/external/webrtc/webrtc/modules/audio_device/include/ |
D | audio_device_defines.h | 87 int sample_rate, in OnDataAvailable() argument 105 int sample_rate, in OnData() argument 118 int sample_rate, in PushCaptureData() argument 127 int sample_rate, in PullRenderData() argument 152 AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer) in AudioParameters() argument 153 : sample_rate_(sample_rate), in AudioParameters() 156 frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {} in AudioParameters() 157 void reset(int sample_rate, size_t channels, size_t frames_per_buffer) { in reset() argument 158 sample_rate_ = sample_rate; in reset() 161 frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100); in reset() [all …]
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/external/autotest/server/site_tests/brillo_PlaybackAudioTest/ |
D | brillo_PlaybackAudioTest.py | 69 def test_playback(self, fb_query, playback_cmd, sample_width, sample_rate, argument 81 sample_rate=sample_rate, 94 def test_audio(self, fb_client, playback_method, sample_rate, sample_width, argument 109 sample_rate=sample_rate, 117 self.host, num_channels, sample_rate, sample_width, 126 sample_rate=sample_rate, 157 for sample_rate in sample_rates_arr: 161 logging.info('Sample rate: %d', sample_rate) 168 sample_rate=sample_rate, 173 failed_params.append((sample_rate, sample_width,
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/external/autotest/server/site_tests/brillo_RecordingAudioTest/ |
D | brillo_RecordingAudioTest.py | 41 sample_rate, num_channels, rec_file): argument 61 (duration_secs, num_channels, sample_rate, sample_width, 67 (duration_secs, num_channels, sample_rate, rec_file)) 70 (num_channels, duration_secs, sample_rate, sample_width, 77 sample_rate, num_channels, duration_secs): argument 96 sample_rate=sample_rate, 102 sample_rate=sample_rate, 150 for sample_rate in sample_rates_arr: 156 logging.info('Sample rate: %d', sample_rate) 164 sample_rate=sample_rate, [all …]
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/external/webrtc/webrtc/common_audio/ |
D | wav_header_unittest.cc | 95 int sample_rate = 0; in TEST() local 122 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 143 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 164 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 186 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 209 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 228 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 240 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 272 int sample_rate = 0; in TEST() local 278 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() [all …]
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D | wav_header.cc | 63 int sample_rate, in CheckWavParameters() argument 70 if (num_channels == 0 || sample_rate <= 0 || bytes_per_sample == 0) in CheckWavParameters() 72 if (static_cast<uint64_t>(sample_rate) > std::numeric_limits<uint32_t>::max()) in CheckWavParameters() 79 if (static_cast<uint64_t>(sample_rate) * num_channels * bytes_per_sample > in CheckWavParameters() 138 static inline uint32_t ByteRate(size_t num_channels, int sample_rate, in ByteRate() argument 140 return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample); in ByteRate() 150 int sample_rate, in WriteWavHeader() argument 154 RTC_CHECK(CheckWavParameters(num_channels, sample_rate, format, in WriteWavHeader() 168 WriteLE32(&header.fmt.SampleRate, sample_rate); in WriteWavHeader() 169 WriteLE32(&header.fmt.ByteRate, ByteRate(num_channels, sample_rate, in WriteWavHeader() [all …]
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D | wav_file.h | 29 virtual int sample_rate() const = 0; 42 WavWriter(const std::string& filename, int sample_rate, size_t num_channels); 53 int sample_rate() const override { return sample_rate_; } in sample_rate() function 81 int sample_rate() const override { return sample_rate_; } in sample_rate() function 104 int sample_rate,
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D | wav_file.cc | 43 s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels() in FormatAsString() 45 << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s"; in FormatAsString() 101 WavWriter::WavWriter(const std::string& filename, int sample_rate, in WavWriter() argument 103 : sample_rate_(sample_rate), in WavWriter() 155 int sample_rate, in rtc_WavOpen() argument 158 new webrtc::WavWriter(filename, sample_rate, num_channels)); in rtc_WavOpen() 172 return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate(); in rtc_WavSampleRate()
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/external/tensorflow/tensorflow/python/ops/signal/ |
D | mel_ops.py | 69 def _validate_arguments(num_mel_bins, sample_rate, argument 74 if sample_rate <= 0.0: 75 raise ValueError('sample_rate must be positive. Got: %s' % sample_rate) 82 if upper_edge_hertz > sample_rate / 2: 85 % (upper_edge_hertz, sample_rate)) 93 sample_rate=8000, argument 154 _validate_arguments(num_mel_bins, sample_rate, 159 sample_rate = ops.convert_to_tensor( 160 sample_rate, dtype, name='sample_rate') 169 nyquist_hertz = sample_rate / 2.0
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/external/autotest/server/brillo/feedback/ |
D | closed_loop_audio_client.py | 151 sample_rate=_DEFAULT_SAMPLE_RATE, argument 162 self.sample_rate = sample_rate 170 (num_channels, duration_secs, sample_rate, sample_width, 192 sample_rate=self.sample_rate, 227 sample_rate=self.sample_rate, 275 sample_rate=_DEFAULT_SAMPLE_RATE, argument 291 self.sample_rate = sample_rate 304 self.sample_rate, self.sample_width, 326 sample_rate=self.sample_rate,
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/external/adhd/cras/src/server/ |
D | cras_dsp_mod_builtin.c | 18 unsigned long sample_rate) in empty_instantiate() argument 63 unsigned long sample_rate) in swap_lr_instantiate() argument 113 unsigned long sample_rate) in invert_lr_instantiate() argument 159 unsigned long sample_rate) in mix_stereo_instantiate() argument 216 unsigned long sample_rate) in dcblock_instantiate() argument 273 int sample_rate; member 280 static int eq_instantiate(struct dsp_module *module, unsigned long sample_rate) in eq_instantiate() argument 286 data->sample_rate = (int) sample_rate; in eq_instantiate() 301 float nyquist = data->sample_rate / 2; in eq_run() 346 int sample_rate; member [all …]
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D | cras_dsp.c | 32 int sample_rate; member 72 if (cras_dsp_pipeline_instantiate(pipeline, ctx->sample_rate) != 0) { in prepare_pipeline() 77 if (cras_dsp_pipeline_get_sample_rate(pipeline) != ctx->sample_rate) { in prepare_pipeline() 80 ctx->sample_rate); in prepare_pipeline() 149 struct cras_dsp_context *cras_dsp_context_new(int sample_rate, in cras_dsp_context_new() argument 156 ctx->sample_rate = sample_rate; in cras_dsp_context_new()
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/external/tensorflow/tensorflow/python/kernel_tests/ |
D | summary_v1_audio_op_test.py | 37 def _CheckProto(self, audio_summ, sample_rate, num_channels, length_frames): argument 47 }""" % (i, sample_rate, num_channels, length_frames) for i in xrange(3)) 60 sample_rate = 8000 62 "snd", const, max_outputs=3, sample_rate=sample_rate) 68 self._CheckProto(audio_summ, sample_rate, channels, num_frames)
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/external/tensorflow/tensorflow/examples/speech_commands/ |
D | freeze.py | 63 def create_inference_graph(wanted_words, sample_rate, clip_duration_ms, argument 89 len(words_list), sample_rate, clip_duration_ms, window_size_ms, 115 sample_rate, 125 sample_rate = model_settings['sample_rate'] 127 1000) / sample_rate 129 1000) / sample_rate 134 sample_rate=sample_rate, 161 FLAGS.wanted_words, FLAGS.sample_rate, FLAGS.clip_duration_ms,
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D | generate_streaming_test_wav.py | 89 len(words_list), FLAGS.sample_rate, FLAGS.clip_duration_ms, 97 output_audio_sample_count = FLAGS.sample_rate * FLAGS.test_duration_seconds 105 (background_segment_duration_ms * FLAGS.sample_rate) / 1000) 107 (FLAGS.clip_duration_ms * FLAGS.sample_rate) / 1000) 109 ((background_crossover_ms / 2) * FLAGS.sample_rate) / 1000) 129 word_stride_samples = int((word_stride_ms * FLAGS.sample_rate) / 1000) 131 (FLAGS.clip_duration_ms * FLAGS.sample_rate) / 1000) 132 word_gap_samples = int((FLAGS.word_gap_ms * FLAGS.sample_rate) / 1000) 140 output_offset_ms = (output_offset * 1000) / FLAGS.sample_rate 162 FLAGS.sample_rate)
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/external/autotest/server/brillo/ |
D | audio_utils.py | 67 def check_wav_file(filename, num_channels=None, sample_rate=None, argument 87 if sample_rate is not None and chk_file.getframerate() != sample_rate: 89 sample_rate, chk_file.getframerate()) 106 def generate_sine_file(host, num_channels, sample_rate, sample_width, argument 130 sample_width * _BITS_PER_BYTE, sample_rate, 175 sample_rate): argument 207 1.0 / sample_rate) 208 fft_freqs_rec = numpy.fft.rfftfreq(len(rec_data), 1.0 / sample_rate)
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/external/tensorflow/tensorflow/core/lib/wav/ |
D | wav_io.cc | 46 char sample_rate[4]; member 135 Status EncodeAudioAsS16LEWav(const float* audio, size_t sample_rate, in EncodeAudioAsS16LEWav() argument 150 if (sample_rate == 0 || sample_rate > kuint32max) { in EncodeAudioAsS16LEWav() 152 sample_rate); in EncodeAudioAsS16LEWav() 162 const size_t bytes_per_second = sample_rate * kBytesPerSample * num_channels; in EncodeAudioAsS16LEWav() 191 core::EncodeFixed32(format_chunk->sample_rate, sample_rate); in EncodeAudioAsS16LEWav() 214 uint32* sample_rate) { in DecodeLin16WaveAsFloatVector() argument 240 TF_RETURN_IF_ERROR(ReadValue<uint32>(wav_string, sample_rate, &offset)); in DecodeLin16WaveAsFloatVector() 262 const uint32 expected_bytes_per_second = bytes_per_sample * *sample_rate; in DecodeLin16WaveAsFloatVector() 267 " (sample_rate=", *sample_rate, ", bytes_per_sample=", bytes_per_sample, in DecodeLin16WaveAsFloatVector()
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/external/webrtc/webrtc/modules/audio_processing/beamformer/ |
D | nonlinear_beamformer_test.cc | 46 WavWriter out_file(FLAGS_o, in_file.sample_rate(), 1); in main() 54 bf.Initialize(kChunkSizeMs, in_file.sample_rate()); in main() 57 FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate()); in main() 59 FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate()); in main() 62 rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond), in main() 65 rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond), in main()
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/external/webrtc/webrtc/audio/ |
D | audio_sink.h | 32 int sample_rate, in Data() 37 sample_rate(sample_rate), in Data() 43 int sample_rate; // Sample rate in Hz. member
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/external/webrtc/webrtc/voice_engine/ |
D | voe_base_impl.h | 78 int sample_rate, 88 int sample_rate, 94 int sample_rate, 98 int sample_rate, 128 const void* audio_data, uint32_t sample_rate, size_t number_of_channels, 132 void GetPlayoutData(int sample_rate, size_t number_of_channels,
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D | voe_base_impl.cc | 114 const int16_t* audio_data, int sample_rate, in OnDataAvailable() argument 123 voe_channels, number_of_voe_channels, audio_data, sample_rate, in OnDataAvailable() 134 PushCaptureData(voe_channels[i], audio_data, 16, sample_rate, in OnDataAvailable() 143 int bits_per_sample, int sample_rate, in OnData() argument 145 PushCaptureData(voe_channel, audio_data, bits_per_sample, sample_rate, in OnData() 150 int bits_per_sample, int sample_rate, in PushCaptureData() argument 159 sample_rate, number_of_frames, number_of_channels); in PushCaptureData() 160 channel_ptr->PrepareEncodeAndSend(sample_rate); in PushCaptureData() 166 int sample_rate, in PullRenderData() argument 172 assert(number_of_frames == static_cast<size_t>(sample_rate / 100)); in PullRenderData() [all …]
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/external/python/cpython2/Lib/ |
D | sunaudio.py | 26 sample_rate = get_long_be(fp.read(4)) 35 return (data_size, encoding, sample_rate, channels, info) 41 data_size, encoding, sample_rate, channels, info = hdr 47 print 'Sample rate:', sample_rate
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/external/tensorflow/tensorflow/lite/experimental/microfrontend/python/kernel_tests/ |
D | audio_microfrontend_op_test.py | 45 sample_rate=SAMPLE_RATE, 64 sample_rate=SAMPLE_RATE, 86 sample_rate=SAMPLE_RATE, 106 sample_rate=SAMPLE_RATE, 129 sample_rate=SAMPLE_RATE, 149 sample_rate=SAMPLE_RATE,
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/external/tensorflow/tensorflow/core/kernels/ |
D | summary_audio_op.cc | 49 float sample_rate = sample_rate_attr_; in Compute() local 52 sample_rate = sample_rate_tensor.scalar<float>()(); in Compute() 54 OP_REQUIRES(c, sample_rate > 0.0f, in Compute() 73 sa->set_sample_rate(sample_rate); in Compute() 83 size_t sample_rate_truncated = lrintf(sample_rate); in Compute()
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/external/flac/libFLAC/ |
D | format.c | 200 FLAC_API FLAC__bool FLAC__format_sample_rate_is_valid(unsigned sample_rate) in FLAC__format_sample_rate_is_valid() argument 202 if(sample_rate == 0 || sample_rate > FLAC__MAX_SAMPLE_RATE) { in FLAC__format_sample_rate_is_valid() 209 FLAC_API FLAC__bool FLAC__format_blocksize_is_subset(unsigned blocksize, unsigned sample_rate) in FLAC__format_blocksize_is_subset() argument 213 else if(sample_rate <= 48000 && blocksize > 4608) in FLAC__format_blocksize_is_subset() 219 FLAC_API FLAC__bool FLAC__format_sample_rate_is_subset(unsigned sample_rate) in FLAC__format_sample_rate_is_subset() argument 222 !FLAC__format_sample_rate_is_valid(sample_rate) || in FLAC__format_sample_rate_is_subset() 224 sample_rate >= (1u << 16) && in FLAC__format_sample_rate_is_subset() 225 !(sample_rate % 1000 == 0 || sample_rate % 10 == 0) in FLAC__format_sample_rate_is_subset()
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/external/webrtc/webrtc/modules/audio_device/android/ |
D | audio_manager.cc | 177 jint sample_rate, in CacheAudioParameters() argument 189 env, sample_rate, channels, hardware_aec, hardware_agc, hardware_ns, in CacheAudioParameters() 194 jint sample_rate, in OnCacheAudioParameters() argument 207 ALOGD("sample_rate: %d", sample_rate); in OnCacheAudioParameters() 217 playout_parameters_.reset(sample_rate, static_cast<size_t>(channels), in OnCacheAudioParameters() 219 record_parameters_.reset(sample_rate, static_cast<size_t>(channels), in OnCacheAudioParameters()
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