/external/tensorflow/tensorflow/lite/experimental/microfrontend/lib/ |
D | frontend_memmap_main.c | 32 int16_t* audio_data = malloc(audio_file_size * sizeof(int16_t)); in main() local 33 int16_t* original_audio_data = audio_data; in main() 35 fread(audio_data, sizeof(int16_t), audio_file_size, fp)) { in main() 44 frontend_state, audio_data, audio_file_size, &num_samples_read); in main() 45 audio_data += num_samples_read; in main()
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D | frontend_main.c | 42 int16_t* audio_data = malloc(audio_file_size * sizeof(int16_t)); in main() local 43 int16_t* original_audio_data = audio_data; in main() 45 fread(audio_data, sizeof(int16_t), audio_file_size, fp)) { in main() 54 &frontend_state, audio_data, audio_file_size, &num_samples_read); in main() 55 audio_data += num_samples_read; in main()
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/external/webrtc/audio/ |
D | audio_state_unittest.cc | 72 std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0); in Create10msTestData() local 77 audio_data[i * num_channels] = static_cast<int16_t>(32767.f * std::sin(w)); in Create10msTestData() 80 return audio_data; in Create10msTestData() 145 auto audio_data = Create10msTestData(kSampleRate, kNumChannels); in TEST() local 148 &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels, in TEST() 201 auto audio_data = Create10msTestData(kSampleRate, kNumChannels); in TEST() local 204 &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels, in TEST() 236 auto audio_data = Create10msTestData(kSampleRate, kNumChannels); in TEST() local 239 &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels, in TEST()
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D | audio_transport_impl.cc | 97 const void* audio_data, in RecordedDataIsAvailable() argument 107 RTC_DCHECK(audio_data); in RecordedDataIsAvailable() 130 voe::RemixAndResample(static_cast<const int16_t*>(audio_data), in RecordedDataIsAvailable() 215 void* audio_data, in PullRenderData() argument 233 static_cast<int16_t*>(audio_data)); in PullRenderData()
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/external/python/cpython3/Lib/test/ |
D | test_winsound.py | 108 audio_data = f.read() 109 safe_PlaySound(audio_data, winsound.SND_MEMORY) 110 audio_data = bytearray(audio_data) 111 safe_PlaySound(audio_data, winsound.SND_MEMORY)
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/external/autotest/client/cros/audio/ |
D | audio_test_data.py | 11 from autotest_lib.client.cros.audio import audio_data 109 bits_src = audio_data.SAMPLE_FORMATS[ 124 bits_dst=audio_data.SAMPLE_FORMATS[ 193 sample_format = audio_data.SAMPLE_FORMATS[data_format['sample_format']]
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D | check_quality.py | 27 from autotest_lib.client.cros.audio import audio_data 32 import audio_data 251 self.raw_data = audio_data.AudioRawData( 311 saturate_value = audio_data.get_maximum_value_from_sample_format( 449 raw_data = audio_data.AudioRawData(
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/external/autotest/client/cros/multimedia/ |
D | bluetooth_facade_native.py | 2465 def start_capturing_audio_subprocess(self, audio_data, recording_device): argument 2474 audio_data = json.loads(audio_data) 2476 audio_data[recording_device], 2477 sample_format=audio_data['format'], 2478 channels=audio_data['channels'], 2479 rate=audio_data['rate'], 2480 duration=audio_data['duration']) 2489 def _generate_playback_file(self, audio_data): argument 2497 if not os.path.exists(audio_data['file']): 2500 channel=audio_data['channels'], [all …]
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/external/webrtc/modules/audio_device/android/ |
D | aaudio_player.cc | 154 aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data, in OnDataCallback() argument 194 memset(audio_data, 0, num_bytes); in OnDataCallback() 197 rtc::MakeArrayView(static_cast<int16_t*>(audio_data), in OnDataCallback()
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D | aaudio_recorder.cc | 154 void* audio_data, in OnDataCallback() argument 166 aaudio_.ClearInputStream(audio_data, num_frames); in OnDataCallback() 187 rtc::MakeArrayView(static_cast<const int16_t*>(audio_data), in OnDataCallback()
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D | aaudio_wrapper.cc | 102 void* audio_data, in DataCallback() argument 105 RTC_DCHECK(audio_data); in DataCallback() 108 return aaudio_wrapper->observer()->OnDataCallback(audio_data, num_frames); in DataCallback() 270 void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) { in ClearInputStream() argument 277 cleared_frames = AAudioStream_read(stream_, audio_data, num_frames, 0); in ClearInputStream()
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D | aaudio_wrapper.h | 31 virtual aaudio_data_callback_result_t OnDataCallback(void* audio_data, 85 void ClearInputStream(void* audio_data, int32_t num_frames);
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/external/webrtc/sdk/android/src/jni/audio_device/ |
D | aaudio_recorder.cc | 166 void* audio_data, in OnDataCallback() argument 178 aaudio_.ClearInputStream(audio_data, num_frames); in OnDataCallback() 199 rtc::MakeArrayView(static_cast<const int16_t*>(audio_data), in OnDataCallback()
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D | aaudio_player.cc | 170 aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data, in OnDataCallback() argument 210 memset(audio_data, 0, num_bytes); in OnDataCallback() 213 rtc::MakeArrayView(static_cast<int16_t*>(audio_data), in OnDataCallback()
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D | aaudio_wrapper.cc | 103 void* audio_data, in DataCallback() argument 106 RTC_DCHECK(audio_data); in DataCallback() 109 return aaudio_wrapper->observer()->OnDataCallback(audio_data, num_frames); in DataCallback() 270 void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) { in ClearInputStream() argument 277 cleared_frames = AAudioStream_read(stream_, audio_data, num_frames, 0); in ClearInputStream()
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D | aaudio_wrapper.h | 31 virtual aaudio_data_callback_result_t OnDataCallback(void* audio_data, 85 void ClearInputStream(void* audio_data, int32_t num_frames);
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/external/webrtc/modules/audio_device/win/ |
D | core_audio_input_win.cc | 297 uint8_t* audio_data; in OnDataCallback() local 302 error = audio_capture_client_->GetBuffer(&audio_data, &num_frames_to_read, in OnDataCallback() 352 rtc::ExplicitZeroMemory(audio_data, in OnDataCallback() 359 rtc::MakeArrayView(reinterpret_cast<const int16_t*>(audio_data), in OnDataCallback()
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D | core_audio_output_win.cc | 319 uint8_t* audio_data; in OnDataCallback() local 320 error = audio_render_client_->GetBuffer(num_requested_frames, &audio_data); in OnDataCallback() 340 rtc::MakeArrayView(reinterpret_cast<int16_t*>(audio_data), in OnDataCallback()
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/external/autotest/server/site_tests/audiovideo_AVSync/ |
D | audiovideo_AVSync.py | 209 audio_data = open(local_path).read() 212 logging.info("audio capture %d bytes, %f seconds", len(audio_data), 213 len(audio_data) / float(self.AUDIO_CAPTURE_RATE) / 32) 217 key_audio = self.compute_audio_keypoint(audio_data)
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/external/webrtc/api/ |
D | media_stream_interface.h | 195 virtual void OnData(const void* audio_data, in OnData() argument 207 virtual void OnData(const void* audio_data, in OnData() argument 215 return OnData(audio_data, bits_per_sample, sample_rate, number_of_channels, in OnData()
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/external/autotest/server/cros/bluetooth/ |
D | bluetooth_device.py | 1137 def start_capturing_audio_subprocess(self, audio_data, recording_device): argument 1147 json.dumps(audio_data), recording_device) 1160 def start_playing_audio_subprocess(self, audio_data): argument 1167 audio_data = json.dumps(audio_data) 1168 return self._proxy.start_playing_audio_subprocess(audio_data) 1181 def play_audio(self, audio_data): argument 1190 return self._proxy.play_audio(json.dumps(audio_data))
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/external/tensorflow/tensorflow/examples/speech_commands/ |
D | test_streaming_accuracy.cc | 217 std::vector<float> audio_data; in main() local 222 wav_string, &audio_data, &sample_count, &channel_count, &sample_rate); in main() 250 const float* input_start = &(audio_data[audio_data_offset]); in main()
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/external/tensorflow/tensorflow/lite/experimental/microfrontend/ |
D | audio_microfrontend.cc | 125 const int16_t* audio_data = GetTensorData<int16_t>(input); in GenerateFeatures() local 142 data->state, audio_data, audio_size, &num_samples_read); in GenerateFeatures() 143 audio_data += num_samples_read; in GenerateFeatures()
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/external/tensorflow/tensorflow/lite/experimental/microfrontend/ops/ |
D | audio_microfrontend_op.cc | 213 auto audio_data = in Compute() local 247 &state, audio_data, audio_size, &num_samples_read); in Compute() 248 audio_data += num_samples_read; in Compute()
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/external/webrtc/pc/ |
D | rtp_sender.h | 217 void OnData(const void* audio_data, 225 void OnData(const void* audio_data, in OnData() argument 230 OnData(audio_data, bits_per_sample, sample_rate, number_of_channels, in OnData()
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