/external/webrtc/webrtc/voice_engine/ |
D | voe_base_impl.h | 77 const int16_t* audio_data, 86 const void* audio_data, 92 const void* audio_data, 101 void* audio_data, 128 const void* audio_data, uint32_t sample_rate, size_t number_of_channels, 134 void* audio_data, int64_t* elapsed_time_ms,
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D | voe_base_impl.cc | 114 const int16_t* audio_data, int sample_rate, in OnDataAvailable() argument 123 voe_channels, number_of_voe_channels, audio_data, sample_rate, in OnDataAvailable() 134 PushCaptureData(voe_channels[i], audio_data, 16, sample_rate, in OnDataAvailable() 142 void VoEBaseImpl::OnData(int voe_channel, const void* audio_data, in OnData() argument 145 PushCaptureData(voe_channel, audio_data, bits_per_sample, sample_rate, in OnData() 149 void VoEBaseImpl::PushCaptureData(int voe_channel, const void* audio_data, in PushCaptureData() argument 158 channel_ptr->Demultiplex(static_cast<const int16_t*>(audio_data), in PushCaptureData() 169 void* audio_data, int64_t* elapsed_time_ms, in PullRenderData() argument 175 audio_data, elapsed_time_ms, ntp_time_ms); in PullRenderData() 702 const void* audio_data, uint32_t sample_rate, size_t number_of_channels, in ProcessRecordedDataWithAPM() argument [all …]
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D | channel.h | 451 void Demultiplex(const int16_t* audio_data,
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D | channel.cc | 3355 void Channel::Demultiplex(const int16_t* audio_data, in Demultiplex() argument 3366 RemixAndResample(audio_data, number_of_frames, number_of_channels, in Demultiplex()
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/external/opencv3/3rdparty/include/ffmpeg_/libavutil/ |
D | samplefmt.h | 190 int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, 214 int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels, 226 int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, 253 int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
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/external/webrtc/webrtc/modules/audio_device/include/ |
D | audio_device_defines.h | 86 const int16_t* audio_data, in OnDataAvailable() argument 103 const void* audio_data, in OnData() argument 116 const void* audio_data, in PushCaptureData() argument 130 void* audio_data, in PullRenderData() argument
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/external/autotest/client/cros/audio/ |
D | audio_test_data.py | 10 from autotest_lib.client.cros.audio import audio_data 85 bits_src=audio_data.SAMPLE_FORMATS[ 90 bits_dst=audio_data.SAMPLE_FORMATS[
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D | audio_analysis_unittest.py | 8 from autotest_lib.client.cros.audio import audio_data 44 data = audio_data.AudioRawData(binary, 2, 'S32_LE') 45 saturate_value = audio_data.get_maximum_value_from_sample_format(
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D | audio_helper.py | 20 from autotest_lib.client.cros.audio import audio_data 804 raw_data = audio_data.AudioRawData( 812 bits = 8 * (audio_data.SAMPLE_FORMATS[ 921 golden_data = audio_data.AudioRawData( 925 test_data = audio_data.AudioRawData(
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/external/autotest/client/cros/chameleon/ |
D | audio_test_utils.py | 19 from autotest_lib.client.cros.audio import audio_data 270 recorded_data = audio_data.AudioRawData( 283 saturate_value = audio_data.get_maximum_value_from_sample_format(
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D | audio_widget.py | 12 from autotest_lib.client.cros.audio import audio_data 182 return audio_data.SAMPLE_FORMATS[
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/external/webrtc/talk/media/base/ |
D | audiorenderer.h | 41 virtual void OnData(const void* audio_data,
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D | fakemediaengine.h | 368 void OnData(const void* audio_data, in OnData() argument
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/external/webrtc/talk/app/webrtc/ |
D | rtpsender.cc | 44 void LocalAudioSinkAdapter::OnData(const void* audio_data, in OnData() argument 51 sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels, in OnData()
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D | rtpsender.h | 57 void OnData(const void* audio_data,
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D | mediastreaminterface.h | 151 virtual void OnData(const void* audio_data,
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/external/webrtc/talk/media/webrtc/ |
D | webrtcvoiceengine.cc | 1155 void OnData(const void* audio_data, in OnData() argument 1164 audio_data, in OnData()
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