| /external/libxaac/encoder/ |
| D | ixheaace_loudness_measurement.c | 33 {0, -1.84460946989011, 0.85584332293064}, /* 96000Hz sample_rate*/
34 {0, -1.83091998796233, 0.84414226108785}, /* 88200Hz sample_rate*/
35 {0, -1.76738637827624, 0.79175893605869}, /* 64000Hz sample_rate*/
36 {0, -1.69065929318241, 0.73248077421585}, /* 48000Hz sample_rate*/
37 {0, -1.66365511325602, 0.71259542807323}, /* 44100Hz sample_rate*/
38 {0, -1.53904509625064, 0.62696685598156}, /* 32000Hz sample_rate*/
39 {0, -1.39023460519282, 0.53683848126040}, /* 24000Hz sample_rate*/
40 {0, -1.33830533606613, 0.50824455891360}, /* 22050Hz sample_rate*/
41 {0, -1.10153376910699, 0.39491236874986}, /* 16000Hz sample_rate*/
42 {0, -0.82398044060334, 0.29429059828526}, /* 12000Hz sample_rate*/
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| D | ixheaace_tns_init.c | 92 ia_enhaacplus_enc_init_tns_configuration(WORD32 bit_rate, WORD32 sample_rate, WORD32 channels, in ia_enhaacplus_enc_init_tns_configuration() argument
114 (const WORD16)(pstr_tns_config->max_order + 1), sample_rate, in ia_enhaacplus_enc_init_tns_configuration()
120 sample_rate, LONG_WINDOW, &pstr_tns_config->tns_max_sfb, pstr_tns_tab->tns_max_bands_table, in ia_enhaacplus_enc_init_tns_configuration()
135 pstr_tns_config->tns_start_freq, sample_rate, pstr_psy_config->sfb_cnt, in ia_enhaacplus_enc_init_tns_configuration()
139 TEMPORAL_NOISE_SHAPING_MODIFY_BEGIN, sample_rate, pstr_psy_config->sfb_cnt, in ia_enhaacplus_enc_init_tns_configuration()
143 RATIO_PATCH_LOWER_BORDER, sample_rate, pstr_psy_config->sfb_cnt, in ia_enhaacplus_enc_init_tns_configuration()
149 pstr_tns_config->conf_tab.lpc_stop_freq, sample_rate, pstr_psy_config->sfb_cnt, in ia_enhaacplus_enc_init_tns_configuration()
157 pstr_tns_config->conf_tab.lpc_start_freq, sample_rate, pstr_psy_config->sfb_cnt, in ia_enhaacplus_enc_init_tns_configuration()
167 WORD32 bit_rate, WORD32 sample_rate, WORD32 channels, in ia_enhaacplus_enc_init_tns_configuration_short() argument
184 sample_rate, SHORT_WINDOW, pstr_tns_config->conf_tab.tns_time_resolution); in ia_enhaacplus_enc_init_tns_configuration_short()
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| D | ixheaace_fd_qc_util.c | 56 VOID iusace_qc_init(ia_qc_data_struct *pstr_qc_data, const WORD32 max_bits, WORD32 sample_rate, in iusace_qc_init() argument
67 pstr_qc_data->padding = sample_rate; in iusace_qc_init()
72 mean_pe = 10.0f * ccfl * bw_limit / (sample_rate / 2.0f); in iusace_qc_init()
80 static WORD32 iusace_calc_frame_len(WORD32 bit_rate, WORD32 sample_rate, WORD32 mode, in iusace_calc_frame_len() argument
87 result %= sample_rate; in iusace_calc_frame_len()
90 result /= sample_rate; in iusace_calc_frame_len()
97 static WORD32 iusace_get_frame_padding(WORD32 bit_rate, WORD32 sample_rate, WORD32 *padding, in iusace_get_frame_padding() argument
102 difference = iusace_calc_frame_len(bit_rate, sample_rate, FRAME_LEN_BYTES_MODULO, ccfl); in iusace_get_frame_padding()
108 *padding += sample_rate; in iusace_get_frame_padding()
114 VOID iusace_adj_bitrate(ia_qc_data_struct *pstr_qc_data, WORD32 bit_rate, WORD32 sample_rate, in iusace_adj_bitrate() argument
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| D | iusace_psy_utils.c | 110 static VOID iusace_sfb_init(WORD32 sample_rate, WORD32 block_type, WORD32 *ptr_sfb_offset, in iusace_sfb_init() argument
115 WORD32 sampling_rate_mapped = iusace_map_sample_rate(sample_rate); in iusace_sfb_init()
241 WORD32 sample_rate) { in iusace_calc_bark_line_value() argument
244 center_freq = (FLOAT32)fft_line * ((FLOAT32)sample_rate * (FLOAT32)0.5f) / (FLOAT32)num_lines; in iusace_calc_bark_line_value()
253 WORD32 sample_rate, FLOAT32 *ptr_b_value) { in iusace_bark_values_init() argument
259 b_val1 = iusace_calc_bark_line_value(num_lines, ptr_sfb_offset[i + 1], sample_rate); in iusace_bark_values_init()
334 static VOID iusace_min_snr_init(const WORD32 bit_rate, const WORD32 sample_rate, in iusace_min_snr_init() argument
352 iusace_bits_to_pe((FLOAT32)bit_rate / (FLOAT32)sample_rate * (FLOAT32)num_lines); in iusace_min_snr_init()
373 VOID iusace_psy_long_config_init(WORD32 bit_rate, WORD32 sample_rate, WORD32 band_width, in iusace_psy_long_config_init() argument
378 iusace_sfb_init(sample_rate, ONLY_LONG_SEQUENCE, pstr_psy_config->sfb_offset, in iusace_psy_long_config_init()
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| D | ixheaace_psy_configuration.c | 57 WORD32 sample_rate, WORD32 aot, in ia_enhaacplus_enc_init_sfb_table() argument
66 if (pstr_psycho_tab->sfb_info_tab[i].sample_rate == sample_rate) { in ia_enhaacplus_enc_init_sfb_table()
133 static FLOAT32 iaace_calc_bark_line_value(WORD32 num_lines, WORD32 fft_line, WORD32 sample_rate) { in iaace_calc_bark_line_value() argument
135 center_freq = (FLOAT32)fft_line * ((FLOAT32)sample_rate * (FLOAT32)0.5f) / (FLOAT32)num_lines; in iaace_calc_bark_line_value()
216 WORD32 sample_rate, FLOAT32 *ptr_b_value, WORD32 aot) { in iaace_bark_values_init() argument
222 b_val1 = iaace_calc_bark_line_value(num_lines, ptr_sfb_offset[i + 1], sample_rate); in iaace_bark_values_init()
235 static VOID iaace_min_snr_init(const WORD32 bit_rate, const WORD32 sample_rate, in iaace_min_snr_init() argument
247 pe_per_window = iaace_bits_to_pe((FLOAT32)bit_rate / (FLOAT32)sample_rate * (FLOAT32)num_lines); in iaace_min_snr_init()
267 WORD32 bit_rate, WORD32 sample_rate, WORD32 bandwidth, WORD32 aot, in ia_enhaacplus_enc_init_psy_configuration() argument
273 error = ia_enhaacplus_enc_init_sfb_table(pstr_aac_tables->pstr_psycho_tab, sample_rate, aot, in ia_enhaacplus_enc_init_psy_configuration()
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| D | ixheaace_write_adts_adif.c | 104 static WORD16 ia_enhaacplus_enc_get_sample_rate_index(WORD32 sample_rate) { in ia_enhaacplus_enc_get_sample_rate_index() argument
105 if (92017 <= sample_rate) { in ia_enhaacplus_enc_get_sample_rate_index()
108 if (75132 <= sample_rate) { in ia_enhaacplus_enc_get_sample_rate_index()
111 if (55426 <= sample_rate) { in ia_enhaacplus_enc_get_sample_rate_index()
114 if (46009 <= sample_rate) { in ia_enhaacplus_enc_get_sample_rate_index()
117 if (37566 <= sample_rate) { in ia_enhaacplus_enc_get_sample_rate_index()
120 if (27713 <= sample_rate) { in ia_enhaacplus_enc_get_sample_rate_index()
123 if (23004 <= sample_rate) { in ia_enhaacplus_enc_get_sample_rate_index()
126 if (18783 <= sample_rate) { in ia_enhaacplus_enc_get_sample_rate_index()
129 if (13856 <= sample_rate) { in ia_enhaacplus_enc_get_sample_rate_index()
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| /external/tensorflow/tensorflow/python/ops/signal/ |
| D | mel_ops.py | 67 def _validate_arguments(num_mel_bins, sample_rate, argument
78 if not isinstance(sample_rate, ops.Tensor):
79 if sample_rate <= 0.0:
80 raise ValueError('sample_rate must be positive. Got: %s' % sample_rate)
81 if upper_edge_hertz > sample_rate / 2:
83 'frequency (sample_rate / 2). Got %s for sample_rate: %s'
84 % (upper_edge_hertz, sample_rate))
93 sample_rate=8000, argument
102 `[0, sample_rate / 2]` into `num_mel_bins` frequency information from
135 sample_rate: An integer or float `Tensor`. Samples per second of the input
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| /external/webrtc/modules/audio_device/include/ |
| D | audio_device_defines.h | 81 int sample_rate,
106 AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer) in AudioParameters() argument
107 : sample_rate_(sample_rate), in AudioParameters()
110 frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {} in AudioParameters()
111 void reset(int sample_rate, size_t channels, size_t frames_per_buffer) { in reset() argument
112 sample_rate_ = sample_rate; in reset()
115 frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100); in reset()
118 void reset(int sample_rate, size_t channels, double buffer_duration) { in reset() argument
119 reset(sample_rate, channels, in reset()
120 static_cast<size_t>(sample_rate * buffer_duration + 0.5)); in reset()
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| /external/tensorflow/tensorflow/lite/experimental/microfrontend/python/kernel_tests/ |
| D | audio_microfrontend_op_test.py | 22 SAMPLE_RATE = 1000 variable
44 sample_rate=SAMPLE_RATE,
62 sample_rate=SAMPLE_RATE,
83 sample_rate=SAMPLE_RATE,
103 sample_rate=SAMPLE_RATE,
125 sample_rate=SAMPLE_RATE,
145 sample_rate=SAMPLE_RATE,
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| /external/webrtc/common_audio/ |
| D | wav_header.cc | 151 int sample_rate, in ByteRate() argument
153 return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample); in ByteRate()
200 int sample_rate, in WritePcmWavHeader() argument
218 header.fmt.SampleRate = sample_rate; in WritePcmWavHeader()
219 header.fmt.ByteRate = ByteRate(num_channels, sample_rate, bytes_per_sample); in WritePcmWavHeader()
231 int sample_rate, in WriteIeeeFloatWavHeader() argument
250 header.fmt.SampleRate = sample_rate; in WriteIeeeFloatWavHeader()
251 header.fmt.ByteRate = ByteRate(num_channels, sample_rate, bytes_per_sample); in WriteIeeeFloatWavHeader()
282 int sample_rate, in CheckWavParameters() argument
286 // num_channels, sample_rate, and bytes_per_sample must be positive, must fit in CheckWavParameters()
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| D | wav_header_unittest.cc | 103 int sample_rate = 0; in TEST() local
134 EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST()
159 EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST()
184 EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST()
210 EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST()
237 EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST()
260 EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST()
276 EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST()
314 int sample_rate = 0; in TEST() local
321 EXPECT_TRUE(ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST()
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| /external/autotest/server/brillo/feedback/ |
| D | closed_loop_audio_client.py | 151 sample_rate=_DEFAULT_SAMPLE_RATE, argument
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, argument
285 @param sample_rate: Recording sample rate in hertz.
291 self.sample_rate = sample_rate
304 self.sample_rate, self.sample_width,
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| /external/webrtc/audio/ |
| D | audio_transport_impl.cc | 113 const uint32_t sample_rate, in RecordedDataIsAvailable() argument
121 sample_rate, audio_delay_milliseconds, clock_drift, volume, key_pressed, in RecordedDataIsAvailable()
132 const uint32_t sample_rate, in RecordedDataIsAvailable() argument
144 RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz); in RecordedDataIsAvailable()
146 RTC_DCHECK_EQ(number_of_frames * 100, sample_rate); in RecordedDataIsAvailable()
161 InitializeCaptureFrame(sample_rate, send_sample_rate_hz, number_of_channels, in RecordedDataIsAvailable()
164 number_of_frames, number_of_channels, sample_rate, in RecordedDataIsAvailable()
241 int sample_rate, in PullRenderData() argument
247 TRACE_EVENT2("webrtc", "AudioTransportImpl::PullRenderData", "sample_rate", in PullRenderData()
248 sample_rate, "number_of_frames", number_of_frames); in PullRenderData()
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| /external/tensorflow/tensorflow/examples/speech_commands/ |
| D | freeze.py | 22 --sample_rate=16000 --dct_coefficient_count=40 --window_size_ms=20 \
29 `sample_rate` and other command line variables here as you did for the training
59 def create_inference_graph(wanted_words, sample_rate, clip_duration_ms, argument
69 sample_rate: How many samples per second are in the input audio files.
88 len(words_list), sample_rate, clip_duration_ms, window_size_ms,
115 sample_rate,
124 sample_rate = model_settings['sample_rate']
126 1000) / sample_rate
128 1000) / sample_rate
133 sample_rate=sample_rate,
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| D | generate_streaming_test_wav.py | 85 len(words_list), FLAGS.sample_rate, FLAGS.clip_duration_ms,
93 output_audio_sample_count = FLAGS.sample_rate * FLAGS.test_duration_seconds
101 (background_segment_duration_ms * FLAGS.sample_rate) / 1000)
103 (FLAGS.clip_duration_ms * FLAGS.sample_rate) / 1000)
105 ((background_crossover_ms / 2) * FLAGS.sample_rate) / 1000)
125 word_stride_samples = int((word_stride_ms * FLAGS.sample_rate) / 1000)
127 (FLAGS.clip_duration_ms * FLAGS.sample_rate) / 1000)
128 word_gap_samples = int((FLAGS.word_gap_ms * FLAGS.sample_rate) / 1000)
136 output_offset_ms = (output_offset * 1000) / FLAGS.sample_rate
158 FLAGS.sample_rate)
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| /external/tensorflow/tensorflow/core/lib/wav/ |
| D | wav_io.cc | 46 char sample_rate[4]; member
141 Status EncodeAudioAsS16LEWav(const float* audio, size_t sample_rate, in EncodeAudioAsS16LEWav() argument
157 if (sample_rate == 0 || sample_rate > kuint32max) { in EncodeAudioAsS16LEWav()
158 return errors::InvalidArgument("sample_rate must be in (0, 2^32), got: ", in EncodeAudioAsS16LEWav()
159 sample_rate); in EncodeAudioAsS16LEWav()
166 const size_t bytes_per_second = sample_rate * kBytesPerSample * num_channels; in EncodeAudioAsS16LEWav()
195 core::EncodeFixed32(format_chunk->sample_rate, sample_rate); in EncodeAudioAsS16LEWav()
216 size_t sample_rate,
221 size_t sample_rate,
229 uint32* sample_rate) { in DecodeLin16WaveAsFloatVector() argument
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| /external/tensorflow/tensorflow/core/kernels/ |
| D | summary_audio_op.cc | 36 context->GetAttr("sample_rate", &sample_rate_attr_).ok(); in SummaryAudioOp()
49 float sample_rate = sample_rate_attr_; in Compute() local
56 "sample_rate must be rank-0 or contain a single value")); in Compute()
57 sample_rate = sample_rate_tensor.scalar<float>()(); in Compute()
59 OP_REQUIRES(c, sample_rate > 0.0f, in Compute()
60 errors::InvalidArgument("sample_rate must be > 0")); in Compute()
78 sa->set_sample_rate(sample_rate); in Compute()
88 size_t sample_rate_truncated = lrintf(sample_rate); in Compute()
111 // Deprecated -- this op is registered with sample_rate as an attribute for
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| D | summary_audio_op_test.cc | 101 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 2 in TEST_F()
104 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 2 in TEST_F()
107 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 2 in TEST_F()
135 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 1 in TEST_F()
138 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 1 in TEST_F()
141 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 1 in TEST_F()
167 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 1 in TEST_F()
170 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 1 in TEST_F()
173 audio { content_type: "audio/wav" sample_rate: 44100 num_channels: 1 in TEST_F()
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| /external/tensorflow/tensorflow/python/kernel_tests/summary_ops/ |
| D | summary_v1_audio_op_test.py | 32 def _CheckProto(self, audio_summ, sample_rate, num_channels, length_frames): argument
40 audio { content_type: "audio/wav" sample_rate: %d
42 }""" % (i, sample_rate, num_channels, length_frames) for i in range(3))
55 sample_rate = 8000
57 "snd", const, max_outputs=3, sample_rate=sample_rate)
63 self._CheckProto(audio_summ, sample_rate, channels, num_frames)
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| /external/autotest/server/brillo/ |
| D | audio_utils.py | 74 def check_wav_file(filename, num_channels=None, sample_rate=None, argument
80 @param sample_rate: Sample rate to expect (None to not check).
94 if sample_rate is not None and chk_file.getframerate() != sample_rate:
96 sample_rate, chk_file.getframerate())
113 def generate_sine_file(host, num_channels, sample_rate, sample_width, argument
120 @param sample_rate: Sample rate to use for sine wave generation.
137 sample_width * _BITS_PER_BYTE, sample_rate,
182 sample_rate): argument
200 @param sample_rate: Sample rate of the files.
214 1.0 / sample_rate)
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| /external/webrtc/modules/audio_mixer/ |
| D | frame_combiner.cc | 45 int sample_rate, in SetAudioFrameFields() argument
49 (sample_rate * webrtc::AudioMixerImpl::kFrameDurationInMs) / 1000); in SetAudioFrameFields()
56 0, nullptr, samples_per_channel, sample_rate, AudioFrame::kUndefined, in SetAudioFrameFields()
120 const size_t sample_rate = mixing_buffer_view.samples_per_channel() * 1000 / in RunLimiter() local
123 limiter->SetSampleRate(sample_rate); in RunLimiter()
162 int sample_rate, in Combine() argument
167 SetAudioFrameFields(mix_list, number_of_channels, sample_rate, in Combine()
171 (sample_rate * webrtc::AudioMixerImpl::kFrameDurationInMs) / 1000); in Combine()
175 RTC_DCHECK_EQ(sample_rate, frame->sample_rate_hz_); in Combine()
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| /external/openscreen/cast/standalone_sender/ |
| D | streaming_opus_encoder.cc | 37 samples_per_cast_frame_(sample_rate() / cast_frames_per_second), in StreamingOpusEncoder()
46 OSP_CHECK_EQ(sample_rate() % cast_frames_per_second, 0); in StreamingOpusEncoder()
52 encoder(), sample_rate(), num_channels_, OPUS_APPLICATION_AUDIO); in StreamingOpusEncoder()
148 sample_rate()); in UpdateCodecDelay()
169 reference_time - start_time_, sample_rate()); in ResolveTimestampsAndMaybeSkip()
176 << rtp_advancement.ToDuration<microseconds>(sample_rate()) in ResolveTimestampsAndMaybeSkip()
185 .ToDuration<Clock::duration>(sample_rate()); in ResolveTimestampsAndMaybeSkip()
192 RtpTimeDelta::FromTicks(1).ToDuration<Clock::duration>(sample_rate()); in ResolveTimestampsAndMaybeSkip()
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| /external/flac/src/share/grabbag/ |
| D | cuesheet.c | 82 static FLAC__int64 local__parse_msf_(const char *s, uint32_t sample_rate) in local__parse_msf_() argument
87 if(sample_rate == 0) in local__parse_msf_()
106 if(field >= INT64_MAX / (60 * sample_rate)) in local__parse_msf_()
108 ret = field * 60 * sample_rate; in local__parse_msf_()
131 ret += field * sample_rate; in local__parse_msf_()
158 ret += field * (sample_rate / 75); in local__parse_msf_()
170 static FLAC__int64 local__parse_ms_(const char *s, uint32_t sample_rate) in local__parse_ms_() argument
176 if(sample_rate == 0) in local__parse_ms_()
195 if(field >= INT64_MAX / (60 * sample_rate)) in local__parse_ms_()
197 ret = field * 60 * sample_rate; in local__parse_ms_()
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| /external/flac/src/test_grabbag/cuesheet/ |
| D | main.c | 32 static int do_cuesheet(const char *infilename, uint32_t sample_rate, FLAC__bool is_cdda, FLAC__uint… in do_cuesheet() argument
52 …if(0 != (cuesheet = grabbag__cuesheet_parse(fin, &error_message, &last_line_read, sample_rate, is_… in do_cuesheet()
88 …if(0 != (cuesheet = grabbag__cuesheet_parse(fin, &error_message, &last_line_read, sample_rate, is_… in do_cuesheet()
119 uint32_t sample_rate = 48000; in main() local
121 …const char *usage = "usage: test_cuesheet cuesheet_file lead_out_offset [ [ sample_rate ] cdda ]\n… in main()
135 sample_rate = (uint32_t)atoi(argv[3]); in main()
146 return do_cuesheet(argv[1], sample_rate, is_cdda, lead_out_offset); in main()
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| /external/flac/src/metaflac/ |
| D | operations_shorthand_cuesheet.c | 33 …ename, FLAC__bool *needs_write, FLAC__uint64 lead_out_offset, unsigned sample_rate, FLAC__bool is_…
43 unsigned sample_rate = 0; in do_shorthand_operation__cuesheet() local
59 sample_rate = block->data.stream_info.sample_rate; in do_shorthand_operation__cuesheet()
60 …m_info.channels == 2) && (block->data.stream_info.bits_per_sample == 16) && (sample_rate == 44100); in do_shorthand_operation__cuesheet()
72 if(sample_rate == 0) { in do_shorthand_operation__cuesheet()
85 …>argument.import_cuesheet_from.filename, needs_write, lead_out_offset, sample_rate, is_cdda, opera… in do_shorthand_operation__cuesheet()
120 …ename, FLAC__bool *needs_write, FLAC__uint64 lead_out_offset, unsigned sample_rate, FLAC__bool is_… in import_cs_from() argument
141 …*cuesheet = grabbag__cuesheet_parse(f, &error_message, &last_line_read, sample_rate, is_cdda, lead… in import_cs_from()
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