chromium/remoting/protocol/webrtc_audio_module.h

// Copyright 2016 The Chromium Authors
// 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_MODULE_H_
#define REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_

#include "base/memory/raw_ptr.h"
#include "base/memory/scoped_refptr.h"
#include "base/synchronization/lock.h"
#include "third_party/webrtc/modules/audio_device/include/audio_device.h"

namespace base {
class RepeatingTimer;
class SingleThreadTaskRunner;
}  // namespace base

namespace remoting::protocol {

// Audio module passed to WebRTC. It doesn't access actual audio devices, but it
// provides all functionality we need to ensure that audio streaming works
// properly in WebRTC. Particularly it's responsible for calling AudioTransport
// on regular intervals when playback is active. This ensures that all incoming
// audio data is processed and passed to webrtc::AudioTrackSinkInterface
// connected to the audio track.
class WebrtcAudioModule : public webrtc::AudioDeviceModule { … };

}  // namespace remoting::protocol

#endif  // REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_