaStream& stream) {
   RemoveStream(stream);
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackRemoveStream(
@@ skipping 229 matching lines @@
   webrtc::SessionDescriptionInterface* native_desc =
       dependency_factory_->CreateSessionDescription(type, sdp, error);
 
   LOG_IF(ERROR, !native_desc) << "Failed to create native session description."
                               << " Type: " << type << " SDP: " << sdp;
 
   return native_desc;
 }
 
 }  // namespace content