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webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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This is the complete list of members for webrtc::test::MockAudioProcessing, including all inherited members.
| AnalyzeReverseStream(const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout)=0 | webrtc::AudioProcessing | pure virtual |
| AnalyzeReverseStream(const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout)=0 | webrtc::AudioProcessing | pure virtual |
| ApplyConfig(const Config &config)=0 | webrtc::AudioProcessing | pure virtual |
| ApplyConfig(const Config &config)=0 | webrtc::AudioProcessing | pure virtual |
| ChannelLayout enum name | webrtc::AudioProcessing | |
| ChannelLayout enum name | webrtc::AudioProcessing | |
| Create() | webrtc::AudioProcessing | static |
| Create(const webrtc::Config &config) | webrtc::AudioProcessing | static |
| Create(const webrtc::Config &config, NonlinearBeamformer *beamformer) | webrtc::AudioProcessing | static |
| Create() | webrtc::AudioProcessing | static |
| Create(const webrtc::Config &config) | webrtc::AudioProcessing | static |
| Create(const webrtc::Config &config, NonlinearBeamformer *beamformer) | webrtc::AudioProcessing | static |
| delay_offset_ms() const =0 | webrtc::AudioProcessing | pure virtual |
| delay_offset_ms() const =0 | webrtc::AudioProcessing | pure virtual |
| echo_cancellation() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| echo_cancellation() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| echo_control_mobile() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| echo_control_mobile() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| Error enum name | webrtc::AudioProcessing | |
| Error enum name | webrtc::AudioProcessing | |
| gain_control() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| gain_control() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| GetStatistics() const | webrtc::AudioProcessing | virtual |
| GetStatistics() const | webrtc::AudioProcessing | virtual |
| high_pass_filter() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| high_pass_filter() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| Initialize()=0 | webrtc::AudioProcessing | pure virtual |
| Initialize(const ProcessingConfig &processing_config)=0 | webrtc::AudioProcessing | pure virtual |
| 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 | webrtc::AudioProcessing | pure virtual |
| Initialize()=0 | webrtc::AudioProcessing | pure virtual |
| Initialize(const ProcessingConfig &processing_config)=0 | webrtc::AudioProcessing | pure virtual |
| 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 | webrtc::AudioProcessing | pure virtual |
| kBadDataLengthError enum value | webrtc::AudioProcessing | |
| kBadNumberChannelsError enum value | webrtc::AudioProcessing | |
| kBadParameterError enum value | webrtc::AudioProcessing | |
| kBadSampleRateError enum value | webrtc::AudioProcessing | |
| kBadStreamParameterWarning enum value | webrtc::AudioProcessing | |
| kChunkSizeMs | webrtc::AudioProcessing | static |
| kCreationFailedError enum value | webrtc::AudioProcessing | |
| kFileError enum value | webrtc::AudioProcessing | |
| kMaxFilenameSize | webrtc::AudioProcessing | static |
| kMaxNativeSampleRateHz | webrtc::AudioProcessing | static |
| kMono enum value | webrtc::AudioProcessing | |
| kMonoAndKeyboard enum value | webrtc::AudioProcessing | |
| kNativeSampleRatesHz | webrtc::AudioProcessing | static |
| kNoError enum value | webrtc::AudioProcessing | |
| kNotEnabledError enum value | webrtc::AudioProcessing | |
| kNullPointerError enum value | webrtc::AudioProcessing | |
| kNumNativeSampleRates | webrtc::AudioProcessing | static |
| kSampleRate16kHz enum value | webrtc::AudioProcessing | |
| kSampleRate32kHz enum value | webrtc::AudioProcessing | |
| kSampleRate48kHz enum value | webrtc::AudioProcessing | |
| kSampleRate8kHz enum value | webrtc::AudioProcessing | |
| kStereo enum value | webrtc::AudioProcessing | |
| kStereoAndKeyboard enum value | webrtc::AudioProcessing | |
| kStreamParameterNotSetError enum value | webrtc::AudioProcessing | |
| kUnspecifiedError enum value | webrtc::AudioProcessing | |
| kUnsupportedComponentError enum value | webrtc::AudioProcessing | |
| kUnsupportedFunctionError enum value | webrtc::AudioProcessing | |
| level_estimator() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| level_estimator() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| MOCK_CONST_METHOD0(proc_sample_rate_hz, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(proc_split_sample_rate_hz, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(num_input_channels, size_t()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(num_proc_channels, size_t()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(num_output_channels, size_t()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(num_reverse_channels, size_t()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(stream_delay_ms, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(was_stream_delay_set, bool()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(delay_offset_ms, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(GetStatistics, AudioProcessingStatistics()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(proc_sample_rate_hz, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(proc_split_sample_rate_hz, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(num_input_channels, size_t()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(num_proc_channels, size_t()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(num_output_channels, size_t()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(num_reverse_channels, size_t()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(stream_delay_ms, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(was_stream_delay_set, bool()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(delay_offset_ms, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_CONST_METHOD0(GetStatistics, AudioProcessingStatistics()) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD0(Initialize, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD0(StopDebugRecording, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD0(UpdateHistogramsOnCallEnd, void()) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD0(Initialize, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD0(StopDebugRecording, int()) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD0(UpdateHistogramsOnCallEnd, void()) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(Initialize, int(const ProcessingConfig &processing_config)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(ApplyConfig, void(const Config &config)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(SetExtraOptions, void(const webrtc::Config &config)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(set_output_will_be_muted, void(bool muted)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(ProcessStream, int(AudioFrame *frame)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(ProcessReverseStream, int(AudioFrame *frame)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(set_stream_delay_ms, int(int delay)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(set_stream_key_pressed, void(bool key_pressed)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(set_delay_offset_ms, void(int offset)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(StartDebugRecording, int(FILE *handle)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(StartDebugRecordingForPlatformFile, int(rtc::PlatformFile handle)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(Initialize, int(const ProcessingConfig &processing_config)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(ApplyConfig, void(const Config &config)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(SetExtraOptions, void(const webrtc::Config &config)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(set_output_will_be_muted, void(bool muted)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(ProcessStream, int(AudioFrame *frame)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(ProcessReverseStream, int(AudioFrame *frame)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(set_stream_delay_ms, int(int delay)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(set_stream_key_pressed, void(bool key_pressed)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(set_delay_offset_ms, void(int offset)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(StartDebugRecording, int(FILE *handle)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD1(StartDebugRecordingForPlatformFile, int(rtc::PlatformFile handle)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD2(StartDebugRecording, int(const char filename[kMaxFilenameSize], int64_t max_log_size_bytes)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD2(StartDebugRecording, int(FILE *handle, int64_t max_log_size_bytes)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD2(StartDebugRecording, int(const char filename[kMaxFilenameSize], int64_t max_log_size_bytes)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD2(StartDebugRecording, int(FILE *handle, int64_t max_log_size_bytes)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD4(ProcessStream, int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD4(AnalyzeReverseStream, int(const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD4(ProcessReverseStream, int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD4(ProcessStream, int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD4(AnalyzeReverseStream, int(const float *const *data, size_t samples_per_channel, int sample_rate_hz, ChannelLayout layout)) | webrtc::test::MockAudioProcessing | |
| MOCK_METHOD4(ProcessReverseStream, int(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)) | 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)) | 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)) | 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)) | 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)) | webrtc::test::MockAudioProcessing | |
| MockAudioProcessing() | webrtc::test::MockAudioProcessing | inline |
| MockAudioProcessing() | webrtc::test::MockAudioProcessing | inline |
| NativeRate enum name | webrtc::AudioProcessing | |
| NativeRate enum name | webrtc::AudioProcessing | |
| noise_suppression() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| noise_suppression() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| num_input_channels() const =0 | webrtc::AudioProcessing | pure virtual |
| num_input_channels() const =0 | webrtc::AudioProcessing | pure virtual |
| num_output_channels() const =0 | webrtc::AudioProcessing | pure virtual |
| num_output_channels() const =0 | webrtc::AudioProcessing | pure virtual |
| num_proc_channels() const =0 | webrtc::AudioProcessing | pure virtual |
| num_proc_channels() const =0 | webrtc::AudioProcessing | pure virtual |
| num_reverse_channels() const =0 | webrtc::AudioProcessing | pure virtual |
| num_reverse_channels() const =0 | webrtc::AudioProcessing | pure virtual |
| proc_sample_rate_hz() const =0 | webrtc::AudioProcessing | pure virtual |
| proc_sample_rate_hz() const =0 | webrtc::AudioProcessing | pure virtual |
| proc_split_sample_rate_hz() const =0 | webrtc::AudioProcessing | pure virtual |
| proc_split_sample_rate_hz() const =0 | webrtc::AudioProcessing | pure virtual |
| ProcessReverseStream(AudioFrame *frame)=0 | webrtc::AudioProcessing | pure virtual |
| ProcessReverseStream(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0 | webrtc::AudioProcessing | pure virtual |
| ProcessReverseStream(AudioFrame *frame)=0 | webrtc::AudioProcessing | pure virtual |
| ProcessReverseStream(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0 | webrtc::AudioProcessing | pure virtual |
| ProcessStream(AudioFrame *frame)=0 | webrtc::AudioProcessing | pure virtual |
| 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 | webrtc::AudioProcessing | pure virtual |
| ProcessStream(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0 | webrtc::AudioProcessing | pure virtual |
| ProcessStream(AudioFrame *frame)=0 | webrtc::AudioProcessing | pure virtual |
| 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 | webrtc::AudioProcessing | pure virtual |
| ProcessStream(const float *const *src, const StreamConfig &input_config, const StreamConfig &output_config, float *const *dest)=0 | webrtc::AudioProcessing | pure virtual |
| set_delay_offset_ms(int offset)=0 | webrtc::AudioProcessing | pure virtual |
| set_delay_offset_ms(int offset)=0 | webrtc::AudioProcessing | pure virtual |
| set_output_will_be_muted(bool muted)=0 | webrtc::AudioProcessing | pure virtual |
| set_output_will_be_muted(bool muted)=0 | webrtc::AudioProcessing | pure virtual |
| set_stream_delay_ms(int delay)=0 | webrtc::AudioProcessing | pure virtual |
| set_stream_delay_ms(int delay)=0 | webrtc::AudioProcessing | pure virtual |
| set_stream_key_pressed(bool key_pressed)=0 | webrtc::AudioProcessing | pure virtual |
| set_stream_key_pressed(bool key_pressed)=0 | webrtc::AudioProcessing | pure virtual |
| SetExtraOptions(const webrtc::Config &config)=0 | webrtc::AudioProcessing | pure virtual |
| SetExtraOptions(const webrtc::Config &config)=0 | webrtc::AudioProcessing | pure virtual |
| StartDebugRecording(const char filename[kMaxFilenameSize], int64_t max_log_size_bytes)=0 | webrtc::AudioProcessing | pure virtual |
| StartDebugRecording(FILE *handle, int64_t max_log_size_bytes)=0 | webrtc::AudioProcessing | pure virtual |
| StartDebugRecording(FILE *handle)=0 | webrtc::AudioProcessing | pure virtual |
| StartDebugRecording(const char filename[kMaxFilenameSize], int64_t max_log_size_bytes)=0 | webrtc::AudioProcessing | pure virtual |
| StartDebugRecording(FILE *handle, int64_t max_log_size_bytes)=0 | webrtc::AudioProcessing | pure virtual |
| StartDebugRecording(FILE *handle)=0 | webrtc::AudioProcessing | pure virtual |
| StartDebugRecordingForPlatformFile(rtc::PlatformFile handle)=0 | webrtc::AudioProcessing | pure virtual |
| StartDebugRecordingForPlatformFile(rtc::PlatformFile handle)=0 | webrtc::AudioProcessing | pure virtual |
| StopDebugRecording()=0 | webrtc::AudioProcessing | pure virtual |
| StopDebugRecording()=0 | webrtc::AudioProcessing | pure virtual |
| stream_delay_ms() const =0 | webrtc::AudioProcessing | pure virtual |
| stream_delay_ms() const =0 | webrtc::AudioProcessing | pure virtual |
| UpdateHistogramsOnCallEnd()=0 | webrtc::AudioProcessing | pure virtual |
| UpdateHistogramsOnCallEnd()=0 | webrtc::AudioProcessing | pure virtual |
| voice_detection() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| voice_detection() const | webrtc::test::MockAudioProcessing | inlinevirtual |
| was_stream_delay_set() const =0 | webrtc::AudioProcessing | pure virtual |
| was_stream_delay_set() const =0 | webrtc::AudioProcessing | pure virtual |
| ~AudioProcessing() | webrtc::AudioProcessing | inlinevirtual |
| ~AudioProcessing() | webrtc::AudioProcessing | inlinevirtual |
| ~MockAudioProcessing() | webrtc::test::MockAudioProcessing | inlinevirtual |
| ~MockAudioProcessing() | webrtc::test::MockAudioProcessing | inlinevirtual |
1.8.13