| /external/webrtc/talk/media/base/ |
| audiorenderer.h | 41 virtual void OnData(const void* audio_data,
|
| fakemediaengine.h | 368 void OnData(const void* audio_data,
|
| /external/opencv3/3rdparty/include/ffmpeg_/libavutil/ |
| samplefmt.h | 164 * The audio_data array is filled with the pointers to the samples data planes:
174 * otherwise the audio_data pointers will point to invalid data.
179 * @param[out] audio_data array to be filled with the pointer for each channel
190 int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
198 * The allocated samples buffer can be freed by using av_freep(&audio_data[0])
204 * @param[out] audio_data array to be filled with the pointer for each channel
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,
247 * @param audio_data array of pointers to data planes
253 int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples
[all...] |
| /external/webrtc/webrtc/voice_engine/ |
| 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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| voe_base_impl.cc | 114 const int16_t* audio_data, int sample_rate,
123 voe_channels, number_of_voe_channels, audio_data, sample_rate,
134 PushCaptureData(voe_channels[i], audio_data, 16, sample_rate,
142 void VoEBaseImpl::OnData(int voe_channel, const void* audio_data,
145 PushCaptureData(voe_channel, audio_data, bits_per_sample, sample_rate,
149 void VoEBaseImpl::PushCaptureData(int voe_channel, const void* audio_data,
158 channel_ptr->Demultiplex(static_cast<const int16_t*>(audio_data),
169 void* audio_data, int64_t* elapsed_time_ms,
175 audio_data, elapsed_time_ms, ntp_time_ms);
702 const void* audio_data, uint32_t sample_rate, size_t number_of_channels
[all...] |
| channel.h | 449 // between this method and the overloaded method above is that |audio_data|
451 void Demultiplex(const int16_t* audio_data,
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| channel.cc | [all...] |
| /external/webrtc/webrtc/modules/audio_device/include/ |
| audio_device_defines.h | 86 const int16_t* audio_data,
103 const void* audio_data,
116 const void* audio_data,
130 void* audio_data,
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| /external/autotest/client/cros/audio/ |
| audio_test_data.py | 10 from autotest_lib.client.cros.audio import audio_data namespace
29 audio_data.SAMPLE_FORMAT.
85 bits_src=audio_data.SAMPLE_FORMATS[
90 bits_dst=audio_data.SAMPLE_FORMATS[
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| audio_analysis_unittest.py | 8 from autotest_lib.client.cros.audio import audio_data namespace
44 data = audio_data.AudioRawData(binary, 2, 'S32_LE')
45 saturate_value = audio_data.get_maximum_value_from_sample_format(
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| audio_helper.py | 20 from autotest_lib.client.cros.audio import audio_data namespace
804 raw_data = audio_data.AudioRawData(
812 bits = 8 * (audio_data.SAMPLE_FORMATS[
[all...] |
| /external/autotest/client/cros/chameleon/ |
| audio_test_utils.py | 19 from autotest_lib.client.cros.audio import audio_data namespace
270 recorded_data = audio_data.AudioRawData(
283 saturate_value = audio_data.get_maximum_value_from_sample_format(
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| audio_widget.py | 12 from autotest_lib.client.cros.audio import audio_data namespace
182 return audio_data.SAMPLE_FORMATS[
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| /external/webrtc/talk/app/webrtc/ |
| mediastreaminterface.h | 151 virtual void OnData(const void* audio_data,
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| rtpsender.h | 57 void OnData(const void* audio_data,
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| rtpsender.cc | 44 void LocalAudioSinkAdapter::OnData(const void* audio_data,
51 sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
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| /device/google/dragon/audio/hal/ |
| audio_hw.c | 2146 const void *audio_data; local
[all...] |
| /external/webrtc/talk/media/webrtc/ |
| webrtcvoiceengine.cc | [all...] |
