chromium/third_party/webrtc/call/audio_state.h

/*
 *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */
#ifndef CALL_AUDIO_STATE_H_
#define CALL_AUDIO_STATE_H_

#include "api/audio/audio_device.h"
#include "api/audio/audio_mixer.h"
#include "api/audio/audio_processing.h"
#include "api/scoped_refptr.h"
#include "modules/async_audio_processing/async_audio_processing.h"
#include "rtc_base/ref_count.h"

namespace webrtc {

class AudioTransport;

// AudioState holds the state which must be shared between multiple instances of
// webrtc::Call for audio processing purposes.
class AudioState : public RefCountInterface { … };
}  // namespace webrtc

#endif  // CALL_AUDIO_STATE_H_