Revert of MediaStream audio object graph untangling and clean-ups.

content/renderer/media/webrtc/peer_connection_dependency_factory.h

 // Copyright 2014 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_
 
 #include <string>
 
 #include "base/files/file.h"
@@ skipping 134 matching lines @@
 
  protected:
   // Asks the PeerConnection factory to create a Local Audio Source.
   virtual scoped_refptr<webrtc::AudioSourceInterface>
       CreateLocalAudioSource(
           const webrtc::MediaConstraintsInterface* constraints);
 
   // Creates a media::AudioCapturerSource with an implementation that is
   // specific for a WebAudio source. The created WebAudioCapturerSource
   // instance will function as audio source instead of the default
-  // WebRtcAudioCapturer. Ownership of the new WebAudioCapturerSource is
-  // transferred to |source|.
-  virtual void CreateWebAudioSource(blink::WebMediaStreamSource* source);
+  // WebRtcAudioCapturer.
+  virtual scoped_refptr<WebAudioCapturerSource> CreateWebAudioSource(
+      blink::WebMediaStreamSource* source);
 
   // Asks the PeerConnection factory to create a Local VideoTrack object with
   // the video source using |capturer|.
   virtual scoped_refptr<webrtc::VideoTrackInterface>
       CreateLocalVideoTrack(const std::string& id,
                             cricket::VideoCapturer* capturer);
 
   virtual const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
       GetPcFactory();
   virtual bool PeerConnectionFactoryCreated();
 
   // Returns a new capturer or existing capturer based on the |render_frame_id|
   // and |device_info|; if both are valid, it reuses existing capture if any --
   // otherwise it creates a new capturer.
-  virtual scoped_ptr<WebRtcAudioCapturer> CreateAudioCapturer(
+  virtual scoped_refptr<WebRtcAudioCapturer> CreateAudioCapturer(
       int render_frame_id,
       const StreamDeviceInfo& device_info,
       const blink::WebMediaConstraints& constraints,
       MediaStreamAudioSource* audio_source);
 
+  // Adds the audio device as a sink to the audio track and starts the local
+  // audio track. This is virtual for test purposes since no real audio device
+  // exist in unit tests.
+  virtual void StartLocalAudioTrack(WebRtcLocalAudioTrack* audio_track);
+
  private:
   // Implement base::MessageLoop::DestructionObserver.
   // This makes sure the libjingle PeerConnectionFactory is released before
   // the renderer message loop is destroyed.
   void WillDestroyCurrentMessageLoop() override;
 
   // Functions related to Stun probing trial to determine how fast we could send
   // Stun request without being dropped by NAT.
   void TryScheduleStunProbeTrial();
   void StartStunProbeTrialOnWorkerThread(const std::string& params);
@@ skipping 34 matching lines @@
   rtc::Thread* worker_thread_;
   base::Thread chrome_signaling_thread_;
   base::Thread chrome_worker_thread_;
 
   DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_