MediaStream audio object graph untangling and clean-ups.

content/renderer/media/media_stream_audio_source.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_MEDIA_STREAM_AUDIO_SOURCE_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_
 
 #include "base/compiler_specific.h"
 #include "base/macros.h"
+#include "base/memory/scoped_ptr.h"
 #include "content/common/content_export.h"
 #include "content/renderer/media/media_stream_source.h"
 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "third_party/webrtc/api/mediastreaminterface.h"
 
 namespace content {
 
+// TODO(miu): In a soon-upcoming set of refactoring changes, this class will
+// become a base class for managing tracks (part of what WebRtcAudioCapturer
+// does today).  Then, the rest of WebRtcAudioCapturer will be rolled into a
+// subclass.   http://crbug.com/577874
 class CONTENT_EXPORT MediaStreamAudioSource
     : NON_EXPORTED_BASE(public MediaStreamSource) {
  public:
   MediaStreamAudioSource(int render_frame_id,
                          const StreamDeviceInfo& device_info,
                          const SourceStoppedCallback& stop_callback,
                          PeerConnectionDependencyFactory* factory);
   MediaStreamAudioSource();
   ~MediaStreamAudioSource() override;
 
+  // Returns the MediaStreamAudioSource instance owned by the given blink
+  // |source| or null.
+  static MediaStreamAudioSource* From(const blink::WebMediaStreamSource& track);
+
   void AddTrack(const blink::WebMediaStreamTrack& track,
                 const blink::WebMediaConstraints& constraints,
                 const ConstraintsCallback& callback);
 
-  void SetLocalAudioSource(webrtc::AudioSourceInterface* source) {
-    local_audio_source_ = source;
-  }
+  WebRtcAudioCapturer* audio_capturer() const { return audio_capturer_.get(); }
 
-  void SetAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer) {
+  void SetAudioCapturer(scoped_ptr<WebRtcAudioCapturer> capturer) {
     DCHECK(!audio_capturer_.get());
-    audio_capturer_ = capturer;
-  }
-
-  const scoped_refptr<WebRtcAudioCapturer>& GetAudioCapturer() {
-    return audio_capturer_;
+    audio_capturer_ = std::move(capturer);
   }
 
   webrtc::AudioSourceInterface* local_audio_source() {
     return local_audio_source_.get();
   }
 
+  void SetLocalAudioSource(scoped_refptr<webrtc::AudioSourceInterface> source) {
+    local_audio_source_ = source;
+  }
+
  protected:
   void DoStopSource() override;
 
  private:
   const int render_frame_id_;
   PeerConnectionDependencyFactory* const factory_;
 
+  scoped_ptr<WebRtcAudioCapturer> audio_capturer_;
+
   // This member holds an instance of webrtc::LocalAudioSource. This is used
   // as a container for audio options.
   scoped_refptr<webrtc::AudioSourceInterface> local_audio_source_;
 
-  scoped_refptr<WebRtcAudioCapturer> audio_capturer_;
-
   DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioSource);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_