| /external/libwebm/webm_parser/tests/ |
| D | audio_parser_test.cc | 27 const Audio audio = parser_.value(); in TEST_F() local 59 const Audio audio = parser_.value(); in TEST_F() local 95 const Audio audio = parser_.value(); in TEST_F() local 119 const Audio audio = parser_.value(); in TEST_F() local 146 const Audio audio = parser_.value(); in TEST_F() local
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| /external/ltp/testcases/kernel/device-drivers/v4l/user_space/ |
| D | test_VIDIOC_ENUMAUDIO.c | 41 struct v4l2_audio audio; in test_VIDIOC_ENUMAUDIO() local 110 struct v4l2_audio audio; in test_VIDIOC_ENUMAUDIO_S32_MAX() local 130 struct v4l2_audio audio; in test_VIDIOC_ENUMAUDIO_S32_MAX_1() local 150 struct v4l2_audio audio; in test_VIDIOC_ENUMAUDIO_U32_MAX() local 171 struct v4l2_audio audio; in test_VIDIOC_ENUMAUDIO_NULL() local
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| D | test_VIDIOC_AUDIO.c | 67 struct v4l2_audio audio; in test_VIDIOC_G_AUDIO() local 129 struct v4l2_audio audio; in test_VIDIOC_G_AUDIO_ignore_index() local 166 struct v4l2_audio audio; in test_VIDIOC_G_AUDIO_NULL() local 296 struct v4l2_audio audio; in test_VIDIOC_S_AUDIO_S32_MAX() local 353 struct v4l2_audio audio; in test_VIDIOC_S_AUDIO_S32_MAX_1() local 410 struct v4l2_audio audio; in test_VIDIOC_S_AUDIO_U32_MAX() local
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| /external/webrtc/test/pc/e2e/ |
| D | peer_connection_e2e_smoke_test.cc | 139 AudioConfig audio; in TEST_F() local 161 AudioConfig audio; in TEST_F() local 212 AudioConfig audio; in TEST_F() local 232 AudioConfig audio; in TEST_F() local 263 AudioConfig audio; in TEST_F() local 282 AudioConfig audio; in TEST_F() local 414 AudioConfig audio; in TEST_F() local 423 AudioConfig audio; in TEST_F() local 451 AudioConfig audio; in TEST_F() local 480 AudioConfig audio("alice-audio"); in TEST_F() local [all …]
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| /external/vboot_reference/firmware/lib/ |
| D | vboot_audio.c | 62 static void VbGetDevMusicNotes(VbAudioContext *audio, int use_short) in VbGetDevMusicNotes() 212 VbAudioContext *audio = &au; in VbAudioOpen() local 256 int VbAudioLooping(VbAudioContext *audio) in VbAudioLooping() 295 void VbAudioClose(VbAudioContext *audio) in VbAudioClose()
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| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | audio_sink.cc | 16 bool AudioSinkFork::WriteArray(const int16_t* audio, size_t num_samples) { in WriteArray() 21 bool VoidAudioSink::WriteArray(const int16_t* audio, size_t num_samples) { in WriteArray()
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| D | output_audio_file.h | 40 bool WriteArray(const int16_t* audio, size_t num_samples) override { in WriteArray()
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| D | audio_checksum.h | 36 bool WriteArray(const int16_t* audio, size_t num_samples) override { in WriteArray()
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| D | output_wav_file.h | 35 bool WriteArray(const int16_t* audio, size_t num_samples) override { in WriteArray()
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| /external/webrtc/modules/audio_coding/codecs/g711/ |
| D | audio_encoder_pcm.cc | 64 rtc::ArrayView<const int16_t> audio, in EncodeImpl() 98 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, in EncodeCall() 112 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, in EncodeCall()
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| /external/webrtc/modules/audio_processing/ns/ |
| D | noise_suppressor_unittest.cc | 38 AudioBuffer* audio) { in PopulateInputFrameWithIdenticalChannels() 52 const AudioBuffer& audio) { in VerifyIdenticalChannels() 78 AudioBuffer audio(rate, num_channels, rate, num_channels, rate, in TEST() local
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| /external/tensorflow/tensorflow/core/lib/wav/ |
| D | wav_io_test.cc | 36 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f}; in TEST() local 72 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f}; in TEST() local 83 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f}; in TEST() local 90 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f}; in TEST() local 131 float audio[] = {0.0f, 1.0f, 0.0f, -1.0f}; in TEST() local 160 float audio[] = {0.0f, 1.0f, 0.0f, -1.0f}; in TEST() local
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| /external/webrtc/test/ |
| D | mock_audio_encoder.cc | 28 rtc::ArrayView<const int16_t> audio, in operator ()() 49 rtc::ArrayView<const int16_t> audio, in operator ()()
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| /external/webrtc/modules/audio_processing/ |
| D | high_pass_filter.cc | 67 void HighPassFilter::Process(AudioBuffer* audio, bool use_split_band_data) { in Process() 85 void HighPassFilter::Process(std::vector<std::vector<float>>* audio) { in Process()
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| D | gain_control_impl.cc | 117 const AudioBuffer& audio, in PackRenderAudioBuffer() 144 int GainControlImpl::AnalyzeCaptureAudio(const AudioBuffer& audio) { in AnalyzeCaptureAudio() 191 int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio, in ProcessCaptureAudio()
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| /external/tensorflow/tensorflow/core/kernels/ |
| D | decode_wav_op_test.cc | 74 const Tensor& audio = outputs[0]; in TEST() local 165 const Tensor& audio = outputs[0]; in TEST() local
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| D | encode_wav_op.cc | 35 const Tensor& audio = context->input(0); in Compute() local
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| /external/autotest/server/site_tests/bluetooth_AdapterMTBF/ |
| D | bluetooth_AdapterMTBF.py | 83 def run_typical_use_cases(self, mouse, phone, audio, keyboard): argument 106 def test_mouse_and_audio(self, mouse, audio): argument 140 def test_hid_and_audio(self, mouse, keyboard, audio): argument
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| /external/armnn/samples/common/include/Audio/ |
| D | AudioCapture.hpp | 16 namespace audio namespace
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| /external/sonivox/arm-wt-22k/lib_src/ |
| D | eas_math.h | 126 #define MULT_AUDIO_COEF(audio,coef) /*lint -e704 <avoid divide for performance>*/ \ argument 144 #define MULT_AUDIO_WET_DRY_COEF(audio,coef) /*lint -e(702) <avoid divide for performance>*/ \ argument 304 #define MULT_AUDIO_DRIVE(audio,drive) /*lint -e(702) <avoid divide for performance>*/ \ argument
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| /external/webrtc/pc/ |
| D | remote_audio_source.cc | 46 void OnData(const AudioSinkInterface::Data& audio) override { in OnData() 151 void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) { in OnData()
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| /external/webrtc/audio/test/ |
| D | pc_low_bandwidth_audio_test.cc | 135 AudioConfig audio; in TEST() local 161 AudioConfig audio; in TEST() local
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| /external/tensorflow/tensorflow/examples/wav_to_spectrogram/ |
| D | wav_to_spectrogram_test.cc | 29 float audio[8] = {-1.0f, 0.0f, 1.0f, 0.0f, -1.0f, 0.0f, 1.0f, 0.0f}; in TEST() local
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| /external/webrtc/modules/audio_coding/codecs/pcm16b/ |
| D | audio_encoder_pcm16b.cc | 18 size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio, in EncodeCall()
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| /external/webrtc/modules/audio_coding/codecs/ |
| D | builtin_audio_encoder_factory_unittest.cc | 67 rtc::BufferT<int16_t> audio; in TEST_P() local 68 audio.SetData(num_samples, [](rtc::ArrayView<int16_t> audio) { in TEST_P()
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