blob: 826b31d12a03a1804dd869f5c7647c79c38503a1 [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.
#include "webrtc/api/audio/audio_mixer.h"
#include "webrtc/rtc_base/refcount.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
namespace webrtc {
class AudioProcessing;
class VoiceEngine;
// This class is under development and is not yet intended for for use outside
// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
// See:
// AudioState holds the state which must be shared between multiple instances of
// webrtc::Call for audio processing purposes.
class AudioState : public rtc::RefCountInterface {
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;
// 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