webkit  2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
Public Member Functions | List of all members
webrtc::MockInitialize Class Reference
Inheritance diagram for webrtc::MockInitialize:
webrtc::AudioProcessingImpl webrtc::AudioProcessing webrtc::AudioProcessing

Public Member Functions

 MockInitialize (const webrtc::Config &config)
 
 MOCK_METHOD0 (InitializeLocked, int())
 
int RealInitializeLocked () NO_THREAD_SAFETY_ANALYSIS
 
- Public Member Functions inherited from webrtc::AudioProcessingImpl
 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
 
EchoCancellationecho_cancellation () const override
 
EchoControlMobileecho_control_mobile () const override
 
GainControlgain_control () const override
 
HighPassFilterhigh_pass_filter () const override
 
LevelEstimatorlevel_estimator () const override
 
NoiseSuppressionnoise_suppression () const override
 
VoiceDetectionvoice_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
 
EchoCancellationecho_cancellation () const override
 
EchoControlMobileecho_control_mobile () const override
 
GainControlgain_control () const override
 
HighPassFilterhigh_pass_filter () const override
 
LevelEstimatorlevel_estimator () const override
 
NoiseSuppressionnoise_suppression () const override
 
VoiceDetectionvoice_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 ()
 

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 AudioProcessingCreate ()
 
static AudioProcessingCreate (const webrtc::Config &config)
 
static AudioProcessingCreate (const webrtc::Config &config, NonlinearBeamformer *beamformer)
 
static AudioProcessingCreate ()
 
static AudioProcessingCreate (const webrtc::Config &config)
 
static AudioProcessingCreate (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
 
- Protected Member Functions inherited from webrtc::AudioProcessingImpl
virtual int InitializeLocked () EXCLUSIVE_LOCKS_REQUIRED(crit_render_
 
virtual int InitializeLocked () EXCLUSIVE_LOCKS_REQUIRED(crit_render_
 
- Protected Attributes inherited from webrtc::AudioProcessingImpl
virtual int crit_capture_
 

Constructor & Destructor Documentation

◆ MockInitialize()

webrtc::MockInitialize::MockInitialize ( const webrtc::Config config)
inlineexplicit

Member Function Documentation

◆ MOCK_METHOD0()

webrtc::MockInitialize::MOCK_METHOD0 ( InitializeLocked  ,
int()   
)

◆ RealInitializeLocked()

int webrtc::MockInitialize::RealInitializeLocked ( )
inline

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