opentera::AudioSource Class Reference

Represents an audio source that can be added to a WebRTC call. More...

#include <AudioSource.h>

Inheritance diagram for opentera::AudioSource:

Public Member Functions
	AudioSource (AudioSourceConfiguration configuration, int bitsPerSample, int sampleRate, size_t numberOfChannels)
	Creates an AudioSource. More...
	DECLARE_NOT_COPYABLE (AudioSource)
	DECLARE_NOT_MOVABLE (AudioSource)
void	AddSink (webrtc::AudioTrackSinkInterface *sink) override
void	RemoveSink (webrtc::AudioTrackSinkInterface *sink) override
bool	remote () const override
	Indicates if this source is remote. More...
SourceState	state () const override
	Indicates if this source is live. More...
const cricket::AudioOptions	options () const override
AudioSourceConfiguration	configuration () const
size_t	bytesPerSample () const
size_t	bytesPerFrame () const
void	setAudioDeviceModule (const rtc::scoped_refptr< OpenteraAudioDeviceModule > &audioDeviceModule)
void	sendFrame (const void *audioData, size_t numberOfFrames)
void	sendFrame (const void *audioData, size_t numberOfFrames, bool isTyping)
void	AddRef () const override
rtc::RefCountReleaseStatus	Release () const override

Detailed Description

Represents an audio source that can be added to a WebRTC call.

Pass a shared_ptr to an instance of this to the StreamClient and call sendFrame for each of your audio frame.

Constructor & Destructor Documentation

AudioSource()

AudioSource::AudioSource	(	AudioSourceConfiguration	configuration,
		int	bitsPerSample,
		int	sampleRate,
		size_t	numberOfChannels
	)

Creates an AudioSource.

Parameters

configuration	the configuration applied to the audio stream by the audio transport layer
bitsPerSample	The audio stream sample size (8, 16 or 32 bits)
sampleRate	The audio stream sample rate
numberOfChannels	The audio stream channel count

Member Function Documentation

AddSink()

void AudioSource::AddSink ( webrtc::AudioTrackSinkInterface *  sink )

override

Do nothing.

bytesPerFrame()

size_t AudioSource::bytesPerFrame

(

)

const

Returns

The frame size

bytesPerSample()

size_t AudioSource::bytesPerSample

(

)

const

Returns

The sample size

configuration()

AudioSourceConfiguration opentera::AudioSource::configuration ( ) const

inline

Returns

The audio source configuration

options()

const cricket::AudioOptions AudioSource::options ( ) const

override

Returns

The audio source options

remote()

bool AudioSource::remote ( ) const

override

Indicates if this source is remote.

Returns

Always false, the source is local

RemoveSink()

void AudioSource::RemoveSink ( webrtc::AudioTrackSinkInterface *  sink )

override

Do nothing.

sendFrame() [1/2]

void opentera::AudioSource::sendFrame ( const void *  audioData, size_t  numberOfFrames  )

inline

Send an audio frame

Parameters

audioData	The audio data
numberOfFrames	The number of frames

sendFrame() [2/2]

void AudioSource::sendFrame	(	const void *	audioData,
		size_t	numberOfFrames,
		bool	isTyping
	)

Send an audio frame

Parameters

audioData	The audio data
numberOfFrames	The number of frames
isTyping	Indicates if the frame contains typing sound. This is only useful with the typing detection option.

setAudioDeviceModule()

void AudioSource::setAudioDeviceModule

(

const rtc::scoped_refptr< OpenteraAudioDeviceModule > &

audioDeviceModule

)

Internal use only.

Parameters

audioDeviceModule

state()

webrtc::MediaSourceInterface::SourceState AudioSource::state ( ) const

override

Indicates if this source is live.

Returns

Always kLive, the source is live

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

• AudioSource.h
• AudioSource.cpp