| /external/webrtc/api/audio_codecs/ |
| D | audio_format.cc | 24 size_t num_channels) in SdpAudioFormat() argument
25 : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {} in SdpAudioFormat()
29 size_t num_channels, in SdpAudioFormat() argument
33 num_channels(num_channels), in SdpAudioFormat()
38 size_t num_channels, in SdpAudioFormat() argument
42 num_channels(num_channels), in SdpAudioFormat()
47 clockrate_hz == o.clockrate_hz && num_channels == o.num_channels; in Matches()
56 a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels && in operator ==()
61 size_t num_channels, in AudioCodecInfo() argument
64 num_channels, in AudioCodecInfo()
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| /external/webrtc/modules/audio_processing/aec3/ |
| D | alignment_mixer_unittest.cc | 29 int num_channels, in ProduceDebugText() argument
35 ss << ", Number of channels: " << num_channels; in ProduceDebugText()
51 for (int num_channels = 2; num_channels < 8; ++num_channels) { in TEST() local
52 for (int strongest_ch = 0; strongest_ch < num_channels; in TEST()
56 prefer_first_two_channels, num_channels, strongest_ch)); in TEST()
59 AlignmentMixer am(num_channels, /*downmix*/ false, in TEST()
64 num_channels, std::vector<float>(kBlockSize, 0.f)); in TEST()
66 for (int ch = 0; ch < num_channels; ++ch) { in TEST()
82 for (int ch = 0; ch < num_channels; ++ch) { in TEST()
122 for (int num_channels = 1; num_channels < 8; ++num_channels) { in TEST() local
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| D | render_signal_analyzer_unittest.cc | 57 void RunNarrowBandDetectionTest(size_t num_channels) { in RunNarrowBandDetectionTest() argument
64 num_channels, std::vector<float>(kBlockSize, 0.f))); in RunNarrowBandDetectionTest()
69 RenderDelayBuffer::Create(config, kSampleRateHz, num_channels)); in RunNarrowBandDetectionTest()
78 ProduceSinusoidInNoise(16000, num_channels - 1, in RunNarrowBandDetectionTest()
111 std::string ProduceDebugText(size_t num_channels) { in ProduceDebugText() argument
113 ss << "number of channels: " << num_channels; in ProduceDebugText()
129 for (auto num_channels : {1, 2, 8}) { in TEST()
130 SCOPED_TRACE(ProduceDebugText(num_channels)); in TEST()
135 num_channels, std::vector<float>(kBlockSize, 0.f))); in TEST()
138 RenderDelayBuffer::Create(EchoCanceller3Config(), 48000, num_channels)); in TEST()
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| D | frame_blocker_unittest.cc | 107 void RunBlockerTest(int sample_rate_hz, size_t num_channels) { in RunBlockerTest() argument
113 num_channels, std::vector<float>(kBlockSize, 0.f))); in RunBlockerTest()
116 num_channels, std::vector<float>(kSubFrameLength, 0.f))); in RunBlockerTest()
118 num_bands, std::vector<rtc::ArrayView<float>>(num_channels)); in RunBlockerTest()
119 FrameBlocker blocker(num_bands, num_channels); in RunBlockerTest()
144 void RunBlockerAndFramerTest(int sample_rate_hz, size_t num_channels) { in RunBlockerAndFramerTest() argument
150 num_channels, std::vector<float>(kBlockSize, 0.f))); in RunBlockerAndFramerTest()
153 num_channels, std::vector<float>(kSubFrameLength, 0.f))); in RunBlockerAndFramerTest()
156 num_channels, std::vector<float>(kSubFrameLength, 0.f))); in RunBlockerAndFramerTest()
158 num_bands, std::vector<rtc::ArrayView<float>>(num_channels)); in RunBlockerAndFramerTest()
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| /external/webrtc/modules/audio_processing/ns/ |
| D | noise_suppressor_unittest.cc | 27 size_t num_channels, in ProduceDebugText() argument
30 ss << "Sample rate: " << sample_rate_hz << ", num_channels: " << num_channels in ProduceDebugText()
35 void PopulateInputFrameWithIdenticalChannels(size_t num_channels, in PopulateInputFrameWithIdenticalChannels() argument
39 for (size_t ch = 0; ch < num_channels; ++ch) { in PopulateInputFrameWithIdenticalChannels()
49 void VerifyIdenticalChannels(size_t num_channels, in VerifyIdenticalChannels() argument
53 EXPECT_GT(num_channels, 1u); in VerifyIdenticalChannels()
54 for (size_t ch = 1; ch < num_channels; ++ch) { in VerifyIdenticalChannels()
69 for (auto num_channels : {1, 4, 8}) { in TEST()
74 SCOPED_TRACE(ProduceDebugText(rate, num_channels, level)); in TEST()
78 AudioBuffer audio(rate, num_channels, rate, num_channels, rate, in TEST()
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| /external/adhd/cras/src/server/ |
| D | cras_fmt_conv.c | 277 num_in_ch = conv->in_fmt.num_channels; in default_all_to_all()
278 num_out_ch = conv->out_fmt.num_channels; in default_all_to_all()
292 num_in_ch = conv->in_fmt.num_channels; in default_some_to_some()
293 num_out_ch = conv->out_fmt.num_channels; in default_some_to_some()
306 num_in_ch = conv->in_fmt.num_channels; in convert_channels()
307 num_out_ch = conv->out_fmt.num_channels; in convert_channels()
395 if (in->num_channels != out->num_channels) { in cras_fmt_conv_create()
398 in->num_channels, out->num_channels); in cras_fmt_conv_create()
402 if (in->num_channels == 1 && out->num_channels == 2) { in cras_fmt_conv_create()
404 } else if (in->num_channels == 1 && out->num_channels == 6) { in cras_fmt_conv_create()
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| /external/webrtc/common_audio/include/ |
| D | audio_util.h | 100 int num_channels, in CopyAudioIfNeeded() argument
102 for (int i = 0; i < num_channels; ++i) { in CopyAudioIfNeeded()
116 size_t num_channels, in Deinterleave() argument
118 for (size_t i = 0; i < num_channels; ++i) { in Deinterleave()
123 interleaved_idx += num_channels; in Deinterleave()
134 size_t num_channels, in Interleave() argument
136 for (size_t i = 0; i < num_channels; ++i) { in Interleave()
141 interleaved_idx += num_channels; in Interleave()
152 int num_channels, in UpmixMonoToInterleaved() argument
156 for (int j = 0; j < num_channels; ++j) { in UpmixMonoToInterleaved()
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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()
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| /external/webrtc/common_audio/ |
| D | wav_header.cc | 154 uint32_t ByteRate(size_t num_channels, in ByteRate() argument
157 return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample); in ByteRate()
160 uint16_t BlockAlign(size_t num_channels, size_t bytes_per_sample) { in BlockAlign() argument
161 return static_cast<uint16_t>(num_channels * bytes_per_sample); in BlockAlign()
203 void WritePcmWavHeader(size_t num_channels, in WritePcmWavHeader() argument
221 header.fmt.NumChannels = static_cast<uint16_t>(num_channels); in WritePcmWavHeader()
223 header.fmt.ByteRate = ByteRate(num_channels, sample_rate, bytes_per_sample); in WritePcmWavHeader()
224 header.fmt.BlockAlign = BlockAlign(num_channels, bytes_per_sample); in WritePcmWavHeader()
234 void WriteIeeeFloatWavHeader(size_t num_channels, in WriteIeeeFloatWavHeader() argument
253 header.fmt.NumChannels = static_cast<uint16_t>(num_channels); in WriteIeeeFloatWavHeader()
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| D | wav_header_unittest.cc | 102 size_t num_channels = 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()
313 size_t num_channels = 0; in TEST() local
321 EXPECT_TRUE(ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST()
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| D | channel_buffer.h | 45 ChannelBuffer(size_t num_frames, size_t num_channels, size_t num_bands = 1)
46 : data_(new T[num_frames * num_channels]()), in data_() argument
47 channels_(new T*[num_channels * num_bands]), in data_()
48 bands_(new T*[num_channels * num_bands]), in data_()
51 num_allocated_channels_(num_channels), in data_()
52 num_channels_(num_channels), in data_()
147 size_t num_channels() const { return num_channels_; } in num_channels() function
151 void set_num_channels(size_t num_channels) { in set_num_channels() argument
152 RTC_DCHECK_LE(num_channels, num_allocated_channels_); in set_num_channels()
153 num_channels_ = num_channels; in set_num_channels()
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| D | channel_buffer_unittest.cc | 24 void ExpectNumChannels(const IFChannelBuffer& ifchb, size_t num_channels) { in ExpectNumChannels() argument
25 EXPECT_EQ(ifchb.ibuf_const()->num_channels(), num_channels); in ExpectNumChannels()
26 EXPECT_EQ(ifchb.fbuf_const()->num_channels(), num_channels); in ExpectNumChannels()
27 EXPECT_EQ(ifchb.num_channels(), num_channels); in ExpectNumChannels()
34 EXPECT_EQ(chb.num_channels(), kStereo); in TEST()
36 EXPECT_EQ(chb.num_channels(), kMono); in TEST()
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| /external/webrtc/modules/audio_processing/ |
| D | splitting_filter.cc | 28 SplittingFilter::SplittingFilter(size_t num_channels, in SplittingFilter() argument
32 two_bands_states_(num_bands_ == 2 ? num_channels : 0), in SplittingFilter()
33 three_band_filter_banks_(num_bands_ == 3 ? num_channels : 0) { in SplittingFilter()
42 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels()); in Analysis()
55 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels()); in Synthesis()
67 RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels()); in TwoBandsAnalysis()
85 RTC_DCHECK_LE(data->num_channels(), two_bands_states_.size()); in TwoBandsSynthesis()
87 for (size_t i = 0; i < data->num_channels(); ++i) { in TwoBandsSynthesis()
102 RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels()); in ThreeBandsAnalysis()
103 RTC_DCHECK_LE(data->num_channels(), three_band_filter_banks_.size()); in ThreeBandsAnalysis()
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| D | high_pass_filter_unittest.cc | 30 stream_config.sample_rate_hz(), stream_config.num_channels(), in ProcessOneFrameAsAudioBuffer()
31 stream_config.sample_rate_hz(), stream_config.num_channels(), in ProcessOneFrameAsAudioBuffer()
32 stream_config.sample_rate_hz(), stream_config.num_channels()); in ProcessOneFrameAsAudioBuffer()
48 stream_config.num_channels(), in ProcessOneFrameAsVector()
52 for (size_t channel = 0; channel < stream_config.num_channels(); in ProcessOneFrameAsVector()
55 frame_input[k * stream_config.num_channels() + channel]; in ProcessOneFrameAsVector()
63 for (size_t channel = 0; channel < stream_config.num_channels(); in ProcessOneFrameAsVector()
74 void RunBitexactnessTest(int num_channels, in RunBitexactnessTest() argument
78 const StreamConfig stream_config(16000, num_channels, false); in RunBitexactnessTest()
79 HighPassFilter high_pass_filter(16000, num_channels); in RunBitexactnessTest()
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| /external/webrtc/audio/utility/ |
| D | channel_mixer_unittest.cc | 46 for (size_t i = 0; i < frame->samples_per_channel() * frame->num_channels(); in SetFrameData()
120 for (size_t i = 0; i < frame->samples_per_channel() * frame->num_channels(); in AllSamplesEquals()
130 EXPECT_EQ(frame1.num_channels(), frame2.num_channels()); in VerifyFramesAreEqual()
134 for (size_t i = 0; i < frame1.samples_per_channel() * frame1.num_channels(); in VerifyFramesAreEqual()
203 static_cast<int>(frame_.num_channels())); in TEST_F()
217 EXPECT_EQ(2u, frame_.num_channels()); in TEST_F()
219 EXPECT_EQ(1u, frame_.num_channels()); in TEST_F()
228 EXPECT_EQ(2u, frame_.num_channels()); in TEST_F()
230 EXPECT_EQ(1u, frame_.num_channels()); in TEST_F()
240 EXPECT_EQ(1u, frame_.num_channels()); in TEST_F()
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| /external/webrtc/modules/audio_device/include/ |
| D | test_audio_device.cc | 217 int num_channels) in PulsedNoiseCapturerImpl() argument
222 num_channels_(num_channels) { in PulsedNoiseCapturerImpl()
271 int num_channels, in WavFileReader() argument
275 num_channels, in WavFileReader()
305 int num_channels, in WavFileReader() argument
308 num_channels_(num_channels), in WavFileReader()
312 RTC_CHECK_EQ(wav_reader_->num_channels(), num_channels); in WavFileReader()
325 int num_channels) in WavFileWriter() argument
328 num_channels), in WavFileWriter()
330 num_channels) {} in WavFileWriter()
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| /external/webrtc/test/fuzzers/ |
| D | audio_processing_fuzzer_helper.cc | 31 size_t num_channels, in GenerateFloatFrame() argument
36 for (size_t i = 0; i < num_channels; ++i) { in GenerateFloatFrame()
56 size_t num_channels, in GenerateFixedFrame() argument
62 fixed_frame->num_channels_ = num_channels; in GenerateFixedFrame()
64 RTC_DCHECK_LE(samples_per_input_channel * num_channels, in GenerateFixedFrame()
66 for (size_t i = 0; i < samples_per_input_channel * num_channels; ++i) { in GenerateFixedFrame()
115 const int num_channels = in FuzzAudioProcessing() local
118 GenerateFloatFrame(fuzz_data, input_rate, num_channels, in FuzzAudioProcessing()
122 ptr_to_float_frames, StreamConfig(input_rate, num_channels), in FuzzAudioProcessing()
123 StreamConfig(output_rate, num_channels), ptr_to_float_frames); in FuzzAudioProcessing()
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| /external/webrtc/modules/audio_coding/codecs/opus/test/ |
| D | audio_ring_buffer_unittest.cc | 29 const size_t num_channels = input.num_channels(); in ReadAndWriteTest() local
31 AudioRingBuffer buf(num_channels, buffer_frames); in ReadAndWriteTest()
32 std::unique_ptr<float*[]> slice(new float*[num_channels]); in ReadAndWriteTest()
39 buf.Write(input.Slice(slice.get(), input_pos), num_channels, in ReadAndWriteTest()
46 buf.Read(output->Slice(slice.get(), output_pos), num_channels, in ReadAndWriteTest()
54 buf.Write(input.Slice(slice.get(), input_pos), num_channels, in ReadAndWriteTest()
58 buf.Read(output->Slice(slice.get(), output_pos), num_channels, in ReadAndWriteTest()
66 const size_t num_channels = ::testing::get<3>(GetParam()); in TEST_P() local
69 ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels)); in TEST_P()
70 for (size_t i = 0; i < num_channels; ++i) in TEST_P()
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| /external/adhd/cras/client/cras_tests/src/ |
| D | audio.rs | 86 fn channel_string(num_channels: usize) -> String { in channel_string()
87 match num_channels { in channel_string()
90 _ => format!("{} Channels", num_channels), in channel_string()
97 num_channels: usize, field
120 let num_channels = spec.channels as usize; in try_new() localVariable
121 if opts.num_channels.is_some() && Some(num_channels) != opts.num_channels { in try_new()
122 eprintln!("Warning: number of channels changed to {}", num_channels); in try_new()
133 num_channels, in try_new()
142 fn num_channels(&self) -> usize { in num_channels() method
143 self.num_channels in num_channels()
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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()
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| /external/webrtc/common_audio/resampler/ |
| D | push_resampler.cc | 32 size_t num_channels) { in CheckValidInitParams() argument
38 RTC_DCHECK_GT(num_channels, 0); in CheckValidInitParams()
44 size_t num_channels, in CheckExpectedBufferSizes() argument
52 const size_t src_size_10ms = src_sample_rate * num_channels / 100; in CheckExpectedBufferSizes()
53 const size_t dst_size_10ms = dst_sample_rate * num_channels / 100; in CheckExpectedBufferSizes()
70 size_t num_channels) { in InitializeIfNeeded() argument
71 CheckValidInitParams(src_sample_rate_hz, dst_sample_rate_hz, num_channels); in InitializeIfNeeded()
75 num_channels == num_channels_) { in InitializeIfNeeded()
80 if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 || num_channels <= 0) { in InitializeIfNeeded()
86 num_channels_ = num_channels; in InitializeIfNeeded()
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| /external/webrtc/api/audio_codecs/opus/ |
| D | audio_decoder_opus.cc | 28 if (num_channels != 1 && num_channels != 2) { in IsOk()
36 const auto num_channels = [&]() -> absl::optional<int> { in SdpToConfig() local
50 format.clockrate_hz == 48000 && format.num_channels == 2 && in SdpToConfig()
51 num_channels) { in SdpToConfig()
53 config.num_channels = *num_channels; in SdpToConfig()
75 return std::make_unique<AudioDecoderOpusImpl>(config.num_channels, in MakeAudioDecoder()
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| /external/adhd/cras/src/common/ |
| D | cras_audio_format.c | 34 size_t num_channels) in cras_audio_format_create() argument
45 fmt->num_channels = num_channels; in cras_audio_format_create()
51 fmt->channel_layout[i] = (i < num_channels) ? i : -1; in cras_audio_format_create()
65 if (layout[i] < -1 || layout[i] >= (int)format->num_channels) in cras_audio_format_set_channel_layout()
80 fmt->channel_layout[i] >= (int)fmt->num_channels) { in cras_audio_format_valid()
128 if (in->channel_layout[i] >= (int)in->num_channels || in cras_channel_conv_matrix_create()
129 out->channel_layout[i] >= (int)out->num_channels) { in cras_channel_conv_matrix_create()
136 mtx = cras_channel_conv_matrix_alloc(in->num_channels, in cras_channel_conv_matrix_create()
137 out->num_channels); in cras_channel_conv_matrix_create()
169 cras_channel_conv_matrix_destroy(mtx, out->num_channels); in cras_channel_conv_matrix_create()
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| /external/webrtc/modules/audio_processing/transient/ |
| D | transient_suppression_test.cc | 51 ABSL_FLAG(int, num_channels, 1, "Number of channels.");
72 int num_channels, in ReadBuffers() argument
81 if (num_channels > 1) { in ReadBuffers()
82 tmpbuf.reset(new int16_t[num_channels * audio_buffer_size]); in ReadBuffers()
85 if (fread(read_ptr, sizeof(*read_ptr), num_channels * audio_buffer_size, in ReadBuffers()
86 in_file) != num_channels * audio_buffer_size) { in ReadBuffers()
90 if (num_channels > 1) { in ReadBuffers()
91 for (int i = 0; i < num_channels; ++i) { in ReadBuffers()
94 read_ptr[i + j * num_channels]; in ReadBuffers()
119 int num_channels, in WritePCM() argument
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| /external/adhd/audio_streams/src/ |
| D | shm_streams.rs | 130 fn num_channels(&self) -> usize; in num_channels() method
199 num_channels: usize, in new_stream()
220 num_channels: usize, field
234 num_channels: usize, in new()
240 num_channels, in new()
243 frame_size: format.sample_bytes() * num_channels, in new()
266 fn num_channels(&self) -> usize { in num_channels() method
267 self.num_channels in num_channels()
306 num_channels: usize, in new_stream()
314 let new_stream = NullShmStream::new(buffer_size, num_channels, format, frame_rate);
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