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/*
* 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_mixer.h"
#include "rtc_base/refcount.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
class AudioProcessing;
class AudioTransport;
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 audio mixer connected to active receive streams. One per
// AudioState.
rtc::scoped_refptr<AudioMixer> audio_mixer;
// The audio processing module.
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
};
virtual AudioProcessing* audio_processing() = 0;
virtual AudioTransport* audio_transport() = 0;
// Enable/disable playout of the audio channels. Enabled by default.
// This will stop playout of the underlying audio device but start a task
// which will poll for audio data every 10ms to ensure that audio processing
// happens and the audio stats are updated.
virtual void SetPlayout(bool enabled) = 0;
// Enable/disable recording of the audio channels. Enabled by default.
// This will stop recording of the underlying audio device and no audio
// packets will be encoded or transmitted.
virtual void SetRecording(bool enabled) = 0;
// 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 // CALL_AUDIO_STATE_H_
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