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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 "api/scoped_refptr.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 {
- public:
- struct Config {
- Config();
- ~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
- #endif // CALL_AUDIO_STATE_H_
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