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Ring Daemon
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#include <webrtc.h>


Public Member Functions | |
| void | enableAutomaticGainControl (bool enabled) override |
| Set the status of automatic gain control. | |
| void | enableEchoCancel (bool enabled) override |
| Set the status of echo cancellation. | |
| void | enableNoiseSuppression (bool enabled) override |
| Set the status of noise suppression includes de-reverb, de-noise, high pass filter, etc. | |
| void | enableVoiceActivityDetection (bool enabled) override |
| Set the status of voice activity detection. | |
| std::shared_ptr< AudioFrame > | getProcessed () override |
| Process and return a single AudioFrame. | |
| WebRTCAudioProcessor (AudioFormat format, unsigned frameSize) | |
| ~WebRTCAudioProcessor ()=default | |
Public Member Functions inherited from jami::AudioProcessor | |
| AudioProcessor (AudioFormat format, unsigned frameSize) | |
| virtual void | putPlayback (const std::shared_ptr< AudioFrame > &buf) |
| virtual void | putRecorded (std::shared_ptr< AudioFrame > &&buf) |
| virtual | ~AudioProcessor ()=default |
Additional Inherited Members | |
Protected Member Functions inherited from jami::AudioProcessor | |
| bool | getStabilizedVoiceActivity (bool voiceStatus) |
| Stablilizes voice activity. | |
| bool | tidyQueues () |
| Helper method for audio processors, should be called at start of getProcessed() Pops frames from audio queues if there's overflow. | |
Protected Attributes inherited from jami::AudioProcessor | |
| unsigned int | consecutiveActiveFrames {0} |
| unsigned int | forceMinimumVoiceActivityMs {1000} |
| unsigned int | forceVoiceActiveFramesLeft {0} |
| AudioFormat | format_ |
| unsigned int | frameDurationMs_ |
| unsigned int | frameSize_ |
| std::unique_ptr< Resampler > | inputResampler_ |
| unsigned int | minimumConsequtiveDurationMs {200} |
| std::unique_ptr< Resampler > | outputResampler_ |
| AudioFrameResizer | playbackQueue_ |
| std::atomic_bool | playbackStarted_ |
| AudioFrameResizer | recordQueue_ |
| std::atomic_bool | recordStarted_ |
| jami::WebRTCAudioProcessor::WebRTCAudioProcessor | ( | AudioFormat | format, |
| unsigned | frameSize | ||
| ) |
Definition at line 33 of file webrtc.cpp.
References jami::emitSignal(), jami::AudioProcessor::format_, jami::AudioProcessor::frameDurationMs_, jami::AudioProcessor::frameSize_, JAMI_ERROR, JAMI_LOG, jami::AudioFormat::nb_channels, jami::AudioFormat::sample_rate, and jami::webrtcNoError.

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Set the status of automatic gain control.
Implements jami::AudioProcessor.
Definition at line 75 of file webrtc.cpp.
References JAMI_ERROR, JAMI_LOG, and jami::webrtcNoError.
Set the status of echo cancellation.
Implements jami::AudioProcessor.
Definition at line 90 of file webrtc.cpp.
References JAMI_ERROR, JAMI_LOG, and jami::webrtcNoError.
Set the status of noise suppression includes de-reverb, de-noise, high pass filter, etc.
Implements jami::AudioProcessor.
Definition at line 60 of file webrtc.cpp.
References JAMI_ERROR, JAMI_LOG, and jami::webrtcNoError.
Set the status of voice activity detection.
Implements jami::AudioProcessor.
Definition at line 107 of file webrtc.cpp.
References JAMI_ERROR, JAMI_LOG, and jami::webrtcNoError.
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overridevirtual |
Process and return a single AudioFrame.
Implements jami::AudioProcessor.
Definition at line 123 of file webrtc.cpp.
References jami::AudioFrameResizer::dequeue(), jami::emitSignal(), jami::AudioProcessor::format_, jami::AudioProcessor::getStabilizedVoiceActivity(), JAMI_ERR, jami::AudioFormat::nb_channels, jami::AudioProcessor::playbackQueue_, jami::AudioProcessor::recordQueue_, jami::AudioFormat::sample_rate, jami::AudioFrameResizer::samples(), jami::AudioProcessor::tidyQueues(), and jami::webrtcNoError.
