| /external/tensorflow/tensorflow/lite/experimental/microfrontend/lib/ |
| window_util.h | 36 struct WindowState* state, int sample_rate);
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| frontend_util.c | 31 struct FrontendState* state, int sample_rate) {
34 if (!WindowPopulateState(&config->window, &state->window, sample_rate)) {
46 sample_rate, state->fft.fft_size / 2 + 1)) {
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| window_util.c | 28 struct WindowState* state, int sample_rate) {
29 state->size = config->size_ms * sample_rate / 1000;
30 state->step = config->step_size_ms * sample_rate / 1000;
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| /external/webrtc/webrtc/common_audio/resampler/ |
| sinusoidal_linear_chirp_source.h | 29 SinusoidalLinearChirpSource(int sample_rate, size_t samples,
|
| sinusoidal_linear_chirp_source.cc | 20 SinusoidalLinearChirpSource::SinusoidalLinearChirpSource(int sample_rate,
24 : sample_rate_(sample_rate),
|
| /external/tensorflow/tensorflow/python/ops/signal/ |
| mel_ops.py | 69 def _validate_arguments(num_mel_bins, sample_rate,
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:
84 'frequency (sample_rate / 2). Got: %s for sample_rate: %s'
85 % (upper_edge_hertz, sample_rate))
93 sample_rate=8000,
102 `[0, sample_rate / 2]` into `num_mel_bins` frequency information fro
[all...] |
| /external/webrtc/webrtc/common_audio/ |
| 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_; }
81 int sample_rate() const override { return sample_rate_; }
104 int sample_rate,
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| wav_header.cc | 63 int sample_rate,
67 // num_channels, sample_rate, and bytes_per_sample must be positive, must fit
70 if (num_channels == 0 || sample_rate <= 0 || bytes_per_sample == 0)
72 if (static_cast<uint64_t>(sample_rate) > std::numeric_limits<uint32_t>::max())
79 if (static_cast<uint64_t>(sample_rate) * num_channels * bytes_per_sample >
138 static inline uint32_t ByteRate(size_t num_channels, int sample_rate,
140 return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample);
150 int sample_rate,
154 RTC_CHECK(CheckWavParameters(num_channels, sample_rate, format,
168 WriteLE32(&header.fmt.SampleRate, sample_rate);
[all...] |
| wav_header_unittest.cc | 95 int sample_rate = 0; local
122 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
143 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
164 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
186 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
209 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
228 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
240 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
272 int sample_rate = 0; local
278 ReadWavHeader(&r, &num_channels, &sample_rate, &format
308 int sample_rate = 0; local
[all...] |
| /frameworks/av/services/audioflinger/ |
| SpdifStreamOut.cpp | 55 mApplicationSampleRate = config->sample_rate;
74 customConfig.sample_rate = config->sample_rate * mRateMultiplier;
84 config->sample_rate,
89 customConfig.sample_rate,
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| /external/webrtc/webrtc/voice_engine/ |
| 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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| /external/adhd/cras/src/dsp/ |
| drc.h | 111 float sample_rate; member in struct:drc
154 struct drc *drc_new(float sample_rate);
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| drc_kernel.h | 21 float sample_rate; member in struct:drc_kernel
82 void dk_init(struct drc_kernel *dk, float sample_rate);
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| /external/libxaac/decoder/drc_src/ |
| impd_drc_peak_limiter.h | 33 UWORD32 sample_rate; member in struct:ia_drc_peak_limiter_struct
48 UWORD32 sample_rate, FLOAT32 *buffer);
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| impd_drc_peak_limiter_struct.h | 33 UWORD32 sample_rate; member in struct:ia_drc_peak_limiter_struct
|
| /frameworks/av/media/libeffects/loudness/common/core/ |
| basic_types.h | 104 int sample_rate(void) const;
105 void set_sample_rate(int sample_rate);
|
| /device/google/cuttlefish_common/guest/hals/audio/legacy/ |
| vsoc_audio_output_stream.h | 62 int SetSampleRate(uint32_t sample_rate) {
63 if (sample_rate != message_header_.frame_rate) {
64 message_header_.frame_rate = sample_rate;
71 sample_rate, GetBufferSize() / frame_size_));
|
| /external/tensorflow/tensorflow/core/summary/ |
| summary_converter.h | 33 int max_outputs, float sample_rate,
|
| /external/tensorflow/tensorflow/lite/experimental/micro/examples/micro_speech/micro_features/ |
| frontend_util.cc | 31 struct FrontendState* state, int sample_rate) {
35 sample_rate)) {
47 &state->filterbank, sample_rate,
|
| /external/webrtc/webrtc/modules/audio_processing/logging/ |
| aec_logging.h | 28 sample_rate, wav_file) \
30 WebRtcAec_ReopenWav(name, instance_index, process_rate, sample_rate, \
78 sample_rate) \
|
| /external/tensorflow/tensorflow/core/kernels/ |
| summary_audio_op.cc | 36 context->GetAttr("sample_rate", &sample_rate_attr_).ok();
49 float sample_rate = sample_rate_attr_; variable
52 sample_rate = sample_rate_tensor.scalar<float>()();
54 OP_REQUIRES(c, sample_rate > 0.0f,
55 errors::InvalidArgument("sample_rate must be > 0"));
73 sa->set_sample_rate(sample_rate);
83 size_t sample_rate_truncated = lrintf(sample_rate);
107 // Deprecated -- this op is registered with sample_rate as an attribute for
|
| /external/adhd/cras/src/server/ |
| cras_dsp.c | 32 int sample_rate; member in struct:cras_dsp_context
72 if (cras_dsp_pipeline_instantiate(pipeline, ctx->sample_rate) != 0) {
77 if (cras_dsp_pipeline_get_sample_rate(pipeline) != ctx->sample_rate) {
80 ctx->sample_rate);
149 struct cras_dsp_context *cras_dsp_context_new(int sample_rate,
156 ctx->sample_rate = sample_rate;
|
| /external/autotest/server/brillo/feedback/ |
| closed_loop_audio_client.py | 151 sample_rate=_DEFAULT_SAMPLE_RATE,
157 @sample_rate: Sample rate to record at.
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
[all...] |
| /external/tensorflow/tensorflow/examples/speech_commands/ |
| 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)
[all...] |
| /external/tensorflow/tensorflow/lite/experimental/microfrontend/python/kernel_tests/ |
| audio_microfrontend_op_test.py | 26 SAMPLE_RATE = 1000
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
[all...] |
