webrtc::test::MockAudioProcessing Class Reference

#include <mock_audio_processing.h>

Inheritance diagram for webrtc::test::MockAudioProcessing:

Public Member Functions
	MockAudioProcessing ()
virtual	~MockAudioProcessing ()
	MOCK_METHOD0 (Initialize, int())
	MOCK_METHOD6 (Initialize, int(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))
	MOCK_METHOD1 (Initialize, int(const ProcessingConfig &processing_config))
	MOCK_METHOD1 (ApplyConfig, void(const Config &config))
	MOCK_METHOD1 (SetExtraOptions, void(const webrtc::Config &config))
	MOCK_CONST_METHOD0 (proc_sample_rate_hz, int())
	MOCK_CONST_METHOD0 (proc_split_sample_rate_hz, int())
	MOCK_CONST_METHOD0 (num_input_channels, size_t())
	MOCK_CONST_METHOD0 (num_proc_channels, size_t())
	MOCK_CONST_METHOD0 (num_output_channels, size_t())
	MOCK_CONST_METHOD0 (num_reverse_channels, size_t())
	MOCK_METHOD1 (set_output_will_be_muted, void(bool muted))
	MOCK_METHOD1 (ProcessStream, int(AudioFrame *frame))
	MOCK_METHOD7 (ProcessStream, int(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))
	MOCK_METHOD4 (ProcessStream, int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest))
	MOCK_METHOD1 (ProcessReverseStream, int(AudioFrame *frame))
	MOCK_METHOD4 (AnalyzeReverseStream, int(const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout))
	MOCK_METHOD4 (ProcessReverseStream, int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest))
	MOCK_METHOD1 (set_stream_delay_ms, int(int delay))
	MOCK_CONST_METHOD0 (stream_delay_ms, int())
	MOCK_CONST_METHOD0 (was_stream_delay_set, bool())
	MOCK_METHOD1 (set_stream_key_pressed, void(bool key_pressed))
	MOCK_METHOD1 (set_delay_offset_ms, void(int offset))
	MOCK_CONST_METHOD0 (delay_offset_ms, int())
	MOCK_METHOD2 (StartDebugRecording, int(const char filename[kMaxFilenameSize], int64_t max_log_size_bytes))
	MOCK_METHOD2 (StartDebugRecording, int(FILE *handle, int64_t max_log_size_bytes))
	MOCK_METHOD1 (StartDebugRecording, int(FILE *handle))
	MOCK_METHOD1 (StartDebugRecordingForPlatformFile, int(rtc::PlatformFile handle))
	MOCK_METHOD0 (StopDebugRecording, int())
	MOCK_METHOD0 (UpdateHistogramsOnCallEnd, void())
	MOCK_CONST_METHOD0 (GetStatistics, AudioProcessingStatistics())
virtual MockEchoCancellation *	echo_cancellation () const
virtual MockEchoControlMobile *	echo_control_mobile () const
virtual MockGainControl *	gain_control () const
virtual MockHighPassFilter *	high_pass_filter () const
virtual MockLevelEstimator *	level_estimator () const
virtual MockNoiseSuppression *	noise_suppression () const
virtual MockVoiceDetection *	voice_detection () const
	MockAudioProcessing ()
virtual	~MockAudioProcessing ()
	MOCK_METHOD0 (Initialize, int())
	MOCK_METHOD6 (Initialize, int(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))
	MOCK_METHOD1 (Initialize, int(const ProcessingConfig &processing_config))
	MOCK_METHOD1 (ApplyConfig, void(const Config &config))
	MOCK_METHOD1 (SetExtraOptions, void(const webrtc::Config &config))
	MOCK_CONST_METHOD0 (proc_sample_rate_hz, int())
	MOCK_CONST_METHOD0 (proc_split_sample_rate_hz, int())
	MOCK_CONST_METHOD0 (num_input_channels, size_t())
	MOCK_CONST_METHOD0 (num_proc_channels, size_t())
	MOCK_CONST_METHOD0 (num_output_channels, size_t())
	MOCK_CONST_METHOD0 (num_reverse_channels, size_t())
	MOCK_METHOD1 (set_output_will_be_muted, void(bool muted))
	MOCK_METHOD1 (ProcessStream, int(AudioFrame *frame))
	MOCK_METHOD7 (ProcessStream, int(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))
	MOCK_METHOD4 (ProcessStream, int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest))
	MOCK_METHOD1 (ProcessReverseStream, int(AudioFrame *frame))
	MOCK_METHOD4 (AnalyzeReverseStream, int(const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout))
	MOCK_METHOD4 (ProcessReverseStream, int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest))
	MOCK_METHOD1 (set_stream_delay_ms, int(int delay))
	MOCK_CONST_METHOD0 (stream_delay_ms, int())
	MOCK_CONST_METHOD0 (was_stream_delay_set, bool())
	MOCK_METHOD1 (set_stream_key_pressed, void(bool key_pressed))
	MOCK_METHOD1 (set_delay_offset_ms, void(int offset))
	MOCK_CONST_METHOD0 (delay_offset_ms, int())
	MOCK_METHOD2 (StartDebugRecording, int(const char filename[kMaxFilenameSize], int64_t max_log_size_bytes))
	MOCK_METHOD2 (StartDebugRecording, int(FILE *handle, int64_t max_log_size_bytes))
	MOCK_METHOD1 (StartDebugRecording, int(FILE *handle))
	MOCK_METHOD1 (StartDebugRecordingForPlatformFile, int(rtc::PlatformFile handle))
	MOCK_METHOD0 (StopDebugRecording, int())
	MOCK_METHOD0 (UpdateHistogramsOnCallEnd, void())
	MOCK_CONST_METHOD0 (GetStatistics, AudioProcessingStatistics())
virtual MockEchoCancellation *	echo_cancellation () const
virtual MockEchoControlMobile *	echo_control_mobile () const
virtual MockGainControl *	gain_control () const
virtual MockHighPassFilter *	high_pass_filter () const
virtual MockLevelEstimator *	level_estimator () const
virtual MockNoiseSuppression *	noise_suppression () const
virtual MockVoiceDetection *	voice_detection () const
Public Member Functions inherited from webrtc::AudioProcessing
virtual	~AudioProcessing ()
virtual int	Initialize ()=0
virtual int	Initialize (const ProcessingConfig &processing_config)=0
virtual 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)=0
virtual void	ApplyConfig (const Config &config)=0
virtual void	SetExtraOptions (const webrtc::Config &config)=0
virtual int	proc_sample_rate_hz () const =0
virtual int	proc_split_sample_rate_hz () const =0
virtual size_t	num_input_channels () const =0
virtual size_t	num_proc_channels () const =0
virtual size_t	num_output_channels () const =0
virtual size_t	num_reverse_channels () const =0
virtual void	set_output_will_be_muted (bool muted)=0
virtual int	ProcessStream (AudioFrame *frame)=0
virtual 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)=0
virtual int	ProcessStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0
virtual int	ProcessReverseStream (AudioFrame *frame)=0
virtual int	AnalyzeReverseStream (const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout)=0
virtual int	ProcessReverseStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0
virtual int	set_stream_delay_ms (int delay)=0
virtual int	stream_delay_ms () const =0
virtual bool	was_stream_delay_set () const =0
virtual void	set_stream_key_pressed (bool key_pressed)=0
virtual void	set_delay_offset_ms (int offset)=0
virtual int	delay_offset_ms () const =0
virtual int	StartDebugRecording (const char filename[kMaxFilenameSize], int64_t max_log_size_bytes)=0
virtual int	StartDebugRecording (FILE *handle, int64_t max_log_size_bytes)=0
virtual int	StartDebugRecording (FILE *handle)=0
virtual int	StartDebugRecordingForPlatformFile (rtc::PlatformFile handle)=0
virtual int	StopDebugRecording ()=0
virtual void	UpdateHistogramsOnCallEnd ()=0
virtual AudioProcessingStatistics	GetStatistics () const
virtual	~AudioProcessing ()
virtual int	Initialize ()=0
virtual int	Initialize (const ProcessingConfig &processing_config)=0
virtual 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)=0
virtual void	ApplyConfig (const Config &config)=0
virtual void	SetExtraOptions (const webrtc::Config &config)=0
virtual int	proc_sample_rate_hz () const =0
virtual int	proc_split_sample_rate_hz () const =0
virtual size_t	num_input_channels () const =0
virtual size_t	num_proc_channels () const =0
virtual size_t	num_output_channels () const =0
virtual size_t	num_reverse_channels () const =0
virtual void	set_output_will_be_muted (bool muted)=0
virtual int	ProcessStream (AudioFrame *frame)=0
virtual 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)=0
virtual int	ProcessStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0
virtual int	ProcessReverseStream (AudioFrame *frame)=0
virtual int	AnalyzeReverseStream (const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout)=0
virtual int	ProcessReverseStream (const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0
virtual int	set_stream_delay_ms (int delay)=0
virtual int	stream_delay_ms () const =0
virtual bool	was_stream_delay_set () const =0
virtual void	set_stream_key_pressed (bool key_pressed)=0
virtual void	set_delay_offset_ms (int offset)=0
virtual int	delay_offset_ms () const =0
virtual int	StartDebugRecording (const char filename[kMaxFilenameSize], int64_t max_log_size_bytes)=0
virtual int	StartDebugRecording (FILE *handle, int64_t max_log_size_bytes)=0
virtual int	StartDebugRecording (FILE *handle)=0
virtual int	StartDebugRecordingForPlatformFile (rtc::PlatformFile handle)=0
virtual int	StopDebugRecording ()=0
virtual void	UpdateHistogramsOnCallEnd ()=0
virtual AudioProcessingStatistics	GetStatistics () const

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

MockAudioProcessing() [1/2]

webrtc::test::MockAudioProcessing::MockAudioProcessing ( )

inline

~MockAudioProcessing() [1/2]

virtual webrtc::test::MockAudioProcessing::~MockAudioProcessing ( )

inlinevirtual

MockAudioProcessing() [2/2]

webrtc::test::MockAudioProcessing::MockAudioProcessing ( )

inline

~MockAudioProcessing() [2/2]

virtual webrtc::test::MockAudioProcessing::~MockAudioProcessing ( )

inlinevirtual

Member Function Documentation

echo_cancellation() [1/2]

virtual MockEchoCancellation* webrtc::test::MockAudioProcessing::echo_cancellation ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

echo_cancellation() [2/2]

virtual MockEchoCancellation* webrtc::test::MockAudioProcessing::echo_cancellation ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

echo_control_mobile() [1/2]

virtual MockEchoControlMobile* webrtc::test::MockAudioProcessing::echo_control_mobile ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

echo_control_mobile() [2/2]

virtual MockEchoControlMobile* webrtc::test::MockAudioProcessing::echo_control_mobile ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

gain_control() [1/2]

virtual MockGainControl* webrtc::test::MockAudioProcessing::gain_control ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

gain_control() [2/2]

virtual MockGainControl* webrtc::test::MockAudioProcessing::gain_control ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

high_pass_filter() [1/2]

virtual MockHighPassFilter* webrtc::test::MockAudioProcessing::high_pass_filter ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

high_pass_filter() [2/2]

virtual MockHighPassFilter* webrtc::test::MockAudioProcessing::high_pass_filter ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

level_estimator() [1/2]

virtual MockLevelEstimator* webrtc::test::MockAudioProcessing::level_estimator ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

level_estimator() [2/2]

virtual MockLevelEstimator* webrtc::test::MockAudioProcessing::level_estimator ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

MOCK_CONST_METHOD0() [1/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	proc_sample_rate_hz	,
		int()
	)

MOCK_CONST_METHOD0() [2/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	proc_sample_rate_hz	,
		int()
	)

MOCK_CONST_METHOD0() [3/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	proc_split_sample_rate_hz	,
		int()
	)

MOCK_CONST_METHOD0() [4/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	proc_split_sample_rate_hz	,
		int()
	)

MOCK_CONST_METHOD0() [5/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	num_input_channels	,
		size_t()
	)

MOCK_CONST_METHOD0() [6/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	num_input_channels	,
		size_t()
	)

MOCK_CONST_METHOD0() [7/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	num_proc_channels	,
		size_t()
	)

MOCK_CONST_METHOD0() [8/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	num_proc_channels	,
		size_t()
	)

MOCK_CONST_METHOD0() [9/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	num_output_channels	,
		size_t()
	)

MOCK_CONST_METHOD0() [10/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	num_output_channels	,
		size_t()
	)

MOCK_CONST_METHOD0() [11/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	num_reverse_channels	,
		size_t()
	)

MOCK_CONST_METHOD0() [12/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	num_reverse_channels	,
		size_t()
	)

MOCK_CONST_METHOD0() [13/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	stream_delay_ms	,
		int()
	)

MOCK_CONST_METHOD0() [14/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	stream_delay_ms	,
		int()
	)

MOCK_CONST_METHOD0() [15/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	was_stream_delay_set	,
		bool()
	)

MOCK_CONST_METHOD0() [16/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	was_stream_delay_set	,
		bool()
	)

MOCK_CONST_METHOD0() [17/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	delay_offset_ms	,
		int()
	)

MOCK_CONST_METHOD0() [18/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	delay_offset_ms	,
		int()
	)

MOCK_CONST_METHOD0() [19/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	GetStatistics	,
		AudioProcessingStatistics()
	)

MOCK_CONST_METHOD0() [20/20]

webrtc::test::MockAudioProcessing::MOCK_CONST_METHOD0	(	GetStatistics	,
		AudioProcessingStatistics()
	)

MOCK_METHOD0() [1/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD0	(	Initialize	,
		int()
	)

MOCK_METHOD0() [2/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD0	(	Initialize	,
		int()
	)

MOCK_METHOD0() [3/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD0	(	StopDebugRecording	,
		int()
	)

MOCK_METHOD0() [4/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD0	(	StopDebugRecording	,
		int()
	)

MOCK_METHOD0() [5/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD0	(	UpdateHistogramsOnCallEnd	,
		void()
	)

MOCK_METHOD0() [6/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD0	(	UpdateHistogramsOnCallEnd	,
		void()
	)

MOCK_METHOD1() [1/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	Initialize	,
		int(const ProcessingConfig &processing_config)
	)

MOCK_METHOD1() [2/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	Initialize	,
		int(const ProcessingConfig &processing_config)
	)

MOCK_METHOD1() [3/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	ApplyConfig	,
		void(const Config &config)
	)

MOCK_METHOD1() [4/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	ApplyConfig	,
		void(const Config &config)
	)

MOCK_METHOD1() [5/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	SetExtraOptions	,
		void(const webrtc::Config &config)
	)

MOCK_METHOD1() [6/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	SetExtraOptions	,
		void(const webrtc::Config &config)
	)

MOCK_METHOD1() [7/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	set_output_will_be_muted	,
		void(bool muted)
	)

MOCK_METHOD1() [8/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	set_output_will_be_muted	,
		void(bool muted)
	)

MOCK_METHOD1() [9/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	ProcessStream	,
		int(AudioFrame *frame)
	)

MOCK_METHOD1() [10/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	ProcessStream	,
		int(AudioFrame *frame)
	)

MOCK_METHOD1() [11/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	ProcessReverseStream	,
		int(AudioFrame *frame)
	)

MOCK_METHOD1() [12/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	ProcessReverseStream	,
		int(AudioFrame *frame)
	)

MOCK_METHOD1() [13/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	set_stream_delay_ms	,
		int(int delay)
	)

MOCK_METHOD1() [14/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	set_stream_delay_ms	,
		int(int delay)
	)

MOCK_METHOD1() [15/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	set_stream_key_pressed	,
		void(bool key_pressed)
	)

MOCK_METHOD1() [16/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	set_stream_key_pressed	,
		void(bool key_pressed)
	)

MOCK_METHOD1() [17/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	set_delay_offset_ms	,
		void(int offset)
	)

MOCK_METHOD1() [18/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	set_delay_offset_ms	,
		void(int offset)
	)

MOCK_METHOD1() [19/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	StartDebugRecording	,
		int(FILE *handle)
	)

MOCK_METHOD1() [20/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	StartDebugRecording	,
		int(FILE *handle)
	)

MOCK_METHOD1() [21/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	StartDebugRecordingForPlatformFile	,
		int(rtc::PlatformFile handle)
	)

MOCK_METHOD1() [22/22]

webrtc::test::MockAudioProcessing::MOCK_METHOD1	(	StartDebugRecordingForPlatformFile	,
		int(rtc::PlatformFile handle)
	)

MOCK_METHOD2() [1/4]

webrtc::test::MockAudioProcessing::MOCK_METHOD2	(	StartDebugRecording	,
		int(const char filename[kMaxFilenameSize], int64_t max_log_size_bytes)
	)

MOCK_METHOD2() [2/4]

webrtc::test::MockAudioProcessing::MOCK_METHOD2	(	StartDebugRecording	,
		int(const char filename[kMaxFilenameSize], int64_t max_log_size_bytes)
	)

MOCK_METHOD2() [3/4]

webrtc::test::MockAudioProcessing::MOCK_METHOD2	(	StartDebugRecording	,
		int(FILE *handle, int64_t max_log_size_bytes)
	)

MOCK_METHOD2() [4/4]

webrtc::test::MockAudioProcessing::MOCK_METHOD2	(	StartDebugRecording	,
		int(FILE *handle, int64_t max_log_size_bytes)
	)

MOCK_METHOD4() [1/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD4	(	ProcessStream	,
		int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)
	)

MOCK_METHOD4() [2/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD4	(	ProcessStream	,
		int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)
	)

MOCK_METHOD4() [3/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD4	(	AnalyzeReverseStream	,
		int(const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout)
	)

MOCK_METHOD4() [4/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD4	(	AnalyzeReverseStream	,
		int(const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout)
	)

MOCK_METHOD4() [5/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD4	(	ProcessReverseStream	,
		int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)
	)

MOCK_METHOD4() [6/6]

webrtc::test::MockAudioProcessing::MOCK_METHOD4	(	ProcessReverseStream	,
		int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)
	)

MOCK_METHOD6() [1/2]

webrtc::test::MockAudioProcessing::MOCK_METHOD6	(	Initialize	,
		int(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)
	)

MOCK_METHOD6() [2/2]

webrtc::test::MockAudioProcessing::MOCK_METHOD6	(	Initialize	,
		int(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)
	)

MOCK_METHOD7() [1/2]

webrtc::test::MockAudioProcessing::MOCK_METHOD7	(	ProcessStream	,
		int(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)
	)

MOCK_METHOD7() [2/2]

webrtc::test::MockAudioProcessing::MOCK_METHOD7	(	ProcessStream	,
		int(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)
	)

noise_suppression() [1/2]

virtual MockNoiseSuppression* webrtc::test::MockAudioProcessing::noise_suppression ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

noise_suppression() [2/2]

virtual MockNoiseSuppression* webrtc::test::MockAudioProcessing::noise_suppression ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

voice_detection() [1/2]

virtual MockVoiceDetection* webrtc::test::MockAudioProcessing::voice_detection ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

voice_detection() [2/2]

virtual MockVoiceDetection* webrtc::test::MockAudioProcessing::voice_detection ( ) const

inlinevirtual

Implements webrtc::AudioProcessing.

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

• DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_processing/include/mock_audio_processing.h