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