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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::MockAudioTransport, including all inherited members.
| 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 |
| 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 |
| 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 |
| 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 |
1.8.13