|
webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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This is the complete list of members for webrtc::test::MockVoiceEngine, including all inherited members.
| _apiCritPtr | webrtc::voe::SharedData | protected |
| _audioDeviceLayer | webrtc::voe::SharedData | protected |
| _audioDevicePtr | webrtc::voe::SharedData | protected |
| _channelManager | webrtc::voe::SharedData | protected |
| _engineStatistics | webrtc::voe::SharedData | protected |
| _instanceId | webrtc::voe::SharedData | protected |
| _moduleProcessThreadPtr | webrtc::voe::SharedData | protected |
| _outputMixerPtr | webrtc::voe::SharedData | protected |
| _ref_count | webrtc::VoiceEngineImpl | protected |
| _transmitMixerPtr | webrtc::voe::SharedData | protected |
| AddRef() | webrtc::VoiceEngineImpl | |
| AddRef() | webrtc::VoiceEngineImpl | |
| AssociateSendChannel(int channel, int accociate_send_channel) override | webrtc::VoEBaseImpl | virtual |
| AssociateSendChannel(int channel, int accociate_send_channel) override | webrtc::VoEBaseImpl | virtual |
| audio_device() | webrtc::voe::SharedData | inline |
| audio_device() | webrtc::voe::SharedData | inline |
| audio_device_layer() const | webrtc::voe::SharedData | inline |
| audio_device_layer() const | webrtc::voe::SharedData | inline |
| audio_device_module() override | webrtc::VoEBaseImpl | inlinevirtual |
| audio_device_module() override | webrtc::VoEBaseImpl | inlinevirtual |
| webrtc::audio_processing() | webrtc::voe::SharedData | inline |
| webrtc::audio_processing() | webrtc::voe::SharedData | inline |
| webrtc::VoEBaseImpl::audio_processing() override | webrtc::VoEBaseImpl | inlinevirtual |
| audio_transport() override | webrtc::VoEBaseImpl | inlinevirtual |
| audio_transport() override | webrtc::VoEBaseImpl | inlinevirtual |
| audioproc_ | webrtc::voe::SharedData | protected |
| BuiltInAECIsAvailable() const override | webrtc::VoEHardwareImpl | virtual |
| BuiltInAECIsAvailable() const override | webrtc::VoEHardwareImpl | virtual |
| BuiltInAGCIsAvailable() const override | webrtc::VoEHardwareImpl | virtual |
| BuiltInAGCIsAvailable() const override | webrtc::VoEHardwareImpl | virtual |
| BuiltInNSIsAvailable() const override | webrtc::VoEHardwareImpl | virtual |
| BuiltInNSIsAvailable() const override | webrtc::VoEHardwareImpl | virtual |
| channel_manager() | webrtc::voe::SharedData | inline |
| channel_manager() | webrtc::voe::SharedData | inline |
| Create() | webrtc::VoiceEngine | static |
| Create() | webrtc::VoiceEngine | static |
| CreateChannel() override | webrtc::VoEBaseImpl | virtual |
| CreateChannel(const ChannelConfig &config) override | webrtc::VoEBaseImpl | virtual |
| CreateChannel() override | webrtc::VoEBaseImpl | virtual |
| CreateChannel(const ChannelConfig &config) override | webrtc::VoEBaseImpl | virtual |
| crit_sec() | webrtc::voe::SharedData | inline |
| crit_sec() | webrtc::voe::SharedData | inline |
| DelayOffsetMs() override | webrtc::VoEAudioProcessingImpl | virtual |
| DelayOffsetMs() override | webrtc::VoEAudioProcessingImpl | virtual |
| Delete(VoiceEngine *&voiceEngine) | webrtc::VoiceEngine | static |
| Delete(VoiceEngine *&voiceEngine) | webrtc::VoiceEngine | static |
| DeleteChannel(int channel) override | webrtc::VoEBaseImpl | virtual |
| DeleteChannel(int channel) override | webrtc::VoEBaseImpl | virtual |
| DeRegisterExternalTransport(int channel) override | webrtc::VoENetworkImpl | virtual |
| DeRegisterExternalTransport(int channel) override | webrtc::VoENetworkImpl | virtual |
| DeRegisterVoiceEngineObserver() override | webrtc::VoEBaseImpl | virtual |
| DeRegisterVoiceEngineObserver() override | webrtc::VoEBaseImpl | virtual |
| DriftCompensationEnabled() override | webrtc::VoEAudioProcessingImpl | virtual |
| DriftCompensationEnabled() override | webrtc::VoEAudioProcessingImpl | virtual |
| DriftCompensationSupported() | webrtc::VoEAudioProcessing | static |
| DriftCompensationSupported() | webrtc::VoEAudioProcessing | static |
| EnableBuiltInAEC(bool enable) override | webrtc::VoEHardwareImpl | virtual |
| EnableBuiltInAEC(bool enable) override | webrtc::VoEHardwareImpl | virtual |
| EnableBuiltInAGC(bool enable) override | webrtc::VoEHardwareImpl | virtual |
| EnableBuiltInAGC(bool enable) override | webrtc::VoEHardwareImpl | virtual |
| EnableBuiltInNS(bool enable) override | webrtc::VoEHardwareImpl | virtual |
| EnableBuiltInNS(bool enable) override | webrtc::VoEHardwareImpl | virtual |
| EnableDriftCompensation(bool enable) override | webrtc::VoEAudioProcessingImpl | virtual |
| EnableDriftCompensation(bool enable) override | webrtc::VoEAudioProcessingImpl | virtual |
| EnableHighPassFilter(bool enable) override | webrtc::VoEAudioProcessingImpl | virtual |
| EnableHighPassFilter(bool enable) override | webrtc::VoEAudioProcessingImpl | virtual |
| EnableStereoChannelSwapping(bool enable) override | webrtc::VoEAudioProcessingImpl | virtual |
| EnableStereoChannelSwapping(bool enable) override | webrtc::VoEAudioProcessingImpl | virtual |
| ErrorCode enum name | webrtc::AudioDeviceObserver | |
| ErrorCode enum name | webrtc::AudioDeviceObserver | |
| GetAecmMode(AecmModes &mode, bool &enabledCNG) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetAecmMode(AecmModes &mode, bool &enabledCNG) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetAgcConfig(AgcConfig &config) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetAgcConfig(AgcConfig &config) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetAgcStatus(bool &enabled, AgcModes &mode) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetAgcStatus(bool &enabled, AgcModes &mode) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetAudioDeviceLayer(AudioLayers &audioLayer) override | webrtc::VoEHardwareImpl | virtual |
| GetAudioDeviceLayer(AudioLayers &audioLayer) override | webrtc::VoEHardwareImpl | virtual |
| GetChannelOutputVolumeScaling(int channel, float &scaling) override | webrtc::VoEVolumeControlImpl | virtual |
| GetChannelOutputVolumeScaling(int channel, float &scaling) override | webrtc::VoEVolumeControlImpl | virtual |
| GetChannelProxy(int channel_id) | webrtc::test::MockVoiceEngine | inlinevirtual |
| GetChannelProxy(int channel_id) | webrtc::test::MockVoiceEngine | inlinevirtual |
| GetCodec(int index, CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| GetCodec(int index, CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| GetDecodingCallStatistics(int channel, AudioDecodingCallStats *stats) const override | webrtc::VoENetEqStatsImpl | virtual |
| GetDecodingCallStatistics(int channel, AudioDecodingCallStats *stats) const override | webrtc::VoENetEqStatsImpl | virtual |
| GetEcDelayMetrics(int &delay_median, int &delay_std, float &fraction_poor_delays) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetEcDelayMetrics(int &delay_median, int &delay_std, float &fraction_poor_delays) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetEchoMetrics(int &ERL, int &ERLE, int &RERL, int &A_NLP) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetEchoMetrics(int &ERL, int &ERLE, int &RERL, int &A_NLP) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetEcMetricsStatus(bool &enabled) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetEcMetricsStatus(bool &enabled) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetEcStatus(bool &enabled, EcModes &mode) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetEcStatus(bool &enabled, EcModes &mode) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetFECStatus(int channel, bool &enabled) override | webrtc::VoECodecImpl | virtual |
| GetFECStatus(int channel, bool &enabled) override | webrtc::VoECodecImpl | virtual |
| GetInputMute(int channel, bool &enabled) override | webrtc::VoEVolumeControlImpl | virtual |
| GetInputMute(int channel, bool &enabled) override | webrtc::VoEVolumeControlImpl | virtual |
| webrtc::GetInterface(VoiceEngine *voiceEngine) | webrtc::VoEAudioProcessing | static |
| webrtc::GetInterface(VoiceEngine *voiceEngine) | webrtc::VoEAudioProcessing | static |
| webrtc::VoECodecImpl::GetInterface(VoiceEngine *voiceEngine) | webrtc::VoECodec | static |
| webrtc::VoEFileImpl::GetInterface(VoiceEngine *voiceEngine) | webrtc::VoEFile | static |
| webrtc::VoEHardwareImpl::GetInterface(VoiceEngine *voiceEngine) | webrtc::VoEHardware | static |
| webrtc::VoENetEqStatsImpl::GetInterface(VoiceEngine *voiceEngine) | webrtc::VoENetEqStats | static |
| webrtc::VoENetworkImpl::GetInterface(VoiceEngine *voiceEngine) | webrtc::VoENetwork | static |
| webrtc::VoERTP_RTCPImpl::GetInterface(VoiceEngine *voiceEngine) | webrtc::VoERTP_RTCP | static |
| webrtc::VoEVolumeControlImpl::GetInterface(VoiceEngine *voiceEngine) | webrtc::VoEVolumeControl | static |
| webrtc::VoEBaseImpl::GetInterface(VoiceEngine *voiceEngine) | webrtc::VoEBase | static |
| GetLocalSSRC(int channel, unsigned int &ssrc) override | webrtc::VoERTP_RTCPImpl | virtual |
| GetLocalSSRC(int channel, unsigned int &ssrc) override | webrtc::VoERTP_RTCPImpl | virtual |
| GetMicVolume(unsigned int &volume) override | webrtc::VoEVolumeControlImpl | virtual |
| GetMicVolume(unsigned int &volume) override | webrtc::VoEVolumeControlImpl | virtual |
| GetNetworkStatistics(int channel, NetworkStatistics &stats) override | webrtc::VoENetEqStatsImpl | virtual |
| GetNetworkStatistics(int channel, NetworkStatistics &stats) override | webrtc::VoENetEqStatsImpl | virtual |
| GetNsStatus(bool &enabled, NsModes &mode) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetNsStatus(bool &enabled, NsModes &mode) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetNumOfPlayoutDevices(int &devices) override | webrtc::VoEHardwareImpl | virtual |
| GetNumOfPlayoutDevices(int &devices) override | webrtc::VoEHardwareImpl | virtual |
| GetNumOfRecordingDevices(int &devices) override | webrtc::VoEHardwareImpl | virtual |
| GetNumOfRecordingDevices(int &devices) override | webrtc::VoEHardwareImpl | virtual |
| GetOpusDtxStatus(int channel, bool *enabled) override | webrtc::VoECodecImpl | virtual |
| GetOpusDtxStatus(int channel, bool *enabled) override | webrtc::VoECodecImpl | virtual |
| GetOutputVolumePan(int channel, float &left, float &right) override | webrtc::VoEVolumeControlImpl | virtual |
| GetOutputVolumePan(int channel, float &left, float &right) override | webrtc::VoEVolumeControlImpl | virtual |
| GetPlayoutDeviceName(int index, char strNameUTF8[128], char strGuidUTF8[128]) override | webrtc::VoEHardwareImpl | virtual |
| GetPlayoutDeviceName(int index, char strNameUTF8[128], char strGuidUTF8[128]) override | webrtc::VoEHardwareImpl | virtual |
| GetRecCodec(int channel, CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| GetRecCodec(int channel, CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| GetRecordingDeviceName(int index, char strNameUTF8[128], char strGuidUTF8[128]) override | webrtc::VoEHardwareImpl | virtual |
| GetRecordingDeviceName(int index, char strNameUTF8[128], char strGuidUTF8[128]) override | webrtc::VoEHardwareImpl | virtual |
| GetRecPayloadType(int channel, CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| GetRecPayloadType(int channel, CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| GetRemoteRTCP_CNAME(int channel, char cName[256]) override | webrtc::VoERTP_RTCPImpl | virtual |
| GetRemoteRTCP_CNAME(int channel, char cName[256]) override | webrtc::VoERTP_RTCPImpl | virtual |
| GetRemoteSSRC(int channel, unsigned int &ssrc) override | webrtc::VoERTP_RTCPImpl | virtual |
| GetRemoteSSRC(int channel, unsigned int &ssrc) override | webrtc::VoERTP_RTCPImpl | virtual |
| GetRTCPStatistics(int channel, CallStatistics &stats) override | webrtc::VoERTP_RTCPImpl | virtual |
| GetRTCPStatistics(int channel, CallStatistics &stats) override | webrtc::VoERTP_RTCPImpl | virtual |
| GetRTCPStatus(int channel, bool &enabled) override | webrtc::VoERTP_RTCPImpl | virtual |
| GetRTCPStatus(int channel, bool &enabled) override | webrtc::VoERTP_RTCPImpl | virtual |
| GetSendCodec(int channel, CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| GetSendCodec(int channel, CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| GetSpeakerVolume(unsigned int &volume) override | webrtc::VoEVolumeControlImpl | virtual |
| GetSpeakerVolume(unsigned int &volume) override | webrtc::VoEVolumeControlImpl | virtual |
| GetSpeechInputLevel(unsigned int &level) override | webrtc::VoEVolumeControlImpl | virtual |
| GetSpeechInputLevel(unsigned int &level) override | webrtc::VoEVolumeControlImpl | virtual |
| GetSpeechInputLevelFullRange(unsigned int &level) override | webrtc::VoEVolumeControlImpl | virtual |
| GetSpeechInputLevelFullRange(unsigned int &level) override | webrtc::VoEVolumeControlImpl | virtual |
| GetSpeechOutputLevel(int channel, unsigned int &level) override | webrtc::VoEVolumeControlImpl | virtual |
| GetSpeechOutputLevel(int channel, unsigned int &level) override | webrtc::VoEVolumeControlImpl | virtual |
| GetSpeechOutputLevelFullRange(int channel, unsigned int &level) override | webrtc::VoEVolumeControlImpl | virtual |
| GetSpeechOutputLevelFullRange(int channel, unsigned int &level) override | webrtc::VoEVolumeControlImpl | virtual |
| GetTypingDetectionStatus(bool &enabled) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetTypingDetectionStatus(bool &enabled) override | webrtc::VoEAudioProcessingImpl | virtual |
| GetVADStatus(int channel, bool &enabled, VadModes &mode, bool &disabledDTX) override | webrtc::VoECodecImpl | virtual |
| GetVADStatus(int channel, bool &enabled, VadModes &mode, bool &disabledDTX) override | webrtc::VoECodecImpl | virtual |
| GetVersion(char version[1024]) override | webrtc::VoEBaseImpl | virtual |
| GetVersion(char version[1024]) override | webrtc::VoEBaseImpl | virtual |
| GetVersionString() | webrtc::VoiceEngine | static |
| GetVersionString() | webrtc::VoiceEngine | static |
| Init(AudioDeviceModule *external_adm=nullptr, AudioProcessing *audioproc=nullptr, const rtc::scoped_refptr< AudioDecoderFactory > &decoder_factory=nullptr) override | webrtc::VoEBaseImpl | virtual |
| Init(AudioDeviceModule *external_adm=nullptr, AudioProcessing *audioproc=nullptr, const rtc::scoped_refptr< AudioDecoderFactory > &decoder_factory=nullptr) override | webrtc::VoEBaseImpl | virtual |
| instance_id() const | webrtc::voe::SharedData | inline |
| instance_id() const | webrtc::voe::SharedData | inline |
| IsHighPassFilterEnabled() override | webrtc::VoEAudioProcessingImpl | virtual |
| IsHighPassFilterEnabled() override | webrtc::VoEAudioProcessingImpl | virtual |
| IsPlayingFileAsMicrophone(int channel) override | webrtc::VoEFileImpl | virtual |
| IsPlayingFileAsMicrophone(int channel) override | webrtc::VoEFileImpl | virtual |
| IsPlayingFileLocally(int channel) override | webrtc::VoEFileImpl | virtual |
| IsPlayingFileLocally(int channel) override | webrtc::VoEFileImpl | virtual |
| IsStereoChannelSwappingEnabled() override | webrtc::VoEAudioProcessingImpl | virtual |
| IsStereoChannelSwappingEnabled() override | webrtc::VoEAudioProcessingImpl | virtual |
| kPlayoutError enum value | webrtc::AudioDeviceObserver | |
| kPlayoutWarning enum value | webrtc::AudioDeviceObserver | |
| kRecordingError enum value | webrtc::AudioDeviceObserver | |
| kRecordingWarning enum value | webrtc::AudioDeviceObserver | |
| LastError() override | webrtc::VoEBaseImpl | virtual |
| LastError() override | webrtc::VoEBaseImpl | virtual |
| MOCK_CONST_METHOD0(BuiltInAECIsAvailable, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_CONST_METHOD0(BuiltInAGCIsAvailable, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_CONST_METHOD0(BuiltInNSIsAvailable, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_CONST_METHOD0(BuiltInAECIsAvailable, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_CONST_METHOD0(BuiltInAGCIsAvailable, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_CONST_METHOD0(BuiltInNSIsAvailable, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_CONST_METHOD1(RecordingSampleRate, int(unsigned int *samples_per_sec)) | webrtc::test::MockVoiceEngine | |
| MOCK_CONST_METHOD1(PlayoutSampleRate, int(unsigned int *samples_per_sec)) | webrtc::test::MockVoiceEngine | |
| MOCK_CONST_METHOD1(RecordingSampleRate, int(unsigned int *samples_per_sec)) | webrtc::test::MockVoiceEngine | |
| MOCK_CONST_METHOD1(PlayoutSampleRate, int(unsigned int *samples_per_sec)) | webrtc::test::MockVoiceEngine | |
| MOCK_CONST_METHOD2(GetDecodingCallStatistics, int(int channel, AudioDecodingCallStats *stats)) | webrtc::test::MockVoiceEngine | |
| MOCK_CONST_METHOD2(GetDecodingCallStatistics, int(int channel, AudioDecodingCallStats *stats)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(DriftCompensationEnabled, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(DelayOffsetMs, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(IsHighPassFilterEnabled, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(StopDebugRecording, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(IsStereoChannelSwappingEnabled, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(DeRegisterVoiceEngineObserver, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(audio_processing, AudioProcessing *()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(audio_device_module, AudioDeviceModule *()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(transmit_mixer, voe::TransmitMixer *()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(Terminate, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(CreateChannel, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(LastError, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(audio_transport, AudioTransport *()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(NumOfCodecs, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(StopRecordingMicrophone, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(DriftCompensationEnabled, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(DelayOffsetMs, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(IsHighPassFilterEnabled, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(StopDebugRecording, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(IsStereoChannelSwappingEnabled, bool()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(DeRegisterVoiceEngineObserver, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(audio_processing, AudioProcessing *()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(audio_device_module, AudioDeviceModule *()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(transmit_mixer, voe::TransmitMixer *()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(Terminate, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(CreateChannel, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(LastError, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(audio_transport, AudioTransport *()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(NumOfCodecs, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD0(StopRecordingMicrophone, int()) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(ChannelProxyFactory, voe::ChannelProxy *(int channel_id)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetAgcConfig, int(AgcConfig config)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetAgcConfig, int(AgcConfig &config)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableDriftCompensation, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetDelayOffsetMs, void(int offset)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableHighPassFilter, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(VoiceActivityIndicator, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetEcMetricsStatus, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetEcMetricsStatus, int(bool &enabled)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StartDebugRecording, int(const char *fileNameUTF8)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StartDebugRecording, int(FILE *file_handle)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetTypingDetectionStatus, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetTypingDetectionStatus, int(bool &enabled)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(TimeSinceLastTyping, int(int &seconds)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableStereoChannelSwapping, void(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(RegisterVoiceEngineObserver, int(VoiceEngineObserver &observer)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(CreateChannel, int(const ChannelConfig &config)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(DeleteChannel, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StartReceive, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopReceive, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StartPlayout, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopPlayout, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StartSend, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopSend, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetVersion, int(char version[1024])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopPlayingFileLocally, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(IsPlayingFileLocally, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopPlayingFileAsMicrophone, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(IsPlayingFileAsMicrophone, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopRecordingPlayout, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetNumOfRecordingDevices, int(int &devices)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetNumOfPlayoutDevices, int(int &devices)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetPlayoutDevice, int(int index)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetAudioDeviceLayer, int(AudioLayers audioLayer)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetAudioDeviceLayer, int(AudioLayers &audioLayer)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetRecordingSampleRate, int(unsigned int samples_per_sec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetPlayoutSampleRate, int(unsigned int samples_per_sec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableBuiltInAEC, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableBuiltInAGC, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableBuiltInNS, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(DeRegisterExternalTransport, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(ChannelProxyFactory, voe::ChannelProxy *(int channel_id)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetAgcConfig, int(AgcConfig config)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetAgcConfig, int(AgcConfig &config)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableDriftCompensation, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetDelayOffsetMs, void(int offset)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableHighPassFilter, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(VoiceActivityIndicator, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetEcMetricsStatus, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetEcMetricsStatus, int(bool &enabled)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StartDebugRecording, int(const char *fileNameUTF8)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StartDebugRecording, int(FILE *file_handle)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetTypingDetectionStatus, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetTypingDetectionStatus, int(bool &enabled)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(TimeSinceLastTyping, int(int &seconds)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableStereoChannelSwapping, void(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(RegisterVoiceEngineObserver, int(VoiceEngineObserver &observer)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(CreateChannel, int(const ChannelConfig &config)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(DeleteChannel, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StartReceive, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopReceive, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StartPlayout, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopPlayout, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StartSend, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopSend, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetVersion, int(char version[1024])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopPlayingFileLocally, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(IsPlayingFileLocally, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopPlayingFileAsMicrophone, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(IsPlayingFileAsMicrophone, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(StopRecordingPlayout, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetNumOfRecordingDevices, int(int &devices)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetNumOfPlayoutDevices, int(int &devices)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetPlayoutDevice, int(int index)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetAudioDeviceLayer, int(AudioLayers audioLayer)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(GetAudioDeviceLayer, int(AudioLayers &audioLayer)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetRecordingSampleRate, int(unsigned int samples_per_sec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(SetPlayoutSampleRate, int(unsigned int samples_per_sec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableBuiltInAEC, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableBuiltInAGC, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(EnableBuiltInNS, int(bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD1(DeRegisterExternalTransport, int(int channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetNsStatus, int(bool enable, NsModes mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetNsStatus, int(bool &enabled, NsModes &mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetAgcStatus, int(bool enable, AgcModes mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetAgcStatus, int(bool &enabled, AgcModes &mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetEcStatus, int(bool enable, EcModes mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetEcStatus, int(bool &enabled, EcModes &mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetAecmMode, int(AecmModes mode, bool enableCNG)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetAecmMode, int(AecmModes &mode, bool &enabledCNG)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(AssociateSendChannel, int(int channel, int accociate_send_channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetCodec, int(int index, CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetSendCodec, int(int channel, const CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetSendCodec, int(int channel, CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetBitRate, int(int channel, int bitrate_bps)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRecCodec, int(int channel, CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetRecPayloadType, int(int channel, const CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRecPayloadType, int(int channel, CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetFECStatus, int(int channel, bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetFECStatus, int(int channel, bool &enabled)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetOpusMaxPlaybackRate, int(int channel, int frequency_hz)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetOpusDtx, int(int channel, bool enable_dtx)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(StartRecordingMicrophone, int(OutStream *stream, CodecInst *compression)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetRecordingDevice, int(int index, StereoChannel recordingChannel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetNetworkStatistics, int(int channel, NetworkStatistics &stats)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(RegisterExternalTransport, int(int channel, Transport &transport)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetLocalSSRC, int(int channel, unsigned int ssrc)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetLocalSSRC, int(int channel, unsigned int &ssrc)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRemoteSSRC, int(int channel, unsigned int &ssrc)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetRTCPStatus, int(int channel, bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRTCPStatus, int(int channel, bool &enabled)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetRTCP_CNAME, int(int channel, const char cName[256])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRTCP_CNAME, int(int channel, char cName[256])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRemoteRTCP_CNAME, int(int channel, char cName[256])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRTCPStatistics, int(int channel, CallStatistics &stats)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRemoteRTCPReportBlocks, int(int channel, std::vector< ReportBlock > *receive_blocks)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetNsStatus, int(bool enable, NsModes mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetNsStatus, int(bool &enabled, NsModes &mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetAgcStatus, int(bool enable, AgcModes mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetAgcStatus, int(bool &enabled, AgcModes &mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetEcStatus, int(bool enable, EcModes mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetEcStatus, int(bool &enabled, EcModes &mode)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetAecmMode, int(AecmModes mode, bool enableCNG)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetAecmMode, int(AecmModes &mode, bool &enabledCNG)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(AssociateSendChannel, int(int channel, int accociate_send_channel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetCodec, int(int index, CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetSendCodec, int(int channel, const CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetSendCodec, int(int channel, CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetBitRate, int(int channel, int bitrate_bps)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRecCodec, int(int channel, CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetRecPayloadType, int(int channel, const CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRecPayloadType, int(int channel, CodecInst &codec)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetFECStatus, int(int channel, bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetFECStatus, int(int channel, bool &enabled)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetOpusMaxPlaybackRate, int(int channel, int frequency_hz)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetOpusDtx, int(int channel, bool enable_dtx)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(StartRecordingMicrophone, int(OutStream *stream, CodecInst *compression)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetRecordingDevice, int(int index, StereoChannel recordingChannel)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetNetworkStatistics, int(int channel, NetworkStatistics &stats)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(RegisterExternalTransport, int(int channel, Transport &transport)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetLocalSSRC, int(int channel, unsigned int ssrc)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetLocalSSRC, int(int channel, unsigned int &ssrc)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRemoteSSRC, int(int channel, unsigned int &ssrc)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetRTCPStatus, int(int channel, bool enable)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRTCPStatus, int(int channel, bool &enabled)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(SetRTCP_CNAME, int(int channel, const char cName[256])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRTCP_CNAME, int(int channel, char cName[256])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRemoteRTCP_CNAME, int(int channel, char cName[256])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRTCPStatistics, int(int channel, CallStatistics &stats)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD2(GetRemoteRTCPReportBlocks, int(int channel, std::vector< ReportBlock > *receive_blocks)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(GetEcDelayMetrics, int(int &delay_median, int &delay_std, float &fraction_poor_delays)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(Init, int(AudioDeviceModule *external_adm, AudioProcessing *audioproc, const rtc::scoped_refptr< AudioDecoderFactory > &decoder_factory)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetSendCNPayloadType, int(int channel, int type, PayloadFrequencies frequency)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(StartRecordingPlayout, int(int channel, OutStream *stream, CodecInst *compression)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(StartRecordingMicrophone, int(const char *fileNameUTF8, CodecInst *compression, int maxSizeBytes)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(GetRecordingDeviceName, int(int index, char strNameUTF8[128], char strGuidUTF8[128])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(GetPlayoutDeviceName, int(int index, char strNameUTF8[128], char strGuidUTF8[128])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(ReceivedRTPPacket, int(int channel, const void *data, size_t length)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(ReceivedRTCPPacket, int(int channel, const void *data, size_t length)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetSendAudioLevelIndicationStatus, int(int channel, bool enable, unsigned char id)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetReceiveAudioLevelIndicationStatus, int(int channel, bool enable, unsigned char id)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetSendAbsoluteSenderTimeStatus, int(int channel, bool enable, unsigned char id)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetReceiveAbsoluteSenderTimeStatus, int(int channel, bool enable, unsigned char id)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetREDStatus, int(int channel, bool enable, int redPayloadtype)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(GetREDStatus, int(int channel, bool &enable, int &redPayloadtype)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetNACKStatus, int(int channel, bool enable, int maxNoPackets)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(GetEcDelayMetrics, int(int &delay_median, int &delay_std, float &fraction_poor_delays)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(Init, int(AudioDeviceModule *external_adm, AudioProcessing *audioproc, const rtc::scoped_refptr< AudioDecoderFactory > &decoder_factory)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetSendCNPayloadType, int(int channel, int type, PayloadFrequencies frequency)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(StartRecordingPlayout, int(int channel, OutStream *stream, CodecInst *compression)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(StartRecordingMicrophone, int(const char *fileNameUTF8, CodecInst *compression, int maxSizeBytes)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(GetRecordingDeviceName, int(int index, char strNameUTF8[128], char strGuidUTF8[128])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(GetPlayoutDeviceName, int(int index, char strNameUTF8[128], char strGuidUTF8[128])) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(ReceivedRTPPacket, int(int channel, const void *data, size_t length)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(ReceivedRTCPPacket, int(int channel, const void *data, size_t length)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetSendAudioLevelIndicationStatus, int(int channel, bool enable, unsigned char id)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetReceiveAudioLevelIndicationStatus, int(int channel, bool enable, unsigned char id)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetSendAbsoluteSenderTimeStatus, int(int channel, bool enable, unsigned char id)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetReceiveAbsoluteSenderTimeStatus, int(int channel, bool enable, unsigned char id)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetREDStatus, int(int channel, bool enable, int redPayloadtype)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(GetREDStatus, int(int channel, bool &enable, int &redPayloadtype)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD3(SetNACKStatus, int(int channel, bool enable, int maxNoPackets)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(GetEchoMetrics, int(int &ERL, int &ERLE, int &RERL, int &A_NLP)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(SetVADStatus, int(int channel, bool enable, VadModes mode, bool disableDTX)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(GetVADStatus, int(int channel, bool &enabled, VadModes &mode, bool &disabledDTX)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(StartRecordingPlayout, int(int channel, const char *fileNameUTF8, CodecInst *compression, int maxSizeBytes)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(ReceivedRTPPacket, int(int channel, const void *data, size_t length, const PacketTime &packet_time)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(GetRTPStatistics, int(int channel, unsigned int &averageJitterMs, unsigned int &maxJitterMs, unsigned int &discardedPackets)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(GetEchoMetrics, int(int &ERL, int &ERLE, int &RERL, int &A_NLP)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(SetVADStatus, int(int channel, bool enable, VadModes mode, bool disableDTX)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(GetVADStatus, int(int channel, bool &enabled, VadModes &mode, bool &disabledDTX)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(StartRecordingPlayout, int(int channel, const char *fileNameUTF8, CodecInst *compression, int maxSizeBytes)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(ReceivedRTPPacket, int(int channel, const void *data, size_t length, const PacketTime &packet_time)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD4(GetRTPStatistics, int(int channel, unsigned int &averageJitterMs, unsigned int &maxJitterMs, unsigned int &discardedPackets)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD5(SetTypingDetectionParameters, int(int timeWindow, int costPerTyping, int reportingThreshold, int penaltyDecay, int typeEventDelay)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD5(StartPlayingFileAsMicrophone, int(int channel, InStream *stream, bool mixWithMicrophone, FileFormats format, float volumeScaling)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD5(SetTypingDetectionParameters, int(int timeWindow, int costPerTyping, int reportingThreshold, int penaltyDecay, int typeEventDelay)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD5(StartPlayingFileAsMicrophone, int(int channel, InStream *stream, bool mixWithMicrophone, FileFormats format, float volumeScaling)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD6(StartPlayingFileLocally, int(int channel, InStream *stream, FileFormats format, float volumeScaling, int startPointMs, int stopPointMs)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD6(StartPlayingFileAsMicrophone, int(int channel, const char fileNameUTF8[1024], bool loop, bool mixWithMicrophone, FileFormats format, float volumeScaling)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD6(StartPlayingFileLocally, int(int channel, InStream *stream, FileFormats format, float volumeScaling, int startPointMs, int stopPointMs)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD6(StartPlayingFileAsMicrophone, int(int channel, const char fileNameUTF8[1024], bool loop, bool mixWithMicrophone, FileFormats format, float volumeScaling)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD7(StartPlayingFileLocally, int(int channel, const char fileNameUTF8[1024], bool loop, FileFormats format, float volumeScaling, int startPointMs, int stopPointMs)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD7(GetRemoteRTCPData, int(int channel, unsigned int &NTPHigh, unsigned int &NTPLow, unsigned int ×tamp, unsigned int &playoutTimestamp, unsigned int *jitter, unsigned short *fractionLost)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD7(StartPlayingFileLocally, int(int channel, const char fileNameUTF8[1024], bool loop, FileFormats format, float volumeScaling, int startPointMs, int stopPointMs)) | webrtc::test::MockVoiceEngine | |
| MOCK_METHOD7(GetRemoteRTCPData, int(int channel, unsigned int &NTPHigh, unsigned int &NTPLow, unsigned int ×tamp, unsigned int &playoutTimestamp, unsigned int *jitter, unsigned short *fractionLost)) | webrtc::test::MockVoiceEngine | |
| MockVoiceEngine(rtc::scoped_refptr< AudioDecoderFactory > decoder_factory=nullptr) | webrtc::test::MockVoiceEngine | inline |
| MockVoiceEngine(rtc::scoped_refptr< AudioDecoderFactory > decoder_factory=nullptr) | webrtc::test::MockVoiceEngine | inline |
| NeedMorePlayData(const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, void *audioSamples, size_t &nSamplesOut, int64_t *elapsed_time_ms, int64_t *ntp_time_ms) override | webrtc::VoEBaseImpl | virtual |
| NeedMorePlayData(const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, void *audioSamples, size_t &nSamplesOut, int64_t *elapsed_time_ms, int64_t *ntp_time_ms) override | webrtc::VoEBaseImpl | virtual |
| NumOfCodecs() override | webrtc::VoECodecImpl | virtual |
| NumOfCodecs() override | webrtc::VoECodecImpl | virtual |
| NumOfPlayingChannels() | webrtc::voe::SharedData | |
| NumOfPlayingChannels() | webrtc::voe::SharedData | |
| NumOfSendingChannels() | webrtc::voe::SharedData | |
| NumOfSendingChannels() | webrtc::voe::SharedData | |
| OnErrorIsReported(const ErrorCode error) override | webrtc::VoEBaseImpl | virtual |
| OnErrorIsReported(const ErrorCode error) override | webrtc::VoEBaseImpl | virtual |
| OnWarningIsReported(const WarningCode warning) override | webrtc::VoEBaseImpl | virtual |
| OnWarningIsReported(const WarningCode warning) override | webrtc::VoEBaseImpl | virtual |
| output_mixer() | webrtc::voe::SharedData | inline |
| output_mixer() | webrtc::voe::SharedData | inline |
| PlayoutSampleRate(unsigned int *samples_per_sec) const override | webrtc::VoEHardwareImpl | virtual |
| PlayoutSampleRate(unsigned int *samples_per_sec) const override | webrtc::VoEHardwareImpl | virtual |
| process_thread() | webrtc::voe::SharedData | inline |
| process_thread() | webrtc::voe::SharedData | inline |
| PullRenderData(int bits_per_sample, int sample_rate, size_t number_of_channels, size_t number_of_frames, void *audio_data, int64_t *elapsed_time_ms, int64_t *ntp_time_ms) override | webrtc::VoEBaseImpl | virtual |
| PullRenderData(int bits_per_sample, int sample_rate, size_t number_of_channels, size_t number_of_frames, void *audio_data, int64_t *elapsed_time_ms, int64_t *ntp_time_ms) override | webrtc::VoEBaseImpl | virtual |
| PushCaptureData(int voe_channel, const void *audio_data, int bits_per_sample, int sample_rate, size_t number_of_channels, size_t number_of_frames) override | webrtc::VoEBaseImpl | virtual |
| PushCaptureData(int voe_channel, const void *audio_data, int bits_per_sample, int sample_rate, size_t number_of_channels, size_t number_of_frames) override | webrtc::VoEBaseImpl | virtual |
| ReceivedRTCPPacket(int channel, const void *data, size_t length) override | webrtc::VoENetworkImpl | virtual |
| ReceivedRTCPPacket(int channel, const void *data, size_t length) override | webrtc::VoENetworkImpl | virtual |
| ReceivedRTPPacket(int channel, const void *data, size_t length) override | webrtc::VoENetworkImpl | virtual |
| ReceivedRTPPacket(int channel, const void *data, size_t length, const PacketTime &packet_time) override | webrtc::VoENetworkImpl | virtual |
| ReceivedRTPPacket(int channel, const void *data, size_t length) override | webrtc::VoENetworkImpl | virtual |
| ReceivedRTPPacket(int channel, const void *data, size_t length, const PacketTime &packet_time) override | webrtc::VoENetworkImpl | virtual |
| RecordedDataIsAvailable(const void *audioSamples, const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, const uint32_t totalDelayMS, const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, uint32_t &newMicLevel) override | webrtc::VoEBaseImpl | virtual |
| RecordedDataIsAvailable(const void *audioSamples, const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, const uint32_t totalDelayMS, const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, uint32_t &newMicLevel) override | webrtc::VoEBaseImpl | virtual |
| RecordingSampleRate(unsigned int *samples_per_sec) const override | webrtc::VoEHardwareImpl | virtual |
| RecordingSampleRate(unsigned int *samples_per_sec) const override | webrtc::VoEHardwareImpl | virtual |
| RegisterExternalTransport(int channel, Transport &transport) override | webrtc::VoENetworkImpl | virtual |
| RegisterExternalTransport(int channel, Transport &transport) override | webrtc::VoENetworkImpl | virtual |
| RegisterVoiceEngineObserver(VoiceEngineObserver &observer) override | webrtc::VoEBaseImpl | virtual |
| RegisterVoiceEngineObserver(VoiceEngineObserver &observer) override | webrtc::VoEBaseImpl | virtual |
| Release() override | webrtc::VoiceEngineImpl | virtual |
| Release() override | webrtc::VoiceEngineImpl | virtual |
| set_audio_device(const rtc::scoped_refptr< AudioDeviceModule > &audio_device) | webrtc::voe::SharedData | |
| set_audio_device(const rtc::scoped_refptr< AudioDeviceModule > &audio_device) | webrtc::voe::SharedData | |
| set_audio_device_layer(AudioDeviceModule::AudioLayer layer) | webrtc::voe::SharedData | inline |
| set_audio_device_layer(AudioDeviceModule::AudioLayer layer) | webrtc::voe::SharedData | inline |
| set_audio_processing(AudioProcessing *audio_processing) | webrtc::voe::SharedData | |
| set_audio_processing(AudioProcessing *audio_processing) | webrtc::voe::SharedData | |
| SetAecmMode(AecmModes mode=kAecmSpeakerphone, bool enableCNG=true) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetAecmMode(AecmModes mode=kAecmSpeakerphone, bool enableCNG=true) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetAgcConfig(AgcConfig config) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetAgcConfig(AgcConfig config) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetAgcStatus(bool enable, AgcModes mode=kAgcUnchanged) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetAgcStatus(bool enable, AgcModes mode=kAgcUnchanged) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetAndroidObjects(void *javaVM, void *context) | webrtc::VoiceEngine | static |
| SetAndroidObjects(void *javaVM, void *context) | webrtc::VoiceEngine | static |
| SetAudioDeviceLayer(AudioLayers audioLayer) override | webrtc::VoEHardwareImpl | virtual |
| SetAudioDeviceLayer(AudioLayers audioLayer) override | webrtc::VoEHardwareImpl | virtual |
| SetBitRate(int channel, int bitrate_bps) override | webrtc::VoECodecImpl | virtual |
| SetBitRate(int channel, int bitrate_bps) override | webrtc::VoECodecImpl | virtual |
| SetChannelOutputVolumeScaling(int channel, float scaling) override | webrtc::VoEVolumeControlImpl | virtual |
| SetChannelOutputVolumeScaling(int channel, float scaling) override | webrtc::VoEVolumeControlImpl | virtual |
| SetDelayOffsetMs(int offset) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetDelayOffsetMs(int offset) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetEcMetricsStatus(bool enable) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetEcMetricsStatus(bool enable) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetEcStatus(bool enable, EcModes mode=kEcUnchanged) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetEcStatus(bool enable, EcModes mode=kEcUnchanged) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetFECStatus(int channel, bool enable) override | webrtc::VoECodecImpl | virtual |
| SetFECStatus(int channel, bool enable) override | webrtc::VoECodecImpl | virtual |
| SetInputMute(int channel, bool enable) override | webrtc::VoEVolumeControlImpl | virtual |
| SetInputMute(int channel, bool enable) override | webrtc::VoEVolumeControlImpl | virtual |
| SetLastError(int32_t error) const | webrtc::voe::SharedData | |
| SetLastError(int32_t error, TraceLevel level) const | webrtc::voe::SharedData | |
| SetLastError(int32_t error, TraceLevel level, const char *msg) const | webrtc::voe::SharedData | |
| SetLastError(int32_t error) const | webrtc::voe::SharedData | |
| SetLastError(int32_t error, TraceLevel level) const | webrtc::voe::SharedData | |
| SetLastError(int32_t error, TraceLevel level, const char *msg) const | webrtc::voe::SharedData | |
| SetLocalSSRC(int channel, unsigned int ssrc) override | webrtc::VoERTP_RTCPImpl | virtual |
| SetLocalSSRC(int channel, unsigned int ssrc) override | webrtc::VoERTP_RTCPImpl | virtual |
| SetMicVolume(unsigned int volume) override | webrtc::VoEVolumeControlImpl | virtual |
| SetMicVolume(unsigned int volume) override | webrtc::VoEVolumeControlImpl | virtual |
| SetNsStatus(bool enable, NsModes mode=kNsUnchanged) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetNsStatus(bool enable, NsModes mode=kNsUnchanged) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetOpusDtx(int channel, bool enable_dtx) override | webrtc::VoECodecImpl | virtual |
| SetOpusDtx(int channel, bool enable_dtx) override | webrtc::VoECodecImpl | virtual |
| SetOpusMaxPlaybackRate(int channel, int frequency_hz) override | webrtc::VoECodecImpl | virtual |
| SetOpusMaxPlaybackRate(int channel, int frequency_hz) override | webrtc::VoECodecImpl | virtual |
| SetOutputVolumePan(int channel, float left, float right) override | webrtc::VoEVolumeControlImpl | virtual |
| SetOutputVolumePan(int channel, float left, float right) override | webrtc::VoEVolumeControlImpl | virtual |
| SetPlayoutDevice(int index) override | webrtc::VoEHardwareImpl | virtual |
| SetPlayoutDevice(int index) override | webrtc::VoEHardwareImpl | virtual |
| SetPlayoutSampleRate(unsigned int samples_per_sec) override | webrtc::VoEHardwareImpl | virtual |
| SetPlayoutSampleRate(unsigned int samples_per_sec) override | webrtc::VoEHardwareImpl | virtual |
| SetRecordingDevice(int index, StereoChannel recordingChannel=kStereoBoth) override | webrtc::VoEHardwareImpl | virtual |
| SetRecordingDevice(int index, StereoChannel recordingChannel=kStereoBoth) override | webrtc::VoEHardwareImpl | virtual |
| SetRecordingSampleRate(unsigned int samples_per_sec) override | webrtc::VoEHardwareImpl | virtual |
| SetRecordingSampleRate(unsigned int samples_per_sec) override | webrtc::VoEHardwareImpl | virtual |
| SetRecPayloadType(int channel, const CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| SetRecPayloadType(int channel, const CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| SetRTCP_CNAME(int channel, const char cName[256]) override | webrtc::VoERTP_RTCPImpl | virtual |
| SetRTCP_CNAME(int channel, const char cName[256]) override | webrtc::VoERTP_RTCPImpl | virtual |
| SetRTCPStatus(int channel, bool enable) override | webrtc::VoERTP_RTCPImpl | virtual |
| SetRTCPStatus(int channel, bool enable) override | webrtc::VoERTP_RTCPImpl | virtual |
| SetSendAudioLevelIndicationStatus(int channel, bool enable, unsigned char id) override | webrtc::VoERTP_RTCPImpl | virtual |
| SetSendAudioLevelIndicationStatus(int channel, bool enable, unsigned char id) override | webrtc::VoERTP_RTCPImpl | virtual |
| SetSendCNPayloadType(int channel, int type, PayloadFrequencies frequency=kFreq16000Hz) override | webrtc::VoECodecImpl | virtual |
| SetSendCNPayloadType(int channel, int type, PayloadFrequencies frequency=kFreq16000Hz) override | webrtc::VoECodecImpl | virtual |
| SetSendCodec(int channel, const CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| SetSendCodec(int channel, const CodecInst &codec) override | webrtc::VoECodecImpl | virtual |
| SetSpeakerVolume(unsigned int volume) override | webrtc::VoEVolumeControlImpl | virtual |
| SetSpeakerVolume(unsigned int volume) override | webrtc::VoEVolumeControlImpl | virtual |
| SetTraceCallback(TraceCallback *callback) | webrtc::VoiceEngine | static |
| SetTraceCallback(TraceCallback *callback) | webrtc::VoiceEngine | static |
| SetTraceFile(const char *fileNameUTF8, bool addFileCounter=false) | webrtc::VoiceEngine | static |
| SetTraceFile(const char *fileNameUTF8, bool addFileCounter=false) | webrtc::VoiceEngine | static |
| SetTraceFilter(unsigned int filter) | webrtc::VoiceEngine | static |
| SetTraceFilter(unsigned int filter) | webrtc::VoiceEngine | static |
| SetTypingDetectionParameters(int timeWindow, int costPerTyping, int reportingThreshold, int penaltyDecay, int typeEventDelay=0) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetTypingDetectionParameters(int timeWindow, int costPerTyping, int reportingThreshold, int penaltyDecay, int typeEventDelay=0) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetTypingDetectionStatus(bool enable) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetTypingDetectionStatus(bool enable) override | webrtc::VoEAudioProcessingImpl | virtual |
| SetVADStatus(int channel, bool enable, VadModes mode=kVadConventional, bool disableDTX=false) override | webrtc::VoECodecImpl | virtual |
| SetVADStatus(int channel, bool enable, VadModes mode=kVadConventional, bool disableDTX=false) override | webrtc::VoECodecImpl | virtual |
| SharedData() | webrtc::voe::SharedData | protected |
| SharedData() | webrtc::voe::SharedData | protected |
| StartDebugRecording(const char *fileNameUTF8) override | webrtc::VoEAudioProcessingImpl | virtual |
| StartDebugRecording(FILE *file_handle) override | webrtc::VoEAudioProcessingImpl | virtual |
| StartDebugRecording(const char *fileNameUTF8) override | webrtc::VoEAudioProcessingImpl | virtual |
| StartDebugRecording(FILE *file_handle) override | webrtc::VoEAudioProcessingImpl | virtual |
| StartPlayingFileAsMicrophone(int channel, const char fileNameUTF8[1024], bool loop=false, bool mixWithMicrophone=false, FileFormats format=kFileFormatPcm16kHzFile, float volumeScaling=1.0) override | webrtc::VoEFileImpl | virtual |
| StartPlayingFileAsMicrophone(int channel, InStream *stream, bool mixWithMicrophone=false, FileFormats format=kFileFormatPcm16kHzFile, float volumeScaling=1.0) override | webrtc::VoEFileImpl | virtual |
| StartPlayingFileAsMicrophone(int channel, const char fileNameUTF8[1024], bool loop=false, bool mixWithMicrophone=false, FileFormats format=kFileFormatPcm16kHzFile, float volumeScaling=1.0) override | webrtc::VoEFileImpl | virtual |
| StartPlayingFileAsMicrophone(int channel, InStream *stream, bool mixWithMicrophone=false, FileFormats format=kFileFormatPcm16kHzFile, float volumeScaling=1.0) override | webrtc::VoEFileImpl | virtual |
| StartPlayingFileLocally(int channel, const char fileNameUTF8[1024], bool loop=false, FileFormats format=kFileFormatPcm16kHzFile, float volumeScaling=1.0, int startPointMs=0, int stopPointMs=0) override | webrtc::VoEFileImpl | virtual |
| StartPlayingFileLocally(int channel, InStream *stream, FileFormats format=kFileFormatPcm16kHzFile, float volumeScaling=1.0, int startPointMs=0, int stopPointMs=0) override | webrtc::VoEFileImpl | virtual |
| StartPlayingFileLocally(int channel, const char fileNameUTF8[1024], bool loop=false, FileFormats format=kFileFormatPcm16kHzFile, float volumeScaling=1.0, int startPointMs=0, int stopPointMs=0) override | webrtc::VoEFileImpl | virtual |
| StartPlayingFileLocally(int channel, InStream *stream, FileFormats format=kFileFormatPcm16kHzFile, float volumeScaling=1.0, int startPointMs=0, int stopPointMs=0) override | webrtc::VoEFileImpl | virtual |
| StartPlayout(int channel) override | webrtc::VoEBaseImpl | virtual |
| StartPlayout(int channel) override | webrtc::VoEBaseImpl | virtual |
| StartReceive(int channel) override | webrtc::VoEBaseImpl | virtual |
| StartReceive(int channel) override | webrtc::VoEBaseImpl | virtual |
| StartRecordingMicrophone(const char *fileNameUTF8, CodecInst *compression=NULL, int maxSizeBytes=-1) override | webrtc::VoEFileImpl | virtual |
| StartRecordingMicrophone(OutStream *stream, CodecInst *compression=NULL) override | webrtc::VoEFileImpl | virtual |
| StartRecordingMicrophone(const char *fileNameUTF8, CodecInst *compression=NULL, int maxSizeBytes=-1) override | webrtc::VoEFileImpl | virtual |
| StartRecordingMicrophone(OutStream *stream, CodecInst *compression=NULL) override | webrtc::VoEFileImpl | virtual |
| StartRecordingPlayout(int channel, const char *fileNameUTF8, CodecInst *compression=NULL, int maxSizeBytes=-1) override | webrtc::VoEFileImpl | virtual |
| StartRecordingPlayout(int channel, OutStream *stream, CodecInst *compression=NULL) override | webrtc::VoEFileImpl | virtual |
| StartRecordingPlayout(int channel, const char *fileNameUTF8, CodecInst *compression=NULL, int maxSizeBytes=-1) override | webrtc::VoEFileImpl | virtual |
| StartRecordingPlayout(int channel, OutStream *stream, CodecInst *compression=NULL) override | webrtc::VoEFileImpl | virtual |
| StartSend(int channel) override | webrtc::VoEBaseImpl | virtual |
| StartSend(int channel) override | webrtc::VoEBaseImpl | virtual |
| statistics() | webrtc::voe::SharedData | inline |
| statistics() | webrtc::voe::SharedData | inline |
| StopDebugRecording() override | webrtc::VoEAudioProcessingImpl | virtual |
| StopDebugRecording() override | webrtc::VoEAudioProcessingImpl | virtual |
| StopPlayingFileAsMicrophone(int channel) override | webrtc::VoEFileImpl | virtual |
| StopPlayingFileAsMicrophone(int channel) override | webrtc::VoEFileImpl | virtual |
| StopPlayingFileLocally(int channel) override | webrtc::VoEFileImpl | virtual |
| StopPlayingFileLocally(int channel) override | webrtc::VoEFileImpl | virtual |
| StopPlayout(int channel) override | webrtc::VoEBaseImpl | virtual |
| StopPlayout(int channel) override | webrtc::VoEBaseImpl | virtual |
| StopReceive(int channel) | webrtc::VoEBase | inlinevirtual |
| StopReceive(int channel) | webrtc::VoEBase | inlinevirtual |
| StopRecordingMicrophone() override | webrtc::VoEFileImpl | virtual |
| StopRecordingMicrophone() override | webrtc::VoEFileImpl | virtual |
| StopRecordingPlayout(int channel) override | webrtc::VoEFileImpl | virtual |
| StopRecordingPlayout(int channel) override | webrtc::VoEFileImpl | virtual |
| StopSend(int channel) override | webrtc::VoEBaseImpl | virtual |
| StopSend(int channel) override | webrtc::VoEBaseImpl | virtual |
| Terminate() override | webrtc::VoEBaseImpl | virtual |
| Terminate() override | webrtc::VoEBaseImpl | virtual |
| TimeSinceLastTyping(int &seconds) override | webrtc::VoEAudioProcessingImpl | virtual |
| TimeSinceLastTyping(int &seconds) override | webrtc::VoEAudioProcessingImpl | virtual |
| webrtc::transmit_mixer() | webrtc::voe::SharedData | inline |
| webrtc::transmit_mixer() | webrtc::voe::SharedData | inline |
| webrtc::VoEBaseImpl::transmit_mixer() override | webrtc::VoEBaseImpl | inlinevirtual |
| VoEAudioProcessing() | webrtc::VoEAudioProcessing | inlineprotected |
| VoEAudioProcessing() | webrtc::VoEAudioProcessing | inlineprotected |
| VoEAudioProcessingImpl(voe::SharedData *shared) | webrtc::VoEAudioProcessingImpl | protected |
| VoEAudioProcessingImpl(voe::SharedData *shared) | webrtc::VoEAudioProcessingImpl | protected |
| VoEBase() | webrtc::VoEBase | inlineprotected |
| VoEBase() | webrtc::VoEBase | inlineprotected |
| VoEBaseImpl(voe::SharedData *shared) | webrtc::VoEBaseImpl | protected |
| VoEBaseImpl(voe::SharedData *shared) | webrtc::VoEBaseImpl | protected |
| VoECodec() | webrtc::VoECodec | inlineprotected |
| VoECodec() | webrtc::VoECodec | inlineprotected |
| VoECodecImpl(voe::SharedData *shared) | webrtc::VoECodecImpl | protected |
| VoECodecImpl(voe::SharedData *shared) | webrtc::VoECodecImpl | protected |
| VoEFile() | webrtc::VoEFile | inlineprotected |
| VoEFile() | webrtc::VoEFile | inlineprotected |
| VoEFileImpl(voe::SharedData *shared) | webrtc::VoEFileImpl | protected |
| VoEFileImpl(voe::SharedData *shared) | webrtc::VoEFileImpl | protected |
| VoEHardware() | webrtc::VoEHardware | inlineprotected |
| VoEHardware() | webrtc::VoEHardware | inlineprotected |
| VoEHardwareImpl(voe::SharedData *shared) | webrtc::VoEHardwareImpl | protected |
| VoEHardwareImpl(voe::SharedData *shared) | webrtc::VoEHardwareImpl | protected |
| VoENetEqStats() | webrtc::VoENetEqStats | inlineprotected |
| VoENetEqStats() | webrtc::VoENetEqStats | inlineprotected |
| VoENetEqStatsImpl(voe::SharedData *shared) | webrtc::VoENetEqStatsImpl | protected |
| VoENetEqStatsImpl(voe::SharedData *shared) | webrtc::VoENetEqStatsImpl | protected |
| VoENetwork() | webrtc::VoENetwork | inlineprotected |
| VoENetwork() | webrtc::VoENetwork | inlineprotected |
| VoENetworkImpl(voe::SharedData *shared) | webrtc::VoENetworkImpl | protected |
| VoENetworkImpl(voe::SharedData *shared) | webrtc::VoENetworkImpl | protected |
| VoERTP_RTCP() | webrtc::VoERTP_RTCP | inlineprotected |
| VoERTP_RTCP() | webrtc::VoERTP_RTCP | inlineprotected |
| VoERTP_RTCPImpl(voe::SharedData *shared) | webrtc::VoERTP_RTCPImpl | protected |
| VoERTP_RTCPImpl(voe::SharedData *shared) | webrtc::VoERTP_RTCPImpl | protected |
| VoEVolumeControl() | webrtc::VoEVolumeControl | inlineprotected |
| VoEVolumeControl() | webrtc::VoEVolumeControl | inlineprotected |
| VoEVolumeControlImpl(voe::SharedData *shared) | webrtc::VoEVolumeControlImpl | protected |
| VoEVolumeControlImpl(voe::SharedData *shared) | webrtc::VoEVolumeControlImpl | protected |
| VoiceActivityIndicator(int channel) override | webrtc::VoEAudioProcessingImpl | virtual |
| VoiceActivityIndicator(int channel) override | webrtc::VoEAudioProcessingImpl | virtual |
| VoiceEngine() | webrtc::VoiceEngine | inlineprotected |
| VoiceEngine() | webrtc::VoiceEngine | inlineprotected |
| VoiceEngineImpl() | webrtc::VoiceEngineImpl | inline |
| VoiceEngineImpl() | webrtc::VoiceEngineImpl | inline |
| WarningCode enum name | webrtc::AudioDeviceObserver | |
| WarningCode enum name | webrtc::AudioDeviceObserver | |
| ~AudioDeviceObserver() | webrtc::AudioDeviceObserver | inlineprotectedvirtual |
| ~AudioDeviceObserver() | webrtc::AudioDeviceObserver | inlineprotectedvirtual |
| ~AudioTransport() | webrtc::AudioTransport | inlineprotectedvirtual |
| ~AudioTransport() | webrtc::AudioTransport | inlineprotectedvirtual |
| ~MockVoiceEngine() | webrtc::test::MockVoiceEngine | inlinevirtual |
| ~MockVoiceEngine() | webrtc::test::MockVoiceEngine | inlinevirtual |
| ~SharedData() | webrtc::voe::SharedData | protectedvirtual |
| ~SharedData() | webrtc::voe::SharedData | protectedvirtual |
| ~VoEAudioProcessing() | webrtc::VoEAudioProcessing | inlineprotectedvirtual |
| ~VoEAudioProcessing() | webrtc::VoEAudioProcessing | inlineprotectedvirtual |
| ~VoEAudioProcessingImpl() override | webrtc::VoEAudioProcessingImpl | protected |
| ~VoEAudioProcessingImpl() override | webrtc::VoEAudioProcessingImpl | protected |
| ~VoEBase() | webrtc::VoEBase | inlineprotectedvirtual |
| ~VoEBase() | webrtc::VoEBase | inlineprotectedvirtual |
| ~VoEBaseImpl() override | webrtc::VoEBaseImpl | protected |
| ~VoEBaseImpl() override | webrtc::VoEBaseImpl | protected |
| ~VoECodec() | webrtc::VoECodec | inlineprotectedvirtual |
| ~VoECodec() | webrtc::VoECodec | inlineprotectedvirtual |
| ~VoECodecImpl() override | webrtc::VoECodecImpl | protected |
| ~VoECodecImpl() override | webrtc::VoECodecImpl | protected |
| ~VoEFile() | webrtc::VoEFile | inlineprotectedvirtual |
| ~VoEFile() | webrtc::VoEFile | inlineprotectedvirtual |
| ~VoEFileImpl() override | webrtc::VoEFileImpl | protected |
| ~VoEFileImpl() override | webrtc::VoEFileImpl | protected |
| ~VoEHardware() | webrtc::VoEHardware | inlineprotectedvirtual |
| ~VoEHardware() | webrtc::VoEHardware | inlineprotectedvirtual |
| ~VoEHardwareImpl() override | webrtc::VoEHardwareImpl | protected |
| ~VoEHardwareImpl() override | webrtc::VoEHardwareImpl | protected |
| ~VoENetEqStats() | webrtc::VoENetEqStats | inlineprotectedvirtual |
| ~VoENetEqStats() | webrtc::VoENetEqStats | inlineprotectedvirtual |
| ~VoENetEqStatsImpl() override | webrtc::VoENetEqStatsImpl | protected |
| ~VoENetEqStatsImpl() override | webrtc::VoENetEqStatsImpl | protected |
| ~VoENetwork() | webrtc::VoENetwork | inlineprotectedvirtual |
| ~VoENetwork() | webrtc::VoENetwork | inlineprotectedvirtual |
| ~VoENetworkImpl() override | webrtc::VoENetworkImpl | protected |
| ~VoENetworkImpl() override | webrtc::VoENetworkImpl | protected |
| ~VoERTP_RTCP() | webrtc::VoERTP_RTCP | inlineprotectedvirtual |
| ~VoERTP_RTCP() | webrtc::VoERTP_RTCP | inlineprotectedvirtual |
| ~VoERTP_RTCPImpl() override | webrtc::VoERTP_RTCPImpl | protected |
| ~VoERTP_RTCPImpl() override | webrtc::VoERTP_RTCPImpl | protected |
| ~VoEVolumeControl() | webrtc::VoEVolumeControl | inlineprotectedvirtual |
| ~VoEVolumeControl() | webrtc::VoEVolumeControl | inlineprotectedvirtual |
| ~VoEVolumeControlImpl() override | webrtc::VoEVolumeControlImpl | protected |
| ~VoEVolumeControlImpl() override | webrtc::VoEVolumeControlImpl | protected |
| ~VoiceEngine() | webrtc::VoiceEngine | inlineprotected |
| ~VoiceEngine() | webrtc::VoiceEngine | inlineprotected |
| ~VoiceEngineImpl() override | webrtc::VoiceEngineImpl | inline |
| ~VoiceEngineImpl() override | webrtc::VoiceEngineImpl | inline |
1.8.13