| AudioDeviceBuffer() | webrtc::AudioDeviceBuffer | |
| AudioDeviceBuffer() | webrtc::AudioDeviceBuffer | |
| DeliverRecordedData() | webrtc::AudioDeviceBuffer | virtual |
| DeliverRecordedData() | webrtc::AudioDeviceBuffer | virtual |
| GetPlayoutData(void *audio_buffer) | webrtc::AudioDeviceBuffer | virtual |
| GetPlayoutData(void *audio_buffer) | webrtc::AudioDeviceBuffer | virtual |
| LOG_ACTIVE enum value | webrtc::AudioDeviceBuffer | |
| LOG_START enum value | webrtc::AudioDeviceBuffer | |
| LOG_STOP enum value | webrtc::AudioDeviceBuffer | |
| LogState enum name | webrtc::AudioDeviceBuffer | |
| LogState enum name | webrtc::AudioDeviceBuffer | |
| MOCK_METHOD0(DeliverRecordedData, int32_t()) | webrtc::MockAudioDeviceBuffer | |
| MOCK_METHOD0(DeliverRecordedData, int32_t()) | webrtc::MockAudioDeviceBuffer | |
| MOCK_METHOD1(RequestPlayoutData, int32_t(size_t nSamples)) | webrtc::MockAudioDeviceBuffer | |
| MOCK_METHOD1(GetPlayoutData, int32_t(void *audioBuffer)) | webrtc::MockAudioDeviceBuffer | |
| MOCK_METHOD1(RequestPlayoutData, int32_t(size_t nSamples)) | webrtc::MockAudioDeviceBuffer | |
| MOCK_METHOD1(GetPlayoutData, int32_t(void *audioBuffer)) | webrtc::MockAudioDeviceBuffer | |
| MOCK_METHOD2(SetRecordedBuffer, int32_t(const void *audioBuffer, size_t nSamples)) | webrtc::MockAudioDeviceBuffer | |
| MOCK_METHOD2(SetRecordedBuffer, int32_t(const void *audioBuffer, size_t nSamples)) | webrtc::MockAudioDeviceBuffer | |
| MOCK_METHOD3(SetVQEData, void(int playDelayMS, int recDelayMS, int clockDrift)) | webrtc::MockAudioDeviceBuffer | |
| MOCK_METHOD3(SetVQEData, void(int playDelayMS, int recDelayMS, int clockDrift)) | webrtc::MockAudioDeviceBuffer | |
| MockAudioDeviceBuffer() | webrtc::MockAudioDeviceBuffer | inline |
| MockAudioDeviceBuffer() | webrtc::MockAudioDeviceBuffer | inline |
| NewMicLevel() const | webrtc::AudioDeviceBuffer | |
| NewMicLevel() const | webrtc::AudioDeviceBuffer | |
| PlayoutChannels() const | webrtc::AudioDeviceBuffer | |
| PlayoutChannels() const | webrtc::AudioDeviceBuffer | |
| PlayoutSampleRate() const | webrtc::AudioDeviceBuffer | |
| PlayoutSampleRate() const | webrtc::AudioDeviceBuffer | |
| RecordingChannel(AudioDeviceModule::ChannelType &channel) const | webrtc::AudioDeviceBuffer | |
| RecordingChannel(AudioDeviceModule::ChannelType &channel) const | webrtc::AudioDeviceBuffer | |
| RecordingChannels() const | webrtc::AudioDeviceBuffer | |
| RecordingChannels() const | webrtc::AudioDeviceBuffer | |
| RecordingSampleRate() const | webrtc::AudioDeviceBuffer | |
| RecordingSampleRate() const | webrtc::AudioDeviceBuffer | |
| RegisterAudioCallback(AudioTransport *audio_callback) | webrtc::AudioDeviceBuffer | |
| RegisterAudioCallback(AudioTransport *audio_callback) | webrtc::AudioDeviceBuffer | |
| RequestPlayoutData(size_t samples_per_channel) | webrtc::AudioDeviceBuffer | virtual |
| RequestPlayoutData(size_t samples_per_channel) | webrtc::AudioDeviceBuffer | virtual |
| SetCurrentMicLevel(uint32_t level) | webrtc::AudioDeviceBuffer | |
| SetCurrentMicLevel(uint32_t level) | webrtc::AudioDeviceBuffer | |
| SetId(uint32_t id) | webrtc::AudioDeviceBuffer | inline |
| SetId(uint32_t id) | webrtc::AudioDeviceBuffer | inline |
| SetPlayoutChannels(size_t channels) | webrtc::AudioDeviceBuffer | |
| SetPlayoutChannels(size_t channels) | webrtc::AudioDeviceBuffer | |
| SetPlayoutSampleRate(uint32_t fsHz) | webrtc::AudioDeviceBuffer | |
| SetPlayoutSampleRate(uint32_t fsHz) | webrtc::AudioDeviceBuffer | |
| SetRecordedBuffer(const void *audio_buffer, size_t samples_per_channel) | webrtc::AudioDeviceBuffer | virtual |
| SetRecordedBuffer(const void *audio_buffer, size_t samples_per_channel) | webrtc::AudioDeviceBuffer | virtual |
| SetRecordingChannel(const AudioDeviceModule::ChannelType channel) | webrtc::AudioDeviceBuffer | |
| SetRecordingChannel(const AudioDeviceModule::ChannelType channel) | webrtc::AudioDeviceBuffer | |
| SetRecordingChannels(size_t channels) | webrtc::AudioDeviceBuffer | |
| SetRecordingChannels(size_t channels) | webrtc::AudioDeviceBuffer | |
| SetRecordingSampleRate(uint32_t fsHz) | webrtc::AudioDeviceBuffer | |
| SetRecordingSampleRate(uint32_t fsHz) | webrtc::AudioDeviceBuffer | |
| SetTypingStatus(bool typing_status) | webrtc::AudioDeviceBuffer | |
| SetTypingStatus(bool typing_status) | webrtc::AudioDeviceBuffer | |
| SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift) | webrtc::AudioDeviceBuffer | virtual |
| SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift) | webrtc::AudioDeviceBuffer | virtual |
| StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]) | webrtc::AudioDeviceBuffer | |
| StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]) | webrtc::AudioDeviceBuffer | |
| StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]) | webrtc::AudioDeviceBuffer | |
| StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]) | webrtc::AudioDeviceBuffer | |
| StartPlayout() | webrtc::AudioDeviceBuffer | |
| StartPlayout() | webrtc::AudioDeviceBuffer | |
| StartRecording() | webrtc::AudioDeviceBuffer | |
| StartRecording() | webrtc::AudioDeviceBuffer | |
| StopInputFileRecording() | webrtc::AudioDeviceBuffer | |
| StopInputFileRecording() | webrtc::AudioDeviceBuffer | |
| StopOutputFileRecording() | webrtc::AudioDeviceBuffer | |
| StopOutputFileRecording() | webrtc::AudioDeviceBuffer | |
| StopPlayout() | webrtc::AudioDeviceBuffer | |
| StopPlayout() | webrtc::AudioDeviceBuffer | |
| StopRecording() | webrtc::AudioDeviceBuffer | |
| StopRecording() | webrtc::AudioDeviceBuffer | |
| ~AudioDeviceBuffer() | webrtc::AudioDeviceBuffer | virtual |
| ~AudioDeviceBuffer() | webrtc::AudioDeviceBuffer | virtual |
| ~MockAudioDeviceBuffer() | webrtc::MockAudioDeviceBuffer | inlinevirtual |
| ~MockAudioDeviceBuffer() | webrtc::MockAudioDeviceBuffer | inlinevirtual |
