webrtc::AudioDeviceBuffer Class Reference

#include <audio_device_buffer.h>

Inheritance diagram for webrtc::AudioDeviceBuffer:

Classes
struct	Stats

Public Types
enum	LogState {   LOG_START = 0, LOG_STOP, LOG_ACTIVE, LOG_START = 0,   LOG_STOP, LOG_ACTIVE }
enum	LogState {   LOG_START = 0, LOG_STOP, LOG_ACTIVE, LOG_START = 0,   LOG_STOP, LOG_ACTIVE }

Public Member Functions
	AudioDeviceBuffer ()
virtual	~AudioDeviceBuffer ()
void	SetId (uint32_t id)
int32_t	RegisterAudioCallback (AudioTransport *audio_callback)
void	StartPlayout ()
void	StartRecording ()
void	StopPlayout ()
void	StopRecording ()
int32_t	SetRecordingSampleRate (uint32_t fsHz)
int32_t	SetPlayoutSampleRate (uint32_t fsHz)
int32_t	RecordingSampleRate () const
int32_t	PlayoutSampleRate () const
int32_t	SetRecordingChannels (size_t channels)
int32_t	SetPlayoutChannels (size_t channels)
size_t	RecordingChannels () const
size_t	PlayoutChannels () const
int32_t	SetRecordingChannel (const AudioDeviceModule::ChannelType channel)
int32_t	RecordingChannel (AudioDeviceModule::ChannelType &channel) const
virtual int32_t	SetRecordedBuffer (const void *audio_buffer, size_t samples_per_channel)
int32_t	SetCurrentMicLevel (uint32_t level)
virtual void	SetVQEData (int play_delay_ms, int rec_delay_ms, int clock_drift)
virtual int32_t	DeliverRecordedData ()
uint32_t	NewMicLevel () const
virtual int32_t	RequestPlayoutData (size_t samples_per_channel)
virtual int32_t	GetPlayoutData (void *audio_buffer)
int32_t	StartInputFileRecording (const char fileName[kAdmMaxFileNameSize])
int32_t	StopInputFileRecording ()
int32_t	StartOutputFileRecording (const char fileName[kAdmMaxFileNameSize])
int32_t	StopOutputFileRecording ()
int32_t	SetTypingStatus (bool typing_status)
	AudioDeviceBuffer ()
virtual	~AudioDeviceBuffer ()
void	SetId (uint32_t id)
int32_t	RegisterAudioCallback (AudioTransport *audio_callback)
void	StartPlayout ()
void	StartRecording ()
void	StopPlayout ()
void	StopRecording ()
int32_t	SetRecordingSampleRate (uint32_t fsHz)
int32_t	SetPlayoutSampleRate (uint32_t fsHz)
int32_t	RecordingSampleRate () const
int32_t	PlayoutSampleRate () const
int32_t	SetRecordingChannels (size_t channels)
int32_t	SetPlayoutChannels (size_t channels)
size_t	RecordingChannels () const
size_t	PlayoutChannels () const
int32_t	SetRecordingChannel (const AudioDeviceModule::ChannelType channel)
int32_t	RecordingChannel (AudioDeviceModule::ChannelType &channel) const
virtual int32_t	SetRecordedBuffer (const void *audio_buffer, size_t samples_per_channel)
int32_t	SetCurrentMicLevel (uint32_t level)
virtual void	SetVQEData (int play_delay_ms, int rec_delay_ms, int clock_drift)
virtual int32_t	DeliverRecordedData ()
uint32_t	NewMicLevel () const
virtual int32_t	RequestPlayoutData (size_t samples_per_channel)
virtual int32_t	GetPlayoutData (void *audio_buffer)
int32_t	StartInputFileRecording (const char fileName[kAdmMaxFileNameSize])
int32_t	StopInputFileRecording ()
int32_t	StartOutputFileRecording (const char fileName[kAdmMaxFileNameSize])
int32_t	StopOutputFileRecording ()
int32_t	SetTypingStatus (bool typing_status)

Member Enumeration Documentation

LogState [1/2]

enum webrtc::AudioDeviceBuffer::LogState

Enumerator
LOG_START
LOG_STOP
LOG_ACTIVE
LOG_START
LOG_STOP
LOG_ACTIVE

LogState [2/2]

enum webrtc::AudioDeviceBuffer::LogState

Enumerator
LOG_START
LOG_STOP
LOG_ACTIVE
LOG_START
LOG_STOP
LOG_ACTIVE

Constructor & Destructor Documentation

AudioDeviceBuffer() [1/2]

webrtc::AudioDeviceBuffer::AudioDeviceBuffer

(

)

~AudioDeviceBuffer() [1/2]

webrtc::AudioDeviceBuffer::~AudioDeviceBuffer ( )

virtual

AudioDeviceBuffer() [2/2]

webrtc::AudioDeviceBuffer::AudioDeviceBuffer

(

)

~AudioDeviceBuffer() [2/2]

virtual webrtc::AudioDeviceBuffer::~AudioDeviceBuffer ( )

virtual

Member Function Documentation

DeliverRecordedData() [1/2]

int32_t webrtc::AudioDeviceBuffer::DeliverRecordedData ( )

virtual

DeliverRecordedData() [2/2]

virtual int32_t webrtc::AudioDeviceBuffer::DeliverRecordedData ( )

virtual

GetPlayoutData() [1/2]

int32_t webrtc::AudioDeviceBuffer::GetPlayoutData ( void *  audio_buffer )

virtual

GetPlayoutData() [2/2]

virtual int32_t webrtc::AudioDeviceBuffer::GetPlayoutData ( void *  audio_buffer )

virtual

NewMicLevel() [1/2]

uint32_t webrtc::AudioDeviceBuffer::NewMicLevel

(

)

const

NewMicLevel() [2/2]

uint32_t webrtc::AudioDeviceBuffer::NewMicLevel

(

)

const

PlayoutChannels() [1/2]

size_t webrtc::AudioDeviceBuffer::PlayoutChannels

(

)

const

PlayoutChannels() [2/2]

size_t webrtc::AudioDeviceBuffer::PlayoutChannels

(

)

const

PlayoutSampleRate() [1/2]

int32_t webrtc::AudioDeviceBuffer::PlayoutSampleRate

(

)

const

PlayoutSampleRate() [2/2]

int32_t webrtc::AudioDeviceBuffer::PlayoutSampleRate

(

)

const

RecordingChannel() [1/2]

int32_t webrtc::AudioDeviceBuffer::RecordingChannel

(

AudioDeviceModule::ChannelType &

channel

)

const

RecordingChannel() [2/2]

int32_t webrtc::AudioDeviceBuffer::RecordingChannel

(

AudioDeviceModule::ChannelType &

channel

)

const

RecordingChannels() [1/2]

size_t webrtc::AudioDeviceBuffer::RecordingChannels

(

)

const

RecordingChannels() [2/2]

size_t webrtc::AudioDeviceBuffer::RecordingChannels

(

)

const

RecordingSampleRate() [1/2]

int32_t webrtc::AudioDeviceBuffer::RecordingSampleRate

(

)

const

RecordingSampleRate() [2/2]

int32_t webrtc::AudioDeviceBuffer::RecordingSampleRate

(

)

const

RegisterAudioCallback() [1/2]

int32_t webrtc::AudioDeviceBuffer::RegisterAudioCallback

(

AudioTransport *

audio_callback

)

RegisterAudioCallback() [2/2]

int32_t webrtc::AudioDeviceBuffer::RegisterAudioCallback

(

AudioTransport *

audio_callback

)

RequestPlayoutData() [1/2]

int32_t webrtc::AudioDeviceBuffer::RequestPlayoutData ( size_t  samples_per_channel )

virtual

RequestPlayoutData() [2/2]

virtual int32_t webrtc::AudioDeviceBuffer::RequestPlayoutData ( size_t  samples_per_channel )

virtual

SetCurrentMicLevel() [1/2]

int32_t webrtc::AudioDeviceBuffer::SetCurrentMicLevel

(

uint32_t

level

)

SetCurrentMicLevel() [2/2]

int32_t webrtc::AudioDeviceBuffer::SetCurrentMicLevel

(

uint32_t

level

)

SetId() [1/2]

void webrtc::AudioDeviceBuffer::SetId ( uint32_t  id )

inline

SetId() [2/2]

void webrtc::AudioDeviceBuffer::SetId ( uint32_t  id )

inline

SetPlayoutChannels() [1/2]

int32_t webrtc::AudioDeviceBuffer::SetPlayoutChannels

(

size_t

channels

)

SetPlayoutChannels() [2/2]

int32_t webrtc::AudioDeviceBuffer::SetPlayoutChannels

(

size_t

channels

)

SetPlayoutSampleRate() [1/2]

int32_t webrtc::AudioDeviceBuffer::SetPlayoutSampleRate

(

uint32_t

fsHz

)

SetPlayoutSampleRate() [2/2]

int32_t webrtc::AudioDeviceBuffer::SetPlayoutSampleRate

(

uint32_t

fsHz

)

SetRecordedBuffer() [1/2]

int32_t webrtc::AudioDeviceBuffer::SetRecordedBuffer ( const void *  audio_buffer, size_t  samples_per_channel  )

virtual

SetRecordedBuffer() [2/2]

virtual int32_t webrtc::AudioDeviceBuffer::SetRecordedBuffer ( const void *  audio_buffer, size_t  samples_per_channel  )

virtual

SetRecordingChannel() [1/2]

int32_t webrtc::AudioDeviceBuffer::SetRecordingChannel

(

const AudioDeviceModule::ChannelType

channel

)

SetRecordingChannel() [2/2]

int32_t webrtc::AudioDeviceBuffer::SetRecordingChannel

(

const AudioDeviceModule::ChannelType

channel

)

SetRecordingChannels() [1/2]

int32_t webrtc::AudioDeviceBuffer::SetRecordingChannels

(

size_t

channels

)

SetRecordingChannels() [2/2]

int32_t webrtc::AudioDeviceBuffer::SetRecordingChannels

(

size_t

channels

)

SetRecordingSampleRate() [1/2]

int32_t webrtc::AudioDeviceBuffer::SetRecordingSampleRate

(

uint32_t

fsHz

)

SetRecordingSampleRate() [2/2]

int32_t webrtc::AudioDeviceBuffer::SetRecordingSampleRate

(

uint32_t

fsHz

)

SetTypingStatus() [1/2]

int32_t webrtc::AudioDeviceBuffer::SetTypingStatus

(

bool

typing_status

)

SetTypingStatus() [2/2]

int32_t webrtc::AudioDeviceBuffer::SetTypingStatus

(

bool

typing_status

)

SetVQEData() [1/2]

virtual void webrtc::AudioDeviceBuffer::SetVQEData ( int  play_delay_ms, int  rec_delay_ms, int  clock_drift  )

virtual

SetVQEData() [2/2]

void webrtc::AudioDeviceBuffer::SetVQEData ( int  play_delay_ms, int  rec_delay_ms, int  clock_drift  )

virtual

StartInputFileRecording() [1/2]

int32_t webrtc::AudioDeviceBuffer::StartInputFileRecording

(

const char

fileName[kAdmMaxFileNameSize]

)

StartInputFileRecording() [2/2]

int32_t webrtc::AudioDeviceBuffer::StartInputFileRecording

(

const char

fileName[kAdmMaxFileNameSize]

)

StartOutputFileRecording() [1/2]

int32_t webrtc::AudioDeviceBuffer::StartOutputFileRecording

(

const char

fileName[kAdmMaxFileNameSize]

)

StartOutputFileRecording() [2/2]

int32_t webrtc::AudioDeviceBuffer::StartOutputFileRecording

(

const char

fileName[kAdmMaxFileNameSize]

)

StartPlayout() [1/2]

void webrtc::AudioDeviceBuffer::StartPlayout

(

)

StartPlayout() [2/2]

void webrtc::AudioDeviceBuffer::StartPlayout

(

)

StartRecording() [1/2]

void webrtc::AudioDeviceBuffer::StartRecording

(

)

StartRecording() [2/2]

void webrtc::AudioDeviceBuffer::StartRecording

(

)

StopInputFileRecording() [1/2]

int32_t webrtc::AudioDeviceBuffer::StopInputFileRecording

(

)

StopInputFileRecording() [2/2]

int32_t webrtc::AudioDeviceBuffer::StopInputFileRecording

(

)

StopOutputFileRecording() [1/2]

int32_t webrtc::AudioDeviceBuffer::StopOutputFileRecording

(

)

StopOutputFileRecording() [2/2]

int32_t webrtc::AudioDeviceBuffer::StopOutputFileRecording

(

)

StopPlayout() [1/2]

void webrtc::AudioDeviceBuffer::StopPlayout

(

)

StopPlayout() [2/2]

void webrtc::AudioDeviceBuffer::StopPlayout

(

)

StopRecording() [1/2]

void webrtc::AudioDeviceBuffer::StopRecording

(

)

StopRecording() [2/2]

void webrtc::AudioDeviceBuffer::StopRecording

(

)

---

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

• DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_device/audio_device_buffer.h
• Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.cc