| Index: remoting/protocol/webrtc_audio_stream.h
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| diff --git a/remoting/protocol/webrtc_audio_stream.h b/remoting/protocol/webrtc_audio_stream.h
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| new file mode 100644
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| index 0000000000000000000000000000000000000000..39e87e021a092e84142ef59963fe56643b16daa6
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| --- /dev/null
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| +++ b/remoting/protocol/webrtc_audio_stream.h
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| @@ -0,0 +1,54 @@
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| +// Copyright 2016 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 REMOTING_PROTOCOL_WEBRTC_AUDIO_STREAM_H_
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| +#define REMOTING_PROTOCOL_WEBRTC_AUDIO_STREAM_H_
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| +
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| +#include <memory>
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| +
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| +#include "base/macros.h"
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| +#include "base/memory/ref_counted.h"
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| +#include "remoting/protocol/audio_stream.h"
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| +
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| +namespace base {
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| +class SingleThreadTaskRunner;
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| +}  // namespace webrtc
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| +
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| +namespace webrtc {
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| +class MediaStreamInterface;
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| +class PeerConnectionInterface;
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| +}  // namespace webrtc
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| +
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| +namespace remoting {
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| +namespace protocol {
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| +
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| +class AudioSource;
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| +class WebrtcAudioSourceAdapter;
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| +class WebrtcTransport;
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| +
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| +class WebrtcAudioStream : public AudioStream {
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| + public:
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| +  WebrtcAudioStream();
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| +  ~WebrtcAudioStream() override;
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| +
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| +  bool Start(scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner,
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| +             std::unique_ptr<AudioSource> audio_source,
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| +             WebrtcTransport* webrtc_transport);
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| +
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| +  // AudioStream interface.
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| +  void Pause(bool pause) override;
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| +
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| + private:
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| +  scoped_refptr<WebrtcAudioSourceAdapter> source_adapter_;
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| +
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| +  scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
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| +  scoped_refptr<webrtc::MediaStreamInterface> stream_;
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| +
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| +  DISALLOW_COPY_AND_ASSIGN(WebrtcAudioStream);
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| +};
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| +
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| +}  // namespace protocol
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| +}  // namespace remoting
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| +
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| +#endif  // REMOTING_PROTOCOL_WEBRTC_AUDIO_STREAM_H_
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| 
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