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// 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"
#include "third_party/webrtc/api/scoped_refptr.h"
namespace base {
class SingleThreadTaskRunner;
} // namespace webrtc
namespace webrtc {
class PeerConnectionInterface;
class RtpSenderInterface;
} // namespace webrtc
namespace remoting {
namespace protocol {
class AudioSource;
class WebrtcAudioSourceAdapter;
class WebrtcTransport;
class WebrtcAudioStream : public AudioStream {
public:
WebrtcAudioStream();
~WebrtcAudioStream() override;
void 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_;
rtc::scoped_refptr<webrtc::RtpSenderInterface> audio_sender_;
DISALLOW_COPY_AND_ASSIGN(WebrtcAudioStream);
};
} // namespace protocol
} // namespace remoting
#endif // REMOTING_PROTOCOL_WEBRTC_AUDIO_STREAM_H_