| /external/webrtc/webrtc/audio/ |
| audio_sink.h | 32 int sample_rate,
37 sample_rate(sample_rate),
43 int sample_rate; // Sample rate in Hz. member in struct:webrtc::AudioSinkInterface::Data
|
| /external/webrtc/webrtc/modules/audio_device/include/ |
| audio_device_defines.h | 87 int sample_rate,
105 int sample_rate,
118 int sample_rate,
127 int sample_rate,
152 AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer)
153 : sample_rate_(sample_rate),
156 frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
157 void reset(int sample_rate, size_t channels, size_t frames_per_buffer) {
158 sample_rate_ = sample_rate;
161 frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100)
171 int sample_rate() const { return sample_rate_; } function in class:webrtc::AudioParameters
[all...] |
| /external/webrtc/webrtc/common_audio/ |
| wav_header.h | 36 int sample_rate,
47 int sample_rate,
57 int* sample_rate,
|
| /device/google/bonito/sdm710/kernel-headers/linux/ |
| msm_audio_g711.h | 7 uint32_t sample_rate; member in struct:msm_audio_g711_enc_config
|
| msm_audio_g711_dec.h | 7 uint32_t sample_rate; member in struct:msm_audio_g711_dec_config
|
| /external/flac/include/share/grabbag/ |
| seektable.h | 33 FLAC__bool grabbag__seektable_convert_specification_to_template(const char *spec, FLAC__bool only_explicit_placeholders, FLAC__uint64 total_samples_to_encode, unsigned sample_rate, FLAC__StreamMetadata *seektable_template, FLAC__bool *spec_has_real_points);
|
| cuesheet.h | 35 FLAC__StreamMetadata *grabbag__cuesheet_parse(FILE *file, const char **error_message, unsigned *last_line_read, unsigned sample_rate, FLAC__bool is_cdda, FLAC__uint64 lead_out_offset);
|
| /external/tensorflow/tensorflow/lite/experimental/micro/examples/micro_speech/micro_features/ |
| filterbank_util.h | 39 struct FilterbankState* state, int sample_rate,
|
| window_util.cc | 33 struct WindowState* state, int sample_rate) {
34 state->size = config->size_ms * sample_rate / 1000;
35 state->step = config->step_size_ms * sample_rate / 1000;
|
| frontend_util.h | 42 struct FrontendState* state, int sample_rate);
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| window_util.h | 35 struct WindowState* state, int sample_rate);
|
| /external/webrtc/talk/media/base/ |
| audiorenderer.h | 43 int sample_rate,
|
| /external/webrtc/webrtc/modules/audio_processing/logging/ |
| aec_logging_file_handling.h | 29 int sample_rate,
|
| aec_logging_file_handling.cc | 25 int sample_rate,
28 if (rtc_WavSampleRate(*wav_file) == sample_rate)
41 *wav_file = rtc_WavOpen(filename, sample_rate, 1);
|
| /external/tensorflow/tensorflow/lite/experimental/microfrontend/lib/ |
| frontend_memmap_generator.c | 31 int sample_rate = 16000; local
33 if (!FrontendPopulateState(&frontend_config, &frontend_state, sample_rate)) {
|
| filterbank_util.h | 40 struct FilterbankState* state, int sample_rate,
|
| frontend_util.h | 43 struct FrontendState* state, int sample_rate);
|
| /external/webrtc/webrtc/modules/audio_device/android/ |
| opensles_common.cc | 21 SLDataFormat_PCM CreatePcmConfiguration(int sample_rate) {
29 configuration.samplesPerSec = sample_rate * 1000;
|
| /external/autotest/server/site_tests/brillo_PlaybackAudioTest/ |
| brillo_PlaybackAudioTest.py | 69 def test_playback(self, fb_query, playback_cmd, sample_width, sample_rate,
77 @param sample_rate: Sample rate to test playback at.
81 sample_rate=sample_rate,
94 def test_audio(self, fb_client, playback_method, sample_rate, sample_width,
101 @param sample_rate: Sample rate to test playback at.
109 sample_rate=sample_rate,
117 self.host, num_channels, sample_rate, sample_width,
126 sample_rate=sample_rate
[all...] |
| /external/autotest/server/site_tests/brillo_RecordingAudioTest/ |
| brillo_RecordingAudioTest.py | 41 sample_rate, num_channels, rec_file):
48 @param sample_rate: Recording sample rate in hertz.
59 '--duration_secs=%d --num_channels=%d --sample_rate=%d '
61 (duration_secs, num_channels, sample_rate, sample_width,
65 '--duration_secs=%d --num_channels=%d --sample_rate=%d '
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):
84 @param sample_rate: Recording sample rate in hertz.
96 sample_rate=sample_rate
[all...] |
| /external/tensorflow/tensorflow/python/kernel_tests/ |
| summary_v1_audio_op_test.py | 37 def _CheckProto(self, audio_summ, sample_rate, num_channels, length_frames):
45 audio { content_type: "audio/wav" sample_rate: %d
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)
|
| /external/webrtc/webrtc/modules/audio_processing/beamformer/ |
| nonlinear_beamformer_test.cc | 46 WavWriter out_file(FLAGS_o, in_file.sample_rate(), 1);
54 bf.Initialize(kChunkSizeMs, in_file.sample_rate());
57 FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
59 FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
62 rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond),
65 rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond),
|
| covariance_matrix_generator.h | 36 int sample_rate,
45 int sample_rate,
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| /external/flac/libFLAC/include/protected/ |
| stream_decoder.h | 47 unsigned sample_rate; /* in Hz */ member in struct:FLAC__StreamDecoderProtected
|
| /external/python/cpython2/Lib/ |
| 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
|
