| Index: content/renderer/media/media_stream_audio_source.h
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| diff --git a/content/renderer/media/media_stream_audio_source.h b/content/renderer/media/media_stream_audio_source.h
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| new file mode 100644
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| index 0000000000000000000000000000000000000000..a8071f77ff6fedfa3f1d351d224acd6fe6f6f07f
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| --- /dev/null
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| +++ b/content/renderer/media/media_stream_audio_source.h
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| @@ -0,0 +1,57 @@
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| +// Copyright 2014 The Chromium Authors. All rights reserved.
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| +// Use of this source code is governed by a BSD-style license that can be
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| +// found in the LICENSE file.
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| +
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| +#ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_
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| +#define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_
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| +
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| +#include "base/compiler_specific.h"
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| +#include "content/common/content_export.h"
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| +#include "content/renderer/media/media_stream_source.h"
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| +#include "content/renderer/media/webrtc_audio_capturer.h"
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| +#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
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| +
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| +namespace content {
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| +
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| +class CONTENT_EXPORT MediaStreamAudioSource
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| +    : NON_EXPORTED_BASE(public MediaStreamSource) {
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| + public:
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| +  MediaStreamAudioSource(const StreamDeviceInfo& device_info,
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| +                         const SourceStoppedCallback& stop_callback);
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| +  MediaStreamAudioSource();
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| +  virtual ~MediaStreamAudioSource();
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| +
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| +  virtual void AddTrack(const blink::WebMediaStreamTrack& track,
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| +                        const blink::WebMediaConstraints& constraints,
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| +                        const ConstraintsCallback& callback) OVERRIDE;
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| +  virtual void RemoveTrack(const blink::WebMediaStreamTrack& track) OVERRIDE;
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| +
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| +  void SetLocalAudioSource(webrtc::AudioSourceInterface* source) {
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| +    local_audio_source_ = source;
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| +  }
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| +
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| +  void SetAudioCapturer(WebRtcAudioCapturer* capturer) {
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| +    DCHECK(!audio_capturer_);
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| +    audio_capturer_ = capturer;
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| +  }
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| +
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| +  webrtc::AudioSourceInterface* local_audio_source() {
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| +    return local_audio_source_.get();
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| +  }
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| +
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| + protected:
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| +  virtual void DoStopSource() OVERRIDE;
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| +
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| + private:
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| +  // This member holds an instance of webrtc::LocalAudioSource. This is used
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| +  // as a container for audio options.
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| +  scoped_refptr<webrtc::AudioSourceInterface> local_audio_source_;
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| +
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| +  scoped_refptr<WebRtcAudioCapturer> audio_capturer_;
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| +
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| +  DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioSource);
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| +};
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| +
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| +}  // namespace content
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| +
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| +#endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_
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| 
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