/external/webrtc/audio/ |
D | audio_state_unittest.cc | 72 std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0); in Create10msTestData() local 145 auto audio_data = Create10msTestData(kSampleRate, kNumChannels); in TEST() local 201 auto audio_data = Create10msTestData(kSampleRate, kNumChannels); in TEST() local 236 auto audio_data = Create10msTestData(kSampleRate, kNumChannels); in TEST() local
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D | audio_transport_impl.cc | 97 const void* audio_data, in RecordedDataIsAvailable() 215 void* audio_data, in PullRenderData()
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/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
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D | frontend_main.c | 42 int16_t* audio_data = malloc(audio_file_size * sizeof(int16_t)); in main() local
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/external/webrtc/api/ |
D | media_stream_interface.h | 195 virtual void OnData(const void* audio_data, in OnData() 207 virtual void OnData(const void* audio_data, in OnData()
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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()
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D | aaudio_wrapper.cc | 102 void* audio_data, in DataCallback() 270 void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) { in ClearInputStream()
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D | aaudio_recorder.cc | 154 void* audio_data, in OnDataCallback()
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/external/webrtc/sdk/android/src/jni/audio_device/ |
D | aaudio_wrapper.cc | 103 void* audio_data, in DataCallback() 270 void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) { in ClearInputStream()
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D | aaudio_recorder.cc | 166 void* 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()
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/external/tensorflow/tensorflow/lite/experimental/microfrontend/ops/ |
D | audio_microfrontend_op.cc | 213 auto audio_data = in Compute() local
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/external/webrtc/modules/audio_device/win/ |
D | core_audio_output_win.cc | 319 uint8_t* audio_data; in OnDataCallback() local
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D | core_audio_input_win.cc | 297 uint8_t* audio_data; in OnDataCallback() local
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/external/webrtc/pc/ |
D | rtp_sender.h | 225 void OnData(const void* audio_data, in OnData()
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D | rtp_sender.cc | 390 const void* audio_data, in OnData()
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/external/webrtc/pc/test/ |
D | fake_audio_capture_module_unittest.cc | 67 void* audio_data, in PullRenderData()
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/external/tensorflow/tensorflow/examples/speech_commands/ |
D | test_streaming_accuracy.cc | 217 std::vector<float> audio_data; in main() local
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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
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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 1160 def start_playing_audio_subprocess(self, audio_data): argument 1181 def play_audio(self, audio_data): argument
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/external/webrtc/modules/audio_processing/agc/ |
D | agc_manager_direct.cc | 538 std::array<int16_t, AudioBuffer::kMaxSampleRate / 100> audio_data; in Process() local
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D | agc_manager_direct_unittest.cc | 130 std::vector<float> audio_data; member in webrtc::AgcManagerDirectTest
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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 2489 def _generate_playback_file(self, audio_data): argument 2513 def start_playing_audio_subprocess(self, audio_data): argument 2539 def play_audio(self, audio_data): argument
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/external/webrtc/examples/unityplugin/ |
D | simple_peer_connection.cc | 555 void SimplePeerConnection::OnData(const void* audio_data, in OnData()
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/external/webrtc/modules/audio_device/ |
D | audio_device_data_observer.cc | 110 void* audio_data, in PullRenderData()
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