| Index: content/renderer/media/media_stream_audio_source.h | 
| diff --git a/content/renderer/media/media_stream_audio_source.h b/content/renderer/media/media_stream_audio_source.h | 
| new file mode 100644 | 
| index 0000000000000000000000000000000000000000..2e73395f7b5acc55abb303b9d41e5667b5105446 | 
| --- /dev/null | 
| +++ b/content/renderer/media/media_stream_audio_source.h | 
| @@ -0,0 +1,67 @@ | 
| +// 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 "content/common/content_export.h" | 
| +#include "content/renderer/media/media_stream_dependency_factory.h" | 
| +#include "content/renderer/media/media_stream_source.h" | 
| +#include "content/renderer/media/webrtc_audio_capturer.h" | 
| +#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 
| + | 
| +namespace content { | 
| + | 
| +class CONTENT_EXPORT MediaStreamAudioSource | 
| +    : NON_EXPORTED_BASE(public MediaStreamSource) { | 
| + public: | 
| +  MediaStreamAudioSource(int render_view_id, | 
| +                         const StreamDeviceInfo& device_info, | 
| +                         const SourceStoppedCallback& stop_callback, | 
| +                         MediaStreamDependencyFactory* factory); | 
| +  MediaStreamAudioSource(); | 
| +  virtual ~MediaStreamAudioSource(); | 
| + | 
| +  virtual void AddTrack(const blink::WebMediaStreamTrack& track, | 
| +                        const blink::WebMediaConstraints& constraints, | 
| +                        const ConstraintsCallback& callback) OVERRIDE; | 
| +  virtual void RemoveTrack(const blink::WebMediaStreamTrack& track) OVERRIDE; | 
| + | 
| +  void SetLocalAudioSource(webrtc::AudioSourceInterface* source) { | 
| +    local_audio_source_ = source; | 
| +  } | 
| + | 
| +  void SetAudioCapturer(WebRtcAudioCapturer* capturer) { | 
| +    DCHECK(!audio_capturer_); | 
| +    audio_capturer_ = capturer; | 
| +  } | 
| + | 
| +  const scoped_refptr<WebRtcAudioCapturer>& GetAudioCapturer() { | 
| +    return audio_capturer_; | 
| +  } | 
| + | 
| +  webrtc::AudioSourceInterface* local_audio_source() { | 
| +    return local_audio_source_.get(); | 
| +  } | 
| + | 
| + protected: | 
| +  virtual void DoStopSource() OVERRIDE; | 
| + | 
| + private: | 
| +  int render_view_id_;  // Render view ID that created this source. | 
| +  // 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_; | 
| + | 
| +  MediaStreamDependencyFactory* factory_; | 
| + | 
| +  DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioSource); | 
| +}; | 
| + | 
| +}  // namespace content | 
| + | 
| +#endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_ | 
|  |