webrtc::test::AudioProcessingSimulator Class Reference

#include <audio_processing_simulator.h>

Inheritance diagram for webrtc::test::AudioProcessingSimulator:

Classes
class	ScopedTimer

Public Member Functions
	AudioProcessingSimulator (const SimulationSettings &settings)
virtual	~AudioProcessingSimulator ()
virtual void	Process ()=0
const TickIntervalStats &	proc_time () const
bool	OutputWasBitexact ()
size_t	get_num_process_stream_calls ()
size_t	get_num_reverse_process_stream_calls ()
	AudioProcessingSimulator (const SimulationSettings &settings)
virtual	~AudioProcessingSimulator ()
virtual void	Process ()=0
const TickIntervalStats &	proc_time () const
bool	OutputWasBitexact ()
size_t	get_num_process_stream_calls ()
size_t	get_num_reverse_process_stream_calls ()

Static Public Attributes
static const int	kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs

Protected Member Functions
TickIntervalStats *	mutable_proc_time ()
void	ProcessStream (bool fixed_interface)
void	ProcessReverseStream (bool fixed_interface)
void	CreateAudioProcessor ()
void	DestroyAudioProcessor ()
void	SetupBuffersConfigsOutputs (int input_sample_rate_hz, int output_sample_rate_hz, int reverse_input_sample_rate_hz, int reverse_output_sample_rate_hz, int input_num_channels, int output_num_channels, int reverse_input_num_channels, int reverse_output_num_channels)
TickIntervalStats *	mutable_proc_time ()
void	ProcessStream (bool fixed_interface)
void	ProcessReverseStream (bool fixed_interface)
void	CreateAudioProcessor ()
void	DestroyAudioProcessor ()
void	SetupBuffersConfigsOutputs (int input_sample_rate_hz, int output_sample_rate_hz, int reverse_input_sample_rate_hz, int reverse_output_sample_rate_hz, int input_num_channels, int output_num_channels, int reverse_input_num_channels, int reverse_output_num_channels)

Protected Attributes
const SimulationSettings	settings_
std::unique_ptr< AudioProcessing >	ap_
std::unique_ptr< ChannelBuffer< float > >	in_buf_
std::unique_ptr< ChannelBuffer< float > >	out_buf_
std::unique_ptr< ChannelBuffer< float > >	reverse_in_buf_
std::unique_ptr< ChannelBuffer< float > >	reverse_out_buf_
StreamConfig	in_config_
StreamConfig	out_config_
StreamConfig	reverse_in_config_
StreamConfig	reverse_out_config_
std::unique_ptr< ChannelBufferWavReader >	buffer_reader_
std::unique_ptr< ChannelBufferWavReader >	reverse_buffer_reader_
AudioFrame	rev_frame_
AudioFrame	fwd_frame_
bool	bitexact_output_ = true

Constructor & Destructor Documentation

AudioProcessingSimulator() [1/2]

webrtc::test::AudioProcessingSimulator::AudioProcessingSimulator ( const SimulationSettings &  settings )

explicit

~AudioProcessingSimulator() [1/2]

webrtc::test::AudioProcessingSimulator::~AudioProcessingSimulator ( )

virtual

AudioProcessingSimulator() [2/2]

webrtc::test::AudioProcessingSimulator::AudioProcessingSimulator ( const SimulationSettings &  settings )

explicit

~AudioProcessingSimulator() [2/2]

virtual webrtc::test::AudioProcessingSimulator::~AudioProcessingSimulator ( )

virtual

Member Function Documentation

CreateAudioProcessor() [1/2]

void webrtc::test::AudioProcessingSimulator::CreateAudioProcessor ( )

protected

CreateAudioProcessor() [2/2]

void webrtc::test::AudioProcessingSimulator::CreateAudioProcessor ( )

protected

DestroyAudioProcessor() [1/2]

void webrtc::test::AudioProcessingSimulator::DestroyAudioProcessor ( )

protected

DestroyAudioProcessor() [2/2]

void webrtc::test::AudioProcessingSimulator::DestroyAudioProcessor ( )

protected

get_num_process_stream_calls() [1/2]

size_t webrtc::test::AudioProcessingSimulator::get_num_process_stream_calls ( )

inline

get_num_process_stream_calls() [2/2]

size_t webrtc::test::AudioProcessingSimulator::get_num_process_stream_calls ( )

inline

get_num_reverse_process_stream_calls() [1/2]

size_t webrtc::test::AudioProcessingSimulator::get_num_reverse_process_stream_calls ( )

inline

get_num_reverse_process_stream_calls() [2/2]

size_t webrtc::test::AudioProcessingSimulator::get_num_reverse_process_stream_calls ( )

inline

mutable_proc_time() [1/2]

TickIntervalStats* webrtc::test::AudioProcessingSimulator::mutable_proc_time ( )

inlineprotected

mutable_proc_time() [2/2]

TickIntervalStats* webrtc::test::AudioProcessingSimulator::mutable_proc_time ( )

inlineprotected

OutputWasBitexact() [1/2]

bool webrtc::test::AudioProcessingSimulator::OutputWasBitexact ( )

inline

OutputWasBitexact() [2/2]

bool webrtc::test::AudioProcessingSimulator::OutputWasBitexact ( )

inline

proc_time() [1/2]

const TickIntervalStats& webrtc::test::AudioProcessingSimulator::proc_time ( ) const

inline

proc_time() [2/2]

const TickIntervalStats& webrtc::test::AudioProcessingSimulator::proc_time ( ) const

inline

Process() [1/2]

virtual void webrtc::test::AudioProcessingSimulator::Process ( )

pure virtual

Implemented in webrtc::test::AecDumpBasedSimulator, webrtc::test::AecDumpBasedSimulator, webrtc::test::WavBasedSimulator, and webrtc::test::WavBasedSimulator.

Process() [2/2]

virtual void webrtc::test::AudioProcessingSimulator::Process ( )

pure virtual

Implemented in webrtc::test::AecDumpBasedSimulator, webrtc::test::AecDumpBasedSimulator, webrtc::test::WavBasedSimulator, and webrtc::test::WavBasedSimulator.

ProcessReverseStream() [1/2]

void webrtc::test::AudioProcessingSimulator::ProcessReverseStream ( bool  fixed_interface )

protected

ProcessReverseStream() [2/2]

void webrtc::test::AudioProcessingSimulator::ProcessReverseStream ( bool  fixed_interface )

protected

ProcessStream() [1/2]

void webrtc::test::AudioProcessingSimulator::ProcessStream ( bool  fixed_interface )

protected

ProcessStream() [2/2]

void webrtc::test::AudioProcessingSimulator::ProcessStream ( bool  fixed_interface )

protected

SetupBuffersConfigsOutputs() [1/2]

void webrtc::test::AudioProcessingSimulator::SetupBuffersConfigsOutputs ( int  input_sample_rate_hz, int  output_sample_rate_hz, int  reverse_input_sample_rate_hz, int  reverse_output_sample_rate_hz, int  input_num_channels, int  output_num_channels, int  reverse_input_num_channels, int  reverse_output_num_channels  )

protected

SetupBuffersConfigsOutputs() [2/2]

void webrtc::test::AudioProcessingSimulator::SetupBuffersConfigsOutputs ( int  input_sample_rate_hz, int  output_sample_rate_hz, int  reverse_input_sample_rate_hz, int  reverse_output_sample_rate_hz, int  input_num_channels, int  output_num_channels, int  reverse_input_num_channels, int  reverse_output_num_channels  )

protected

Member Data Documentation

ap_

std::unique_ptr< AudioProcessing > webrtc::test::AudioProcessingSimulator::ap_

protected

bitexact_output_

bool webrtc::test::AudioProcessingSimulator::bitexact_output_ = true

protected

buffer_reader_

std::unique_ptr< ChannelBufferWavReader > webrtc::test::AudioProcessingSimulator::buffer_reader_

protected

fwd_frame_

AudioFrame webrtc::test::AudioProcessingSimulator::fwd_frame_

protected

in_buf_

std::unique_ptr< ChannelBuffer< float > > webrtc::test::AudioProcessingSimulator::in_buf_

protected

in_config_

StreamConfig webrtc::test::AudioProcessingSimulator::in_config_

protected

kChunksPerSecond

static const int webrtc::test::AudioProcessingSimulator::kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs

static

out_buf_

std::unique_ptr< ChannelBuffer< float > > webrtc::test::AudioProcessingSimulator::out_buf_

protected

out_config_

StreamConfig webrtc::test::AudioProcessingSimulator::out_config_

protected

rev_frame_

AudioFrame webrtc::test::AudioProcessingSimulator::rev_frame_

protected

reverse_buffer_reader_

std::unique_ptr< ChannelBufferWavReader > webrtc::test::AudioProcessingSimulator::reverse_buffer_reader_

protected

reverse_in_buf_

std::unique_ptr< ChannelBuffer< float > > webrtc::test::AudioProcessingSimulator::reverse_in_buf_

protected

reverse_in_config_

StreamConfig webrtc::test::AudioProcessingSimulator::reverse_in_config_

protected

reverse_out_buf_

std::unique_ptr< ChannelBuffer< float > > webrtc::test::AudioProcessingSimulator::reverse_out_buf_

protected

reverse_out_config_

StreamConfig webrtc::test::AudioProcessingSimulator::reverse_out_config_

protected

settings_

const SimulationSettings webrtc::test::AudioProcessingSimulator::settings_

protected

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

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