| /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/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/webrtc/modules/audio_coding/codecs/g711/ |
| D | audio_encoder_pcm.cc | 82 rtc::ArrayView<const int16_t> audio, in EncodeInternal()
110 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, in EncodeCall()
123 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, in EncodeCall()
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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
71 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f}; in TEST() local
82 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f}; in TEST() local
89 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f}; in TEST() local
130 float audio[] = {0.0f, 1.0f, 0.0f, -1.0f}; in TEST() local
159 float audio[] = {0.0f, 1.0f, 0.0f, -1.0f}; in TEST() local
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| /external/webrtc/webrtc/modules/audio_processing/agc/ |
| D | agc.cc | 42 float Agc::AnalyzePreproc(const int16_t* audio, size_t length) { in AnalyzePreproc()
52 int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) { in Process()
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| /external/autotest/client/site_tests/audio_AudioCorruption/ |
| D | audio_AudioCorruption.py | 17 def run_once(self, audio): argument
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| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| D | output_wav_file.h | 30 bool WriteArray(const int16_t* audio, size_t num_samples) override { in WriteArray()
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| D | output_audio_file.h | 37 bool WriteArray(const int16_t* audio, size_t num_samples) override { in WriteArray()
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| D | audio_checksum.h | 29 bool WriteArray(const int16_t* audio, size_t num_samples) override { in WriteArray()
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| D | audio_sink.h | 50 bool WriteArray(const int16_t* audio, size_t num_samples) override { in WriteArray()
|
| /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/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/talk/app/webrtc/ |
| D | remoteaudiosource.cc | 63 void OnData(const AudioSinkInterface::Data& audio) override { in OnData()
154 void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) { in OnData()
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| /external/tensorflow/tensorflow/core/kernels/ |
| D | encode_wav_op.cc | 35 const Tensor& audio = context->input(0); in Compute() local
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| D | encode_wav_op_test.cc | 63 const Tensor& audio = outputs[0]; in TEST() local
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_utility.h | 36 bool audio; member
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| /external/webrtc/webrtc/modules/audio_processing/vad/ |
| D | voice_activity_detector.cc | 37 void VoiceActivityDetector::ProcessChunk(const int16_t* audio, in ProcessChunk()
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| /external/tensorflow/tensorflow/contrib/ffmpeg/ |
| D | ffmpeg_ops.py | 75 def encode_audio(audio, file_format=None, samples_per_second=None): argument
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| /external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
| D | audio_encoder_pcm16b.cc | 19 size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio, in EncodeCall()
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| /external/webrtc/talk/media/base/ |
| D | audioframe.h | 41 AudioFrame(int16_t* audio, size_t audio_length, int sample_freq, bool stereo) in AudioFrame()
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| /external/webrtc/webrtc/modules/audio_processing/ |
| D | noise_suppression_impl.cc | 70 void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { in AnalyzeCaptureAudio()
87 void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) { in ProcessCaptureAudio()
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| D | level_estimator_impl.cc | 31 void LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) { in ProcessStream()
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| /external/webrtc/webrtc/common_audio/vad/ |
| D | vad.cc | 28 Activity VoiceActivity(const int16_t* audio, in VoiceActivity()
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| /external/webrtc/webrtc/voice_engine/ |
| D | transmit_mixer_unittest.cc | 23 int16_t audio[], size_t samples_per_channel, in Process()
|