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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/call/audio_state.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/call/audio_state.h')
-rw-r--r-- | third_party/libwebrtc/call/audio_state.h | 69 |
1 files changed, 69 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/audio_state.h b/third_party/libwebrtc/call/audio_state.h new file mode 100644 index 0000000000..79fb5cf981 --- /dev/null +++ b/third_party/libwebrtc/call/audio_state.h @@ -0,0 +1,69 @@ +/* + * 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/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 { + 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; + + rtc::scoped_refptr<AsyncAudioProcessing::Factory> + async_audio_processing_factory; + }; + + 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_ |