blob: ac912773aa5fdeb12a59120b794c269605614050 [file] [log] [blame]
/*
* 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 WEBRTC_API_CALL_AUDIO_STATE_H_
#define WEBRTC_API_CALL_AUDIO_STATE_H_
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
namespace webrtc {
class AudioDeviceModule;
class VoiceEngine;
// WORK IN PROGRESS
// This class is under development and is not yet intended for for use outside
// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
// AudioState holds the state which must be shared between multiple instances of
// webrtc::Call for audio processing purposes.
class AudioState : public rtc::RefCountInterface {
public:
struct Config {
// VoiceEngine used for audio streams and audio/video synchronization.
// AudioState will tickle the VoE refcount to keep it alive for as long as
// the AudioState itself.
VoiceEngine* voice_engine = nullptr;
// The AudioDeviceModule associated with the Calls.
AudioDeviceModule* audio_device_module = nullptr;
};
// TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
static rtc::scoped_refptr<AudioState> Create(
const AudioState::Config& config);
virtual ~AudioState() {}
};
} // namespace webrtc
#endif // WEBRTC_API_CALL_AUDIO_STATE_H_