/external/webrtc/webrtc/common_audio/include/ |
D | audio_util.h | 74 int num_channels, in CopyAudioIfNeeded() argument 76 for (int i = 0; i < num_channels; ++i) { in CopyAudioIfNeeded() 90 size_t num_channels, in Deinterleave() argument 92 for (size_t i = 0; i < num_channels; ++i) { in Deinterleave() 97 interleaved_idx += num_channels; in Deinterleave() 108 size_t num_channels, in Interleave() argument 110 for (size_t i = 0; i < num_channels; ++i) { in Interleave() 115 interleaved_idx += num_channels; in Interleave() 126 int num_channels, in UpmixMonoToInterleaved() argument 130 for (int j = 0; j < num_channels; ++j) { in UpmixMonoToInterleaved() [all …]
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/external/libxaac/decoder/drc_src/ |
D | impd_drc_peak_limiter.c | 37 FLOAT32 limit_threshold, UWORD32 num_channels, in impd_peak_limiter_init() argument 55 peak_limiter->num_channels = num_channels; in impd_peak_limiter_init() 74 peak_limiter->num_channels * in impd_peak_limiter_reinit() 87 UWORD32 num_channels = peak_limiter->num_channels; in impd_limiter_process() local 101 for (j = 0; j < num_channels; j++) { in impd_limiter_process() 102 tmp = max(tmp, (FLOAT32)fabs(samples[i * num_channels + j])); in impd_limiter_process() 142 for (j = 0; j < num_channels; j++) { in impd_limiter_process() 143 tmp = delayed_input[delayed_input_index * num_channels + j]; in impd_limiter_process() 144 delayed_input[delayed_input_index * num_channels + j] = in impd_limiter_process() 145 samples[i * num_channels + j]; in impd_limiter_process() [all …]
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/external/adhd/cras/src/server/ |
D | cras_fmt_conv.c | 372 unsigned int num_in_ch = conv->in_fmt.num_channels; in s16_default_all_to_all() 373 unsigned int num_out_ch = conv->out_fmt.num_channels; in s16_default_all_to_all() 437 for (i = 0; i < conv->out_fmt.num_channels; i++) in convert_channels() 441 conv->in_fmt.num_channels); in convert_channels() 442 in_idx += conv->in_fmt.num_channels; in convert_channels() 443 out_idx += conv->out_fmt.num_channels; in convert_channels() 560 if (in->num_channels != out->num_channels) { in cras_fmt_conv_create() 563 in->num_channels, out->num_channels); in cras_fmt_conv_create() 567 if (in->num_channels == 1 && out->num_channels == 2) { in cras_fmt_conv_create() 569 } else if (in->num_channels == 1 && out->num_channels == 6) { in cras_fmt_conv_create() [all …]
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D | cras_audio_area.c | 13 struct cras_audio_area *cras_audio_area_create(int num_channels) in cras_audio_area_create() argument 18 sz = sizeof(*area) + num_channels * sizeof(struct cras_channel_area); in cras_audio_area_create() 20 area->num_channels = num_channels; in cras_audio_area_create() 39 for (src_idx = 0; src_idx < src->num_channels; src_idx++) { in cras_audio_area_copy() 41 for (dst_idx = 0; dst_idx < dst->num_channels; dst_idx++) { in cras_audio_area_copy() 77 if ((fmt->num_channels == 1) && in cras_audio_area_config_channels() 85 for (i = 0; i < fmt->num_channels; i++) { in cras_audio_area_config_channels() 102 for (i = 0 ; i < area->num_channels; i++) { in cras_audio_area_config_buf_pointers()
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D | float_buffer.h | 20 unsigned int num_channels; member 31 unsigned int num_channels) in float_buffer_create() argument 37 b->num_channels = num_channels; in float_buffer_create() 38 b->fp = (float **)malloc(num_channels * sizeof(float *)); in float_buffer_create() 41 max_size * num_channels * sizeof(float)); in float_buffer_create() 65 for (i = 0; i < b->num_channels; i++, data += b->buf->max_size) in float_buffer_write_pointer() 104 for (i = 0; i < b->num_channels; i++, data += b->buf->max_size) in float_buffer_read_pointer()
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/external/webrtc/webrtc/common_audio/ |
D | wav_header.cc | 62 bool CheckWavParameters(size_t num_channels, in CheckWavParameters() argument 70 if (num_channels == 0 || sample_rate <= 0 || bytes_per_sample == 0) in CheckWavParameters() 74 if (num_channels > std::numeric_limits<uint16_t>::max()) in CheckWavParameters() 79 if (static_cast<uint64_t>(sample_rate) * num_channels * bytes_per_sample > in CheckWavParameters() 108 if (num_samples % num_channels != 0) 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() 143 static inline uint16_t BlockAlign(size_t num_channels, in BlockAlign() argument 145 return static_cast<uint16_t>(num_channels * bytes_per_sample); in BlockAlign() 149 size_t num_channels, in WriteWavHeader() argument [all …]
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D | audio_ring_buffer_unittest.cc | 27 const size_t num_channels = input.num_channels(); in ReadAndWriteTest() local 29 AudioRingBuffer buf(num_channels, buffer_frames); in ReadAndWriteTest() 30 rtc::scoped_ptr<float* []> slice(new float* [num_channels]); in ReadAndWriteTest() 37 buf.Write(input.Slice(slice.get(), input_pos), num_channels, in ReadAndWriteTest() 44 buf.Read(output->Slice(slice.get(), output_pos), num_channels, in ReadAndWriteTest() 52 buf.Write(input.Slice(slice.get(), input_pos), num_channels, in ReadAndWriteTest() 56 buf.Read(output->Slice(slice.get(), output_pos), num_channels, in ReadAndWriteTest() 64 const size_t num_channels = ::testing::get<3>(GetParam()); in TEST_P() local 67 ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels)); in TEST_P() 68 for (size_t i = 0; i < num_channels; ++i) in TEST_P() [all …]
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D | wav_header_unittest.cc | 94 size_t num_channels = 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() 271 size_t num_channels = 0; in TEST() local 278 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() [all …]
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D | channel_buffer.h | 43 size_t num_channels, 45 : data_(new T[num_frames * num_channels]()), in data_() argument 46 channels_(new T*[num_channels * num_bands]), in data_() 47 bands_(new T*[num_channels * num_bands]), in data_() 50 num_channels_(num_channels), in data_() 118 size_t num_channels() const { return num_channels_; } in num_channels() function 145 IFChannelBuffer(size_t num_frames, size_t num_channels, size_t num_bands = 1); 154 size_t num_channels() const { return ibuf_.num_channels(); } in num_channels() function
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D | blocker.cc | 25 size_t num_channels, in AddFrames() argument 28 for (size_t i = 0; i < num_channels; ++i) { in AddFrames() 40 size_t num_channels, in CopyFrames() argument 43 for (size_t i = 0; i < num_channels; ++i) { in CopyFrames() 54 size_t num_channels, in MoveFrames() argument 57 for (size_t i = 0; i < num_channels; ++i) { in MoveFrames() 67 size_t num_channels) { in ZeroOut() argument 68 for (size_t i = 0; i < num_channels; ++i) { in ZeroOut() 78 size_t num_channels, in ApplyWindow() argument 80 for (size_t i = 0; i < num_channels; ++i) { in ApplyWindow()
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/external/autotest/server/site_tests/brillo_PlaybackAudioTest/ |
D | brillo_PlaybackAudioTest.py | 70 num_channels, play_file_path=None): argument 83 num_channels=num_channels) 95 num_channels): argument 111 num_channels=num_channels) 117 self.host, num_channels, sample_rate, sample_width, 128 num_channels=num_channels, 159 for num_channels in num_channels_arr: 163 logging.info('Number of channels: %d', num_channels) 170 num_channels=num_channels) 174 num_channels))
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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 97 num_channels=num_channels, 103 num_channels=num_channels, 152 for num_channels in num_channels_arr: 158 logging.info('Number of channels: %d', num_channels) 165 num_channels=num_channels, [all …]
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/external/tensorflow/tensorflow/lite/experimental/microfrontend/lib/ |
D | filterbank_test.cc | 38 config_.num_channels = 2; in FilterbankTestConfig() 79 TF_LITE_MICRO_EXPECT_EQ(state.num_channels + 1, in TF_LITE_MICRO_TEST() 82 for (i = 0; i <= state.num_channels; ++i) { in TF_LITE_MICRO_TEST() 96 TF_LITE_MICRO_EXPECT_EQ(state.num_channels + 1, in TF_LITE_MICRO_TEST() 99 for (i = 0; i <= state.num_channels; ++i) { in TF_LITE_MICRO_TEST() 113 TF_LITE_MICRO_EXPECT_EQ(state.num_channels + 1, in TF_LITE_MICRO_TEST() 116 for (i = 0; i <= state.num_channels; ++i) { in TF_LITE_MICRO_TEST() 132 TF_LITE_MICRO_EXPECT_EQ(state.channel_weight_starts[state.num_channels] + in TF_LITE_MICRO_TEST() 133 state.channel_widths[state.num_channels], in TF_LITE_MICRO_TEST() 152 TF_LITE_MICRO_EXPECT_EQ(state.channel_weight_starts[state.num_channels] + in TF_LITE_MICRO_TEST() [all …]
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/external/webrtc/webrtc/modules/audio_processing/transient/ |
D | transient_suppression_test.cc | 56 DEFINE_int32(num_channels, 1, "Number of channels."); 79 int num_channels, in ReadBuffers() argument 88 if (num_channels > 1) { in ReadBuffers() 89 tmpbuf.reset(new int16_t[num_channels * audio_buffer_size]); in ReadBuffers() 94 num_channels * audio_buffer_size, in ReadBuffers() 95 in_file) != num_channels * audio_buffer_size) { in ReadBuffers() 99 if (num_channels > 1) { in ReadBuffers() 100 for (int i = 0; i < num_channels; ++i) { in ReadBuffers() 103 read_ptr[i + j * num_channels]; in ReadBuffers() 128 int num_channels, in WritePCM() argument [all …]
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/external/tensorflow/tensorflow/lite/experimental/micro/examples/micro_speech/micro_features/ |
D | filterbank_test.cc | 38 config_.num_channels = 2; in FilterbankTestConfig() 84 TF_LITE_MICRO_EXPECT_EQ(state.num_channels + 1, in TF_LITE_MICRO_TEST() 87 for (i = 0; i <= state.num_channels; ++i) { in TF_LITE_MICRO_TEST() 102 TF_LITE_MICRO_EXPECT_EQ(state.num_channels + 1, in TF_LITE_MICRO_TEST() 105 for (i = 0; i <= state.num_channels; ++i) { in TF_LITE_MICRO_TEST() 120 TF_LITE_MICRO_EXPECT_EQ(state.num_channels + 1, in TF_LITE_MICRO_TEST() 123 for (i = 0; i <= state.num_channels; ++i) { in TF_LITE_MICRO_TEST() 140 TF_LITE_MICRO_EXPECT_EQ(state.channel_weight_starts[state.num_channels] + in TF_LITE_MICRO_TEST() 141 state.channel_widths[state.num_channels], in TF_LITE_MICRO_TEST() 161 TF_LITE_MICRO_EXPECT_EQ(state.channel_weight_starts[state.num_channels] + in TF_LITE_MICRO_TEST() [all …]
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/external/webrtc/webrtc/modules/audio_processing/ |
D | splitting_filter.cc | 19 SplittingFilter::SplittingFilter(size_t num_channels, in SplittingFilter() argument 25 two_bands_states_.resize(num_channels); in SplittingFilter() 27 for (size_t i = 0; i < num_channels; ++i) { in SplittingFilter() 36 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels()); in Analysis() 49 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels()); in Synthesis() 61 RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels()); in TwoBandsAnalysis() 74 RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels()); in TwoBandsSynthesis() 87 RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels()); in ThreeBandsAnalysis() 97 RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels()); in ThreeBandsSynthesis()
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/external/adhd/cras/src/common/ |
D | cras_audio_format.c | 36 size_t num_channels) in cras_audio_format_create() argument 47 fmt->num_channels = num_channels; in cras_audio_format_create() 53 fmt->channel_layout[i] = (i < num_channels) ? i : -1; in cras_audio_format_create() 67 if (layout[i] >= (int)format->num_channels) in cras_audio_format_set_channel_layout() 117 if (in->channel_layout[i] >= (int)in->num_channels || in cras_channel_conv_matrix_create() 118 out->channel_layout[i] >= (int)out->num_channels) { in cras_channel_conv_matrix_create() 125 mtx = cras_channel_conv_matrix_alloc(in->num_channels, in cras_channel_conv_matrix_create() 126 out->num_channels); in cras_channel_conv_matrix_create() 160 cras_channel_conv_matrix_destroy(mtx, out->num_channels); in cras_channel_conv_matrix_create()
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D | cras_audio_format.h | 77 size_t num_channels; member 94 uint32_t num_channels; member 103 dest->num_channels = src->num_channels; in pack_cras_audio_format() 113 dest->num_channels = src->num_channels; in unpack_cras_audio_format() 124 return (size_t)bytes * fmt->num_channels; in cras_get_format_bytes() 134 format->channel_layout[i] = i < format->num_channels ? i : -1; in cras_audio_format_set_default_channel_layout() 140 size_t num_channels);
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/external/adhd/cras/src/tests/ |
D | fmt_conv_unittest.cc | 58 in_fmt.num_channels = out_fmt.num_channels = 1; in TEST() 107 in_fmt.num_channels = out_fmt.num_channels = 2; in TEST() 127 in_fmt.num_channels = 1; in TEST() 128 out_fmt.num_channels = 2; in TEST() 178 in_fmt.num_channels = 2; in TEST() 179 out_fmt.num_channels = 1; in TEST() 236 in_fmt.num_channels = 2; in TEST() 237 out_fmt.num_channels = 1; in TEST() 289 in_fmt.num_channels = 6; in TEST() 290 out_fmt.num_channels = 2; in TEST() [all …]
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/external/webrtc/webrtc/modules/audio_processing/test/ |
D | test_utils.cc | 40 RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels()); in Read() 48 Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(), in Read() 57 RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels()); in Write() 59 Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(), in Write() 79 size_t num_channels, in WriteFloatData() argument 82 size_t length = num_channels * samples_per_channel; in WriteFloatData() 84 Interleave(data, samples_per_channel, num_channels, buffer.get()); in WriteFloatData() 119 AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) { in LayoutFromChannels() argument 120 switch (num_channels) { in LayoutFromChannels()
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/external/autotest/server/brillo/feedback/ |
D | closed_loop_audio_client.py | 152 num_channels=_DEFAULT_NUM_CHANNELS, argument 161 self.num_channels = num_channels 170 (num_channels, duration_secs, sample_rate, sample_width, 191 num_channels=self.num_channels, 226 num_channels=self.num_channels, 276 num_channels=_DEFAULT_NUM_CHANNELS, argument 293 self.num_channels = num_channels 303 self.client.host, self.num_channels, 325 captured_audio_file, num_channels=self.num_channels,
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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 84 if num_channels is not None and chk_file.getnchannels() != num_channels: 86 num_channels, chk_file.getnchannels()) 106 def generate_sine_file(host, num_channels, sample_rate, sample_width, argument 129 'vol 0.9' % (num_channels, byte_format, 174 def _compare_frames(reference_file_frames, rec_file_frames, num_channels, argument 199 for channel in range(num_channels): 200 reference_data = reference_file_frames[channel::num_channels] 201 rec_data = rec_file_frames[channel::num_channels] 256 num_channels = reference_file.getnchannels()
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/external/webrtc/webrtc/modules/audio_processing/beamformer/ |
D | nonlinear_beamformer_test.cc | 48 const size_t num_mics = in_file.num_channels(); 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() 63 in_file.num_channels()); in main() 66 out_file.num_channels()); in main() 73 in_buf.num_channels(), in_buf.channels()); in main() 78 out_buf.num_channels(), &interleaved[0]); in main()
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
D | neteq_impl_unittest.cc | 469 size_t num_channels; in TEST_F() local 474 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type)); in TEST_F() 476 EXPECT_EQ(1u, num_channels); in TEST_F() 548 size_t num_channels; in TEST_F() local 553 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type)); in TEST_F() 555 EXPECT_EQ(1u, num_channels); in TEST_F() 584 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type)); in TEST_F() 586 EXPECT_EQ(1u, num_channels); in TEST_F() 625 size_t num_channels; in TEST_F() local 629 &num_channels, &type)); in TEST_F() [all …]
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/external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
D | audio_decoder_pcm.h | 21 explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) { in AudioDecoderPcmU() argument 22 RTC_DCHECK_GE(num_channels, 1u); in AudioDecoderPcmU() 42 explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) { in AudioDecoderPcmA() argument 43 RTC_DCHECK_GE(num_channels, 1u); in AudioDecoderPcmA()
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