webrtc::AudioProcessingImpl Class Reference

#include <audio_processing_impl.h>

Inheritance diagram for webrtc::AudioProcessingImpl:

Classes
struct	ApmPrivateSubmodules
struct	ApmPublicSubmodules

Public Member Functions
	AudioProcessingImpl (const webrtc::Config &config)
	AudioProcessingImpl (const webrtc::Config &config, NonlinearBeamformer *beamformer)
	~AudioProcessingImpl () override
int	Initialize () override
int	Initialize (int capture_input_sample_rate_hz, int capture_output_sample_rate_hz, int render_sample_rate_hz, ChannelLayout capture_input_layout, ChannelLayout capture_output_layout, ChannelLayout render_input_layout) override
int	Initialize (const ProcessingConfig &processing_config) override
void	ApplyConfig (const AudioProcessing::Config &config) override
void	SetExtraOptions (const webrtc::Config &config) override
void	UpdateHistogramsOnCallEnd () override
int	StartDebugRecording (const char filename[kMaxFilenameSize], int64_t max_log_size_bytes) override
int	StartDebugRecording (FILE *handle, int64_t max_log_size_bytes) override
int	StartDebugRecording (FILE *handle) override
int	StartDebugRecordingForPlatformFile (rtc::PlatformFile handle) override
int	StopDebugRecording () override
int	ProcessStream (AudioFrame *frame) override
int	ProcessStream (const float *const *src, size_t samples_per_channel, int input_sample_rate_hz, ChannelLayout input_layout, int output_sample_rate_hz, ChannelLayout output_layout, float *const *dest) override
int	ProcessStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest) override
void	set_output_will_be_muted (bool muted) override
int	set_stream_delay_ms (int delay) override
void	set_delay_offset_ms (int offset) override
int	delay_offset_ms () const override
void	set_stream_key_pressed (bool key_pressed) override
int	ProcessReverseStream (AudioFrame *frame) override
int	AnalyzeReverseStream (const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout) override
int	ProcessReverseStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest) override
int	proc_sample_rate_hz () const override
int	proc_split_sample_rate_hz () const override
size_t	num_input_channels () const override
size_t	num_proc_channels () const override
size_t	num_output_channels () const override
size_t	num_reverse_channels () const override
int	stream_delay_ms () const override
bool	was_stream_delay_set () const override EXCLUSIVE_LOCKS_REQUIRED(crit_capture_)
AudioProcessingStatistics	GetStatistics () const override
EchoCancellation *	echo_cancellation () const override
EchoControlMobile *	echo_control_mobile () const override
GainControl *	gain_control () const override
HighPassFilter *	high_pass_filter () const override
LevelEstimator *	level_estimator () const override
NoiseSuppression *	noise_suppression () const override
VoiceDetection *	voice_detection () const override
void	MutateConfig (rtc::FunctionView< void(AudioProcessing::Config *)> mutator)
AudioProcessing::Config	GetConfig () const
	AudioProcessingImpl (const webrtc::Config &config)
	AudioProcessingImpl (const webrtc::Config &config, NonlinearBeamformer *beamformer)
	~AudioProcessingImpl () override
int	Initialize () override
int	Initialize (int capture_input_sample_rate_hz, int capture_output_sample_rate_hz, int render_sample_rate_hz, ChannelLayout capture_input_layout, ChannelLayout capture_output_layout, ChannelLayout render_input_layout) override
int	Initialize (const ProcessingConfig &processing_config) override
void	ApplyConfig (const AudioProcessing::Config &config) override
void	SetExtraOptions (const webrtc::Config &config) override
void	UpdateHistogramsOnCallEnd () override
int	StartDebugRecording (const char filename[kMaxFilenameSize], int64_t max_log_size_bytes) override
int	StartDebugRecording (FILE *handle, int64_t max_log_size_bytes) override
int	StartDebugRecording (FILE *handle) override
int	StartDebugRecordingForPlatformFile (rtc::PlatformFile handle) override
int	StopDebugRecording () override
int	ProcessStream (AudioFrame *frame) override
int	ProcessStream (const float *const *src, size_t samples_per_channel, int input_sample_rate_hz, ChannelLayout input_layout, int output_sample_rate_hz, ChannelLayout output_layout, float *const *dest) override
int	ProcessStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest) override
void	set_output_will_be_muted (bool muted) override
int	set_stream_delay_ms (int delay) override
void	set_delay_offset_ms (int offset) override
int	delay_offset_ms () const override
void	set_stream_key_pressed (bool key_pressed) override
int	ProcessReverseStream (AudioFrame *frame) override
int	AnalyzeReverseStream (const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout) override
int	ProcessReverseStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest) override
int	proc_sample_rate_hz () const override
int	proc_split_sample_rate_hz () const override
size_t	num_input_channels () const override
size_t	num_proc_channels () const override
size_t	num_output_channels () const override
size_t	num_reverse_channels () const override
int	stream_delay_ms () const override
bool	was_stream_delay_set () const override EXCLUSIVE_LOCKS_REQUIRED(crit_capture_)
AudioProcessingStatistics	GetStatistics () const override
EchoCancellation *	echo_cancellation () const override
EchoControlMobile *	echo_control_mobile () const override
GainControl *	gain_control () const override
HighPassFilter *	high_pass_filter () const override
LevelEstimator *	level_estimator () const override
NoiseSuppression *	noise_suppression () const override
VoiceDetection *	voice_detection () const override
void	MutateConfig (rtc::FunctionView< void(AudioProcessing::Config *)> mutator)
AudioProcessing::Config	GetConfig () const
Public Member Functions inherited from webrtc::AudioProcessing
virtual	~AudioProcessing ()
virtual	~AudioProcessing ()

Protected Member Functions
virtual int	InitializeLocked () EXCLUSIVE_LOCKS_REQUIRED(crit_render_
virtual int	InitializeLocked () EXCLUSIVE_LOCKS_REQUIRED(crit_render_

Protected Attributes
virtual int	crit_capture_

Additional Inherited Members
Public Types inherited from webrtc::AudioProcessing
enum	ChannelLayout {   kMono, kStereo, kMonoAndKeyboard, kStereoAndKeyboard,   kMono, kStereo, kMonoAndKeyboard, kStereoAndKeyboard }
enum	Error {   kNoError = 0, kUnspecifiedError = -1, kCreationFailedError = -2, kUnsupportedComponentError = -3,   kUnsupportedFunctionError = -4, kNullPointerError = -5, kBadParameterError = -6, kBadSampleRateError = -7,   kBadDataLengthError = -8, kBadNumberChannelsError = -9, kFileError = -10, kStreamParameterNotSetError = -11,   kNotEnabledError = -12, kBadStreamParameterWarning = -13, kNoError = 0, kUnspecifiedError = -1,   kCreationFailedError = -2, kUnsupportedComponentError = -3, kUnsupportedFunctionError = -4, kNullPointerError = -5,   kBadParameterError = -6, kBadSampleRateError = -7, kBadDataLengthError = -8, kBadNumberChannelsError = -9,   kFileError = -10, kStreamParameterNotSetError = -11, kNotEnabledError = -12, kBadStreamParameterWarning = -13 }
enum	NativeRate {   kSampleRate8kHz = 8000, kSampleRate16kHz = 16000, kSampleRate32kHz = 32000, kSampleRate48kHz = 48000,   kSampleRate8kHz = 8000, kSampleRate16kHz = 16000, kSampleRate32kHz = 32000, kSampleRate48kHz = 48000 }
enum	ChannelLayout {   kMono, kStereo, kMonoAndKeyboard, kStereoAndKeyboard,   kMono, kStereo, kMonoAndKeyboard, kStereoAndKeyboard }
enum	Error {   kNoError = 0, kUnspecifiedError = -1, kCreationFailedError = -2, kUnsupportedComponentError = -3,   kUnsupportedFunctionError = -4, kNullPointerError = -5, kBadParameterError = -6, kBadSampleRateError = -7,   kBadDataLengthError = -8, kBadNumberChannelsError = -9, kFileError = -10, kStreamParameterNotSetError = -11,   kNotEnabledError = -12, kBadStreamParameterWarning = -13, kNoError = 0, kUnspecifiedError = -1,   kCreationFailedError = -2, kUnsupportedComponentError = -3, kUnsupportedFunctionError = -4, kNullPointerError = -5,   kBadParameterError = -6, kBadSampleRateError = -7, kBadDataLengthError = -8, kBadNumberChannelsError = -9,   kFileError = -10, kStreamParameterNotSetError = -11, kNotEnabledError = -12, kBadStreamParameterWarning = -13 }
enum	NativeRate {   kSampleRate8kHz = 8000, kSampleRate16kHz = 16000, kSampleRate32kHz = 32000, kSampleRate48kHz = 48000,   kSampleRate8kHz = 8000, kSampleRate16kHz = 16000, kSampleRate32kHz = 32000, kSampleRate48kHz = 48000 }
Static Public Member Functions inherited from webrtc::AudioProcessing
static AudioProcessing *	Create ()
static AudioProcessing *	Create (const webrtc::Config &config)
static AudioProcessing *	Create (const webrtc::Config &config, NonlinearBeamformer *beamformer)
static AudioProcessing *	Create ()
static AudioProcessing *	Create (const webrtc::Config &config)
static AudioProcessing *	Create (const webrtc::Config &config, NonlinearBeamformer *beamformer)
Static Public Attributes inherited from webrtc::AudioProcessing
static const size_t	kMaxFilenameSize = 1024
static constexpr int	kNativeSampleRatesHz [4]
static constexpr size_t	kNumNativeSampleRates
static constexpr int	kMaxNativeSampleRateHz
static const int	kChunkSizeMs = 10

Constructor & Destructor Documentation

AudioProcessingImpl() [1/4]

webrtc::AudioProcessingImpl::AudioProcessingImpl ( const webrtc::Config &  config )

explicit

AudioProcessingImpl() [2/4]

webrtc::AudioProcessingImpl::AudioProcessingImpl	(	const webrtc::Config &	config,
		NonlinearBeamformer *	beamformer
	)

~AudioProcessingImpl() [1/2]

webrtc::AudioProcessingImpl::~AudioProcessingImpl ( )

override

AudioProcessingImpl() [3/4]

webrtc::AudioProcessingImpl::AudioProcessingImpl ( const webrtc::Config &  config )

explicit

AudioProcessingImpl() [4/4]

webrtc::AudioProcessingImpl::AudioProcessingImpl	(	const webrtc::Config &	config,
		NonlinearBeamformer *	beamformer
	)

~AudioProcessingImpl() [2/2]

webrtc::AudioProcessingImpl::~AudioProcessingImpl ( )

override

Member Function Documentation

AnalyzeReverseStream() [1/2]

int webrtc::AudioProcessingImpl::AnalyzeReverseStream ( const float *const *  data, size_t  samples_per_channel, int  sample_rate_hz, ChannelLayout  layout  )

overridevirtual

Implements webrtc::AudioProcessing.

AnalyzeReverseStream() [2/2]

int webrtc::AudioProcessingImpl::AnalyzeReverseStream ( const float *const *  data, size_t  samples_per_channel, int  sample_rate_hz, ChannelLayout  layout  )

overridevirtual

Implements webrtc::AudioProcessing.

ApplyConfig() [1/2]

void webrtc::AudioProcessingImpl::ApplyConfig ( const AudioProcessing::Config &  config )

overridevirtual

Implements webrtc::AudioProcessing.

ApplyConfig() [2/2]

void webrtc::AudioProcessingImpl::ApplyConfig ( const AudioProcessing::Config &  config )

overridevirtual

Implements webrtc::AudioProcessing.

delay_offset_ms() [1/2]

int webrtc::AudioProcessingImpl::delay_offset_ms ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

delay_offset_ms() [2/2]

int webrtc::AudioProcessingImpl::delay_offset_ms ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

echo_cancellation() [1/2]

EchoCancellation * webrtc::AudioProcessingImpl::echo_cancellation ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

echo_cancellation() [2/2]

EchoCancellation* webrtc::AudioProcessingImpl::echo_cancellation ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

echo_control_mobile() [1/2]

EchoControlMobile * webrtc::AudioProcessingImpl::echo_control_mobile ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

echo_control_mobile() [2/2]

EchoControlMobile* webrtc::AudioProcessingImpl::echo_control_mobile ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

gain_control() [1/2]

GainControl * webrtc::AudioProcessingImpl::gain_control ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

gain_control() [2/2]

GainControl* webrtc::AudioProcessingImpl::gain_control ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

GetConfig() [1/2]

AudioProcessing::Config webrtc::AudioProcessingImpl::GetConfig

(

)

const

GetConfig() [2/2]

AudioProcessing::Config webrtc::AudioProcessingImpl::GetConfig

(

)

const

GetStatistics() [1/2]

AudioProcessingStatistics webrtc::AudioProcessingImpl::GetStatistics ( ) const

overridevirtual

Reimplemented from webrtc::AudioProcessing.

GetStatistics() [2/2]

AudioProcessing::AudioProcessingStatistics webrtc::AudioProcessingImpl::GetStatistics ( ) const

overridevirtual

Reimplemented from webrtc::AudioProcessing.

high_pass_filter() [1/2]

HighPassFilter * webrtc::AudioProcessingImpl::high_pass_filter ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

high_pass_filter() [2/2]

HighPassFilter* webrtc::AudioProcessingImpl::high_pass_filter ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

Initialize() [1/6]

int webrtc::AudioProcessingImpl::Initialize ( )

overridevirtual

Implements webrtc::AudioProcessing.

Initialize() [2/6]

int webrtc::AudioProcessingImpl::Initialize ( )

overridevirtual

Implements webrtc::AudioProcessing.

Initialize() [3/6]

int webrtc::AudioProcessingImpl::Initialize ( int  capture_input_sample_rate_hz, int  capture_output_sample_rate_hz, int  render_sample_rate_hz, ChannelLayout  capture_input_layout, ChannelLayout  capture_output_layout, ChannelLayout  render_input_layout  )

overridevirtual

Implements webrtc::AudioProcessing.

Initialize() [4/6]

int webrtc::AudioProcessingImpl::Initialize ( int  capture_input_sample_rate_hz, int  capture_output_sample_rate_hz, int  render_sample_rate_hz, ChannelLayout  capture_input_layout, ChannelLayout  capture_output_layout, ChannelLayout  render_input_layout  )

overridevirtual

Implements webrtc::AudioProcessing.

Initialize() [5/6]

int webrtc::AudioProcessingImpl::Initialize ( const ProcessingConfig &  processing_config )

overridevirtual

Implements webrtc::AudioProcessing.

Initialize() [6/6]

int webrtc::AudioProcessingImpl::Initialize ( const ProcessingConfig &  processing_config )

overridevirtual

Implements webrtc::AudioProcessing.

InitializeLocked() [1/2]

int webrtc::AudioProcessingImpl::InitializeLocked ( )

protectedvirtual

InitializeLocked() [2/2]

virtual int webrtc::AudioProcessingImpl::InitializeLocked ( )

protectedvirtual

level_estimator() [1/2]

LevelEstimator * webrtc::AudioProcessingImpl::level_estimator ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

level_estimator() [2/2]

LevelEstimator* webrtc::AudioProcessingImpl::level_estimator ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

MutateConfig() [1/2]

void webrtc::AudioProcessingImpl::MutateConfig

(

rtc::FunctionView< void(AudioProcessing::Config *)>

mutator

)

MutateConfig() [2/2]

void webrtc::AudioProcessingImpl::MutateConfig

(

rtc::FunctionView< void(AudioProcessing::Config *)>

mutator

)

noise_suppression() [1/2]

NoiseSuppression * webrtc::AudioProcessingImpl::noise_suppression ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

noise_suppression() [2/2]

NoiseSuppression* webrtc::AudioProcessingImpl::noise_suppression ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

num_input_channels() [1/2]

size_t webrtc::AudioProcessingImpl::num_input_channels ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

num_input_channels() [2/2]

size_t webrtc::AudioProcessingImpl::num_input_channels ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

num_output_channels() [1/2]

size_t webrtc::AudioProcessingImpl::num_output_channels ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

num_output_channels() [2/2]

size_t webrtc::AudioProcessingImpl::num_output_channels ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

num_proc_channels() [1/2]

size_t webrtc::AudioProcessingImpl::num_proc_channels ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

num_proc_channels() [2/2]

size_t webrtc::AudioProcessingImpl::num_proc_channels ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

num_reverse_channels() [1/2]

size_t webrtc::AudioProcessingImpl::num_reverse_channels ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

num_reverse_channels() [2/2]

size_t webrtc::AudioProcessingImpl::num_reverse_channels ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

proc_sample_rate_hz() [1/2]

int webrtc::AudioProcessingImpl::proc_sample_rate_hz ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

proc_sample_rate_hz() [2/2]

int webrtc::AudioProcessingImpl::proc_sample_rate_hz ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

proc_split_sample_rate_hz() [1/2]

int webrtc::AudioProcessingImpl::proc_split_sample_rate_hz ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

proc_split_sample_rate_hz() [2/2]

int webrtc::AudioProcessingImpl::proc_split_sample_rate_hz ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

ProcessReverseStream() [1/4]

int webrtc::AudioProcessingImpl::ProcessReverseStream ( AudioFrame *  frame )

overridevirtual

Implements webrtc::AudioProcessing.

ProcessReverseStream() [2/4]

int webrtc::AudioProcessingImpl::ProcessReverseStream ( AudioFrame *  frame )

overridevirtual

Implements webrtc::AudioProcessing.

ProcessReverseStream() [3/4]

int webrtc::AudioProcessingImpl::ProcessReverseStream ( const float *const *  src, const StreamConfig &  input_config, const StreamConfig &  output_config, float *const *  dest  )

overridevirtual

Implements webrtc::AudioProcessing.

ProcessReverseStream() [4/4]

int webrtc::AudioProcessingImpl::ProcessReverseStream ( const float *const *  src, const StreamConfig &  input_config, const StreamConfig &  output_config, float *const *  dest  )

overridevirtual

Implements webrtc::AudioProcessing.

ProcessStream() [1/6]

int webrtc::AudioProcessingImpl::ProcessStream ( AudioFrame *  frame )

overridevirtual

Implements webrtc::AudioProcessing.

ProcessStream() [2/6]

int webrtc::AudioProcessingImpl::ProcessStream ( AudioFrame *  frame )

overridevirtual

Implements webrtc::AudioProcessing.

ProcessStream() [3/6]

int webrtc::AudioProcessingImpl::ProcessStream ( const float *const *  src, size_t  samples_per_channel, int  input_sample_rate_hz, ChannelLayout  input_layout, int  output_sample_rate_hz, ChannelLayout  output_layout, float *const *  dest  )

overridevirtual

Implements webrtc::AudioProcessing.

ProcessStream() [4/6]

int webrtc::AudioProcessingImpl::ProcessStream ( const float *const *  src, size_t  samples_per_channel, int  input_sample_rate_hz, ChannelLayout  input_layout, int  output_sample_rate_hz, ChannelLayout  output_layout, float *const *  dest  )

overridevirtual

Implements webrtc::AudioProcessing.

ProcessStream() [5/6]

int webrtc::AudioProcessingImpl::ProcessStream ( const float *const *  src, const StreamConfig &  input_config, const StreamConfig &  output_config, float *const *  dest  )

overridevirtual

Implements webrtc::AudioProcessing.

ProcessStream() [6/6]

int webrtc::AudioProcessingImpl::ProcessStream ( const float *const *  src, const StreamConfig &  input_config, const StreamConfig &  output_config, float *const *  dest  )

overridevirtual

Implements webrtc::AudioProcessing.

set_delay_offset_ms() [1/2]

void webrtc::AudioProcessingImpl::set_delay_offset_ms ( int  offset )

overridevirtual

Implements webrtc::AudioProcessing.

set_delay_offset_ms() [2/2]

void webrtc::AudioProcessingImpl::set_delay_offset_ms ( int  offset )

overridevirtual

Implements webrtc::AudioProcessing.

set_output_will_be_muted() [1/2]

void webrtc::AudioProcessingImpl::set_output_will_be_muted ( bool  muted )

overridevirtual

Implements webrtc::AudioProcessing.

set_output_will_be_muted() [2/2]

void webrtc::AudioProcessingImpl::set_output_will_be_muted ( bool  muted )

overridevirtual

Implements webrtc::AudioProcessing.

set_stream_delay_ms() [1/2]

int webrtc::AudioProcessingImpl::set_stream_delay_ms ( int  delay )

overridevirtual

Implements webrtc::AudioProcessing.

set_stream_delay_ms() [2/2]

int webrtc::AudioProcessingImpl::set_stream_delay_ms ( int  delay )

overridevirtual

Implements webrtc::AudioProcessing.

set_stream_key_pressed() [1/2]

void webrtc::AudioProcessingImpl::set_stream_key_pressed ( bool  key_pressed )

overridevirtual

Implements webrtc::AudioProcessing.

set_stream_key_pressed() [2/2]

void webrtc::AudioProcessingImpl::set_stream_key_pressed ( bool  key_pressed )

overridevirtual

Implements webrtc::AudioProcessing.

SetExtraOptions() [1/2]

void webrtc::AudioProcessingImpl::SetExtraOptions ( const webrtc::Config &  config )

overridevirtual

Implements webrtc::AudioProcessing.

SetExtraOptions() [2/2]

void webrtc::AudioProcessingImpl::SetExtraOptions ( const webrtc::Config &  config )

overridevirtual

Implements webrtc::AudioProcessing.

StartDebugRecording() [1/6]

int webrtc::AudioProcessingImpl::StartDebugRecording ( const char  filename[kMaxFilenameSize], int64_t  max_log_size_bytes  )

overridevirtual

Implements webrtc::AudioProcessing.

StartDebugRecording() [2/6]

int webrtc::AudioProcessingImpl::StartDebugRecording ( const char  filename[kMaxFilenameSize], int64_t  max_log_size_bytes  )

overridevirtual

Implements webrtc::AudioProcessing.

StartDebugRecording() [3/6]

int webrtc::AudioProcessingImpl::StartDebugRecording ( FILE *  handle, int64_t  max_log_size_bytes  )

overridevirtual

Implements webrtc::AudioProcessing.

StartDebugRecording() [4/6]

int webrtc::AudioProcessingImpl::StartDebugRecording ( FILE *  handle, int64_t  max_log_size_bytes  )

overridevirtual

Implements webrtc::AudioProcessing.

StartDebugRecording() [5/6]

int webrtc::AudioProcessingImpl::StartDebugRecording ( FILE *  handle )

overridevirtual

Implements webrtc::AudioProcessing.

StartDebugRecording() [6/6]

int webrtc::AudioProcessingImpl::StartDebugRecording ( FILE *  handle )

overridevirtual

Implements webrtc::AudioProcessing.

StartDebugRecordingForPlatformFile() [1/2]

int webrtc::AudioProcessingImpl::StartDebugRecordingForPlatformFile ( rtc::PlatformFile  handle )

overridevirtual

Implements webrtc::AudioProcessing.

StartDebugRecordingForPlatformFile() [2/2]

int webrtc::AudioProcessingImpl::StartDebugRecordingForPlatformFile ( rtc::PlatformFile  handle )

overridevirtual

Implements webrtc::AudioProcessing.

StopDebugRecording() [1/2]

int webrtc::AudioProcessingImpl::StopDebugRecording ( )

overridevirtual

Implements webrtc::AudioProcessing.

StopDebugRecording() [2/2]

int webrtc::AudioProcessingImpl::StopDebugRecording ( )

overridevirtual

Implements webrtc::AudioProcessing.

stream_delay_ms() [1/2]

int webrtc::AudioProcessingImpl::stream_delay_ms ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

stream_delay_ms() [2/2]

int webrtc::AudioProcessingImpl::stream_delay_ms ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

UpdateHistogramsOnCallEnd() [1/2]

void webrtc::AudioProcessingImpl::UpdateHistogramsOnCallEnd ( )

overridevirtual

Implements webrtc::AudioProcessing.

UpdateHistogramsOnCallEnd() [2/2]

void webrtc::AudioProcessingImpl::UpdateHistogramsOnCallEnd ( )

overridevirtual

Implements webrtc::AudioProcessing.

voice_detection() [1/2]

VoiceDetection* webrtc::AudioProcessingImpl::voice_detection ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

voice_detection() [2/2]

VoiceDetection * webrtc::AudioProcessingImpl::voice_detection ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

was_stream_delay_set() [1/2]

bool webrtc::AudioProcessingImpl::was_stream_delay_set ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

was_stream_delay_set() [2/2]

bool webrtc::AudioProcessingImpl::was_stream_delay_set ( ) const

overridevirtual

Implements webrtc::AudioProcessing.

Member Data Documentation

crit_capture_

rtc::CriticalSection webrtc::AudioProcessingImpl::crit_capture_

protected

---

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

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