jami::WebRTCAudioProcessor Class Reference

#include <webrtc.h>

Inheritance diagram for jami::WebRTCAudioProcessor:

Collaboration diagram for jami::WebRTCAudioProcessor:

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_
unsigned int	minimumConsequtiveDurationMs {200}
AudioFrameResizer	playbackQueue_
std::atomic_bool	playbackStarted_
AudioFrameResizer	recordQueue_
std::atomic_bool	recordStarted_
std::unique_ptr< Resampler >	resampler_

Detailed Description

Definition at line 27 of file webrtc.h.

Constructor & Destructor Documentation

WebRTCAudioProcessor()

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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~WebRTCAudioProcessor()

jami::WebRTCAudioProcessor::~WebRTCAudioProcessor ( )

default

Member Function Documentation

enableAutomaticGainControl()

void jami::WebRTCAudioProcessor::enableAutomaticGainControl ( bool  enabled )

overridevirtual

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.

enableEchoCancel()

void jami::WebRTCAudioProcessor::enableEchoCancel ( bool  enabled )

overridevirtual

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.

enableNoiseSuppression()

void jami::WebRTCAudioProcessor::enableNoiseSuppression ( bool  enabled )

overridevirtual

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.

enableVoiceActivityDetection()

void jami::WebRTCAudioProcessor::enableVoiceActivityDetection ( bool  enabled )

overridevirtual

Set the status of voice activity detection.

Implements jami::AudioProcessor.

Definition at line 108 of file webrtc.cpp.

References JAMI_ERROR, JAMI_LOG, and jami::webrtcNoError.

getProcessed()

std::shared_ptr< AudioFrame > jami::WebRTCAudioProcessor::getProcessed ( )

overridevirtual

Process and return a single AudioFrame.

Implements jami::AudioProcessor.

Definition at line 125 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.

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The documentation for this class was generated from the following files:

• webrtc.h
• webrtc.cpp