blob: 79fb5cf981a5598f365ef5528acfc71adb539f78 [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 "api/audio/audio_mixer.h"
#include "api/scoped_refptr.h"
#include "modules/async_audio_processing/async_audio_processing.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/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 rtc::RefCountInterface {
struct Config {
// 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;
// TODO(solenberg): Temporary: audio device module.
rtc::scoped_refptr<webrtc::AudioDeviceModule> audio_device_module;
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;
virtual void SetStereoChannelSwapping(bool enable) = 0;
static rtc::scoped_refptr<AudioState> Create(
const AudioState::Config& config);
~AudioState() override {}
} // namespace webrtc