Searched refs:num_channels (Results 1 - 25 of 197) sorted by relevance

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/external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/
H A Daudio_decoder_pcm16b.cc18 AudioDecoderPcm16B::AudioDecoderPcm16B(size_t num_channels) argument
19 : num_channels_(num_channels) {
20 RTC_DCHECK_GE(num_channels, 1u);
/external/autotest/server/site_tests/brillo_RecordingAudioTest/
H A Dbrillo_RecordingAudioTest.py36 def _get_recording_cmd(self, recording_method, duration_secs, num_channels,
43 @param num_channels: Number of channels to use for recording.
54 '--duration_secs=%d --num_channels=%d --filename=%s' %
55 (duration_secs, num_channels, rec_file) )
58 '--duration_secs=%d --num_channels=%d --filename=%s' %
59 (duration_secs, num_channels, rec_file))
68 sample_rate, num_channels):
76 @param num_channels: Number of channels to use for recording.
85 num_channels=num_channels,
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/external/webrtc/webrtc/modules/audio_coding/codecs/g711/
H A Daudio_decoder_pcm.h21 explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) { argument
22 RTC_DCHECK_GE(num_channels, 1u);
42 explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) { argument
43 RTC_DCHECK_GE(num_channels, 1u);
/external/webrtc/webrtc/modules/audio_processing/beamformer/
H A Dnonlinear_beamformer_test.cc48 const size_t num_mics = in_file.num_channels();
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());
63 in_file.num_channels());
66 out_file.num_channels());
73 in_buf.num_channels(), in_buf.channels());
78 out_buf.num_channels(), &interleaved[0]);
/external/webrtc/webrtc/common_audio/
H A Daudio_ring_buffer_unittest.cc27 const size_t num_channels = input.num_channels(); local
29 AudioRingBuffer buf(num_channels, buffer_frames);
30 rtc::scoped_ptr<float* []> slice(new float* [num_channels]);
37 buf.Write(input.Slice(slice.get(), input_pos), num_channels,
44 buf.Read(output->Slice(slice.get(), output_pos), num_channels,
52 buf.Write(input.Slice(slice.get(), input_pos), num_channels,
56 buf.Read(output->Slice(slice.get(), output_pos), num_channels,
64 const size_t num_channels = ::testing::get<3>(GetParam()); local
67 ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels));
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H A Dchannel_buffer.cc16 size_t num_channels,
19 ibuf_(num_frames, num_channels, num_bands),
21 fbuf_(num_frames, num_channels, num_bands) {}
50 for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
64 for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
15 IFChannelBuffer(size_t num_frames, size_t num_channels, size_t num_bands) argument
H A Dwav_header.cc62 bool CheckWavParameters(size_t num_channels, argument
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)
74 if (num_channels > std::numeric_limits<uint16_t>::max())
79 if (static_cast<uint64_t>(sample_rate) * num_channels * bytes_per_sample >
108 if (num_samples % num_channels != 0)
138 static inline uint32_t ByteRate(size_t num_channels, int sample_rate, argument
140 return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample);
143 static inline uint16_t BlockAlign(size_t num_channels, argument
145 return static_cast<uint16_t>(num_channels * bytes_per_sampl
148 WriteWavHeader(uint8_t* buf, size_t num_channels, int sample_rate, WavFormat format, size_t bytes_per_sample, size_t num_samples) argument
183 ReadWavHeader(ReadableWav* readable, size_t* num_channels, int* sample_rate, WavFormat* format, size_t* bytes_per_sample, size_t* num_samples) argument
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H A Dwav_header.h35 bool CheckWavParameters(size_t num_channels,
46 size_t num_channels,
56 size_t* num_channels,
H A Dblocker.cc25 size_t num_channels,
28 for (size_t i = 0; i < num_channels; ++i) {
40 size_t num_channels,
43 for (size_t i = 0; i < num_channels; ++i) {
54 size_t num_channels,
57 for (size_t i = 0; i < num_channels; ++i) {
67 size_t num_channels) {
68 for (size_t i = 0; i < num_channels; ++i) {
78 size_t num_channels,
80 for (size_t i = 0; i < num_channels;
20 AddFrames(const float* const* a, size_t a_start_index, const float* const* b, int b_start_index, size_t num_frames, size_t num_channels, float* const* result, size_t result_start_index) argument
37 CopyFrames(const float* const* src, size_t src_start_index, size_t num_frames, size_t num_channels, float* const* dst, size_t dst_start_index) argument
51 MoveFrames(const float* const* src, size_t src_start_index, size_t num_frames, size_t num_channels, float* const* dst, size_t dst_start_index) argument
64 ZeroOut(float* const* buffer, size_t starting_idx, size_t num_frames, size_t num_channels) argument
76 ApplyWindow(const float* window, size_t num_frames, size_t num_channels, float* const* frames) argument
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H A Dwav_file.h30 virtual size_t num_channels() const = 0;
42 WavWriter(const std::string& filename, int sample_rate, size_t num_channels);
54 size_t num_channels() const override { return num_channels_; }
82 size_t num_channels() const override { return num_channels_; }
105 size_t num_channels);
H A Dchannel_buffer.h43 size_t num_channels,
45 : data_(new T[num_frames * num_channels]()),
46 channels_(new T*[num_channels * num_bands]),
47 bands_(new T*[num_channels * num_bands]),
50 num_channels_(num_channels),
118 size_t num_channels() const { return num_channels_; } function in class:webrtc::ChannelBuffer
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(); } function in class:webrtc::IFChannelBuffer
42 ChannelBuffer(size_t num_frames, size_t num_channels, size_t num_bands = 1) argument
H A Dwav_header_unittest.cc94 size_t num_channels = 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,
271 size_t num_channels = 0; local
278 ReadWavHeader(&r, &num_channels,
307 size_t num_channels = 0; local
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/external/autotest/server/site_tests/brillo_PlaybackAudioTest/
H A Dbrillo_PlaybackAudioTest.py65 duration_secs, num_channels, play_file_path=None):
73 @param num_channels: Number of channels to test playback with.
78 num_channels=num_channels)
92 num_channels=_DEFAULT_NUM_CHANNELS,
104 @param num_channels: Number of channels to test playback with.
115 num_channels=num_channels,
129 'sine %d vol 0.9' % (num_channels, sine_format,
150 num_channels
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/external/webrtc/webrtc/common_audio/include/
H A Daudio_util.h74 int num_channels,
76 for (int i = 0; i < num_channels; ++i) {
90 size_t num_channels,
92 for (size_t i = 0; i < num_channels; ++i) {
97 interleaved_idx += num_channels;
104 // (|samples_per_channel| * |num_channels|).
108 size_t num_channels,
110 for (size_t i = 0; i < num_channels; ++i) {
115 interleaved_idx += num_channels;
122 // |interleaved| (|samples_per_channel| * |num_channels|)
72 CopyAudioIfNeeded(const T* const* src, int num_frames, int num_channels, T* const* dest) argument
88 Deinterleave(const T* interleaved, size_t samples_per_channel, size_t num_channels, T* const* deinterleaved) argument
106 Interleave(const T* const* deinterleaved, size_t samples_per_channel, size_t num_channels, T* interleaved) argument
124 UpmixMonoToInterleaved(const T* mono, int num_frames, int num_channels, T* interleaved) argument
137 DownmixToMono(const T* const* input_channels, size_t num_frames, int num_channels, T* out) argument
153 DownmixInterleavedToMonoImpl(const T* interleaved, size_t num_frames, int num_channels, T* deinterleaved) argument
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/external/webrtc/webrtc/modules/audio_coding/neteq/
H A Daccelerate.h32 Accelerate(int sample_rate_hz, size_t num_channels, argument
34 : TimeStretch(sample_rate_hz, num_channels, background_noise) {
76 size_t num_channels,
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/
H A Dmock_expand.h27 size_t num_channels)
33 num_channels) {}
60 size_t num_channels));
22 MockExpand(BackgroundNoise* background_noise, SyncBuffer* sync_buffer, RandomVector* random_vector, StatisticsCalculator* statistics, int fs, size_t num_channels) argument
/external/webrtc/webrtc/modules/audio_processing/
H A Dsplitting_filter.cc19 SplittingFilter::SplittingFilter(size_t num_channels, argument
25 two_bands_states_.resize(num_channels);
27 for (size_t i = 0; i < num_channels; ++i) {
36 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels());
49 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels());
61 RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
74 RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
87 RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels());
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/external/webrtc/webrtc/modules/audio_processing/test/
H A Dtest_utils.cc40 RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
48 Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
57 RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
59 Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
79 size_t num_channels,
82 size_t length = num_channels * samples_per_channel;
84 Interleave(data, samples_per_channel, num_channels, buffer.get());
119 AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) { argument
77 WriteFloatData(const float* const* data, size_t samples_per_channel, size_t num_channels, WavWriter* wav_file, RawFile* raw_file) argument
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/external/webrtc/webrtc/common_audio/resampler/
H A Dpush_resampler.cc35 size_t num_channels) {
38 num_channels == num_channels_)
43 num_channels <= 0 || num_channels > 2)
48 num_channels_ = num_channels;
33 InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz, size_t num_channels) argument
/external/webrtc/webrtc/modules/audio_processing/transient/
H A Dtransient_suppression_test.cc56 DEFINE_int32(num_channels, 1, "Number of channels.");
65 "num_channels and sample_rate_hz, the detection signal from the\n"
79 int num_channels,
88 if (num_channels > 1) {
89 tmpbuf.reset(new int16_t[num_channels * audio_buffer_size]);
94 num_channels * audio_buffer_size,
95 in_file) != num_channels * audio_buffer_size) {
99 if (num_channels > 1) {
100 for (int i = 0; i < num_channels; ++i) {
103 read_ptr[i + j * num_channels];
77 ReadBuffers(FILE* in_file, size_t audio_buffer_size, int num_channels, int16_t* audio_buffer, FILE* detection_file, size_t detection_buffer_size, float* detection_buffer, FILE* reference_file, float* reference_buffer) argument
126 WritePCM(FILE* f, size_t num_samples, int num_channels, const float* buffer) argument
[all...]
/external/mesa3d/src/gallium/auxiliary/vl/
H A Dvl_zscan.h45 unsigned num_channels; member in struct:vl_zscan
78 unsigned num_channels);
/external/opencv3/3rdparty/libwebp/utils/
H A Drescaler.h26 int num_channels; // bytes to jump between pixels member in struct:__anon15403
44 int num_channels,
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/
H A Daudio_decoder_opus.cc17 AudioDecoderOpus::AudioDecoderOpus(size_t num_channels) argument
18 : channels_(num_channels) {
19 RTC_DCHECK(num_channels == 1 || num_channels == 2);
/external/webrtc/webrtc/voice_engine/
H A Dutility.h39 // |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as
43 size_t num_channels,
/external/webrtc/webrtc/common_audio/resampler/include/
H A Dresampler.h31 Resampler(int inFreq, int outFreq, size_t num_channels);
35 int Reset(int inFreq, int outFreq, size_t num_channels);
38 int ResetIfNeeded(int inFreq, int outFreq, size_t num_channels);

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