webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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This is the complete list of members for webrtc::MockAudioTransportAndroid, including all inherited members.
HandleCallbacks(EventWrapper *test_is_done, AudioStreamInterface *audio_stream, int num_callbacks) | webrtc::MockAudioTransportAndroid | inline |
MOCK_METHOD10(RecordedDataIsAvailable, int32_t(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)) | webrtc::test::MockAudioTransport | |
MOCK_METHOD10(RecordedDataIsAvailable, int32_t(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)) | webrtc::test::MockAudioTransport | |
MOCK_METHOD6(PushCaptureData, void(int voe_channel, const void *audio_data, int bits_per_sample, int sample_rate, size_t number_of_channels, size_t number_of_frames)) | webrtc::test::MockAudioTransport | |
MOCK_METHOD6(PushCaptureData, void(int voe_channel, const void *audio_data, int bits_per_sample, int sample_rate, size_t number_of_channels, size_t number_of_frames)) | webrtc::test::MockAudioTransport | |
MOCK_METHOD7(PullRenderData, void(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)) | webrtc::test::MockAudioTransport | |
MOCK_METHOD7(PullRenderData, void(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)) | webrtc::test::MockAudioTransport | |
MOCK_METHOD8(NeedMorePlayData, int32_t(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)) | webrtc::test::MockAudioTransport | |
MOCK_METHOD8(NeedMorePlayData, int32_t(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)) | webrtc::test::MockAudioTransport | |
MockAudioTransport() | webrtc::test::MockAudioTransport | inline |
MockAudioTransport() | webrtc::test::MockAudioTransport | inline |
MockAudioTransportAndroid(int type) | webrtc::MockAudioTransportAndroid | inlineexplicit |
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)=0 | webrtc::AudioTransport | pure 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)=0 | webrtc::AudioTransport | pure virtual |
play_mode() const | webrtc::MockAudioTransportAndroid | 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)=0 | webrtc::AudioTransport | pure 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)=0 | webrtc::AudioTransport | pure 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)=0 | webrtc::AudioTransport | pure 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)=0 | webrtc::AudioTransport | pure virtual |
RealNeedMorePlayData(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) | webrtc::MockAudioTransportAndroid | inline |
RealRecordedDataIsAvailable(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) | webrtc::MockAudioTransportAndroid | inline |
rec_mode() const | webrtc::MockAudioTransportAndroid | inline |
ReceivedEnoughCallbacks() | webrtc::MockAudioTransportAndroid | inline |
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)=0 | webrtc::AudioTransport | pure 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)=0 | webrtc::AudioTransport | pure virtual |
~AudioTransport() | webrtc::AudioTransport | inlineprotectedvirtual |
~AudioTransport() | webrtc::AudioTransport | inlineprotectedvirtual |
~MockAudioTransport() | webrtc::test::MockAudioTransport | inline |
~MockAudioTransport() | webrtc::test::MockAudioTransport | inline |
~MockAudioTransportAndroid() | webrtc::MockAudioTransportAndroid | inlinevirtual |