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Searched refs:num_channels_ (Results 1 – 25 of 109) sorted by relevance

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/external/webrtc/modules/audio_coding/neteq/
Daudio_multi_vector.cc28 num_channels_ = N; in AudioMultiVector()
38 num_channels_ = N; in AudioMultiVector()
50 for (size_t i = 0; i < num_channels_; ++i) { in Clear()
56 for (size_t i = 0; i < num_channels_; ++i) { in Zeros()
64 for (size_t i = 0; i < num_channels_; ++i) { in CopyTo()
72 RTC_DCHECK_EQ(append_this.size() % num_channels_, 0); in PushBackInterleaved()
73 if (num_channels_ == 1) { in PushBackInterleaved()
78 size_t length_per_channel = append_this.size() / num_channels_; in PushBackInterleaved()
80 for (size_t channel = 0; channel < num_channels_; ++channel) { in PushBackInterleaved()
86 source_ptr += num_channels_; // Jump to next element of this channel. in PushBackInterleaved()
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Daudio_multi_vector_unittest.cc34 : num_channels_(GetParam()), // Get the test parameter. in AudioMultiVectorTest()
35 array_interleaved_(num_channels_ * array_length()) {} in AudioMultiVectorTest()
49 for (size_t j = 1; j <= num_channels_; ++j) { in SetUp()
58 const size_t num_channels_; member in webrtc::AudioMultiVectorTest
66 AudioMultiVector vec1(num_channels_); in TEST_P()
68 EXPECT_EQ(num_channels_, vec1.Channels()); in TEST_P()
72 AudioMultiVector vec2(num_channels_, initial_size); in TEST_P()
74 EXPECT_EQ(num_channels_, vec2.Channels()); in TEST_P()
80 AudioMultiVector vec(num_channels_, array_length()); in TEST_P()
81 for (size_t channel = 0; channel < num_channels_; ++channel) { in TEST_P()
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Dneteq_stereo_unittest.cc55 : num_channels_(GetParam().num_channels), in NetEqStereoTest()
77 input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_]; in NetEqStereoTest()
79 new uint8_t[frame_size_samples_ * 2 * num_channels_]; in NetEqStereoTest()
93 RTC_CHECK_GE(num_channels_, 2); in SetUp()
98 SdpAudioFormat("l16", sample_rate_hz_, num_channels_))); in SetUp()
116 input_multi_channel_, frame_size_samples_ * num_channels_, in GetNewPackets()
118 if (frame_size_samples_ * 2 * num_channels_ != multi_payload_size_bytes_) { in GetNewPackets()
128 input_, frame_size_samples_, num_channels_, input_multi_channel_); in MakeMultiChannelInput()
135 for (size_t j = 0; j < num_channels_; ++j) { in VerifyOutput()
137 output_multi_channel_data[i * num_channels_ + j]) in VerifyOutput()
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Dpreemptive_expand.cc31 if (num_channels_ == 0 || in Process()
32 input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ || in Process()
33 old_data_length >= input_length / num_channels_ - overlap_samples_) { in Process()
82 input, (unmodified_length + peak_index) * num_channels_)); in CheckCriteriaAndStretch()
84 AudioMultiVector temp_vector(num_channels_); in CheckCriteriaAndStretch()
86 &input[(unmodified_length - peak_index) * num_channels_], in CheckCriteriaAndStretch()
87 peak_index * num_channels_)); in CheckCriteriaAndStretch()
92 &input[unmodified_length * num_channels_], in CheckCriteriaAndStretch()
93 input_length - unmodified_length * num_channels_)); in CheckCriteriaAndStretch()
Dexpand_unittest.cc76 num_channels_(1), in ExpandTest()
77 background_noise_(num_channels_), in ExpandTest()
78 sync_buffer_(num_channels_, in ExpandTest()
85 num_channels_) { in ExpandTest()
100 ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels."; in SetUp()
105 size_t num_channels_; member in webrtc::ExpandTest
118 AudioMultiVector output(num_channels_); in TEST_F()
137 AudioMultiVector output(num_channels_); in TEST_F()
154 AudioMultiVector output(num_channels_); in TEST_F()
189 ExpandUntilMuted(num_channels_, &expand_); in TEST_F()
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Daccelerate.cc27 if (num_channels_ == 0 || in Process()
28 input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) { in Process()
75 rtc::ArrayView<const int16_t>(input, fs_mult_120 * num_channels_)); in CheckCriteriaAndStretch()
77 AudioMultiVector temp_vector(num_channels_); in CheckCriteriaAndStretch()
79 &input[fs_mult_120 * num_channels_], peak_index * num_channels_)); in CheckCriteriaAndStretch()
84 &input[(fs_mult_120 + peak_index) * num_channels_], in CheckCriteriaAndStretch()
85 input_length - (fs_mult_120 + peak_index) * num_channels_)); in CheckCriteriaAndStretch()
Dmerge_unittest.cc51 num_channels_(1), in MergeTest()
52 background_noise_(num_channels_), in MergeTest()
53 sync_buffer_(num_channels_, in MergeTest()
60 num_channels_), in MergeTest()
61 merge_(test_sample_rate_hz_, num_channels_, &expand_, &sync_buffer_) { in MergeTest()
79 ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels."; in SetUp()
85 size_t num_channels_; member in webrtc::MergeTest
95 AudioMultiVector output(num_channels_); in TEST_P()
Dbackground_noise.cc34 : num_channels_(num_channels), in BackgroundNoise()
35 channel_parameters_(new ChannelParameters[num_channels_]) { in BackgroundNoise()
43 for (size_t channel = 0; channel < num_channels_; ++channel) { in Reset()
62 for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) { in Update()
181 assert(channel < num_channels_); in Energy()
186 assert(channel < num_channels_); in SetMuteFactor()
191 assert(channel < num_channels_); in MuteFactor()
196 assert(channel < num_channels_); in Filter()
201 assert(channel < num_channels_); in FilterState()
207 assert(channel < num_channels_); in SetFilterState()
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/external/webrtc/audio/utility/
Daudio_frame_operations_unittest.cc24 frame_.num_channels_ = 2; in AudioFrameOperationsTest()
56 for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_; in SetFrameData()
63 EXPECT_EQ(frame1.num_channels_, frame2.num_channels_); in VerifyFramesAreEqual()
67 for (size_t i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_; in VerifyFramesAreEqual()
84 frame->num_channels_ = channels; in InitFrame()
93 RTC_DCHECK_LT(channel, frame.num_channels_); in GetChannelData()
95 return frame.data()[index * frame.num_channels_ + channel]; in GetChannelData()
113 frame_.num_channels_ = 1; in TEST_F()
119 frame_.num_channels_ = 1; in TEST_F()
123 EXPECT_EQ(2u, frame_.num_channels_); in TEST_F()
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Daudio_frame_operations.cc36 RTC_DCHECK_GT(result_frame->num_channels_, 0); in Add()
37 RTC_DCHECK_EQ(result_frame->num_channels_, frame_to_add.num_channels_); in Add()
62 frame_to_add.samples_per_channel_ * frame_to_add.num_channels_; in Add()
76 if (frame->num_channels_ != 1) { in MonoToStereo()
84 if (frame->num_channels_ != 2) { in StereoToMono()
88 return frame->num_channels_ == 1 ? 0 : -1; in StereoToMono()
104 if (frame->num_channels_ != 4) { in QuadToStereo()
115 frame->num_channels_ = 2; in QuadToStereo()
140 RTC_DCHECK_LE(frame->samples_per_channel_ * frame->num_channels_, in DownmixChannels()
142 if (frame->num_channels_ > 1 && dst_channels == 1) { in DownmixChannels()
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/external/webrtc/modules/audio_coding/codecs/g722/
Daudio_encoder_g722.cc29 : num_channels_(config.num_channels), in AudioEncoderG722Impl()
35 encoders_(new EncoderState[num_channels_]), in AudioEncoderG722Impl()
36 interleave_buffer_(2 * num_channels_) { in AudioEncoderG722Impl()
40 for (size_t i = 0; i < num_channels_; ++i) { in AudioEncoderG722Impl()
54 return num_channels_; in NumChannels()
78 for (size_t i = 0; i < num_channels_; ++i) in Reset()
98 for (size_t j = 0; j < num_channels_; ++j) in EncodeImpl()
99 encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j]; in EncodeImpl()
110 for (size_t i = 0; i < num_channels_; ++i) { in EncodeImpl()
117 const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_; in EncodeImpl()
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/external/webrtc/common_audio/resampler/
Dpush_resampler.cc62 : src_sample_rate_hz_(0), dst_sample_rate_hz_(0), num_channels_(0) {} in PushResampler()
75 num_channels == num_channels_) { in InitializeIfNeeded()
86 num_channels_ = num_channels; in InitializeIfNeeded()
102 channel_data_array_.resize(num_channels_); in InitializeIfNeeded()
112 CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_, in Resample()
122 const size_t src_length_mono = src_length / num_channels_; in Resample()
123 const size_t dst_capacity_mono = dst_capacity / num_channels_; in Resample()
125 for (size_t ch = 0; ch < num_channels_; ++ch) { in Resample()
129 Deinterleave(src, src_length_mono, num_channels_, channel_data_array_.data()); in Resample()
139 for (size_t ch = 0; ch < num_channels_; ++ch) { in Resample()
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/external/webrtc/modules/audio_coding/acm2/
Dacm_remixing.cc18 RTC_DCHECK_EQ(input.num_channels_, 2); in DownMixFrame()
36 RTC_DCHECK(!(input.num_channels_ == 0 && num_output_channels > 0 && in ReMixFrame()
51 if (input.num_channels_ == 0) { in ReMixFrame()
60 if (input.num_channels_ == 1 && input.num_channels_ < num_output_channels) { in ReMixFrame()
77 if (input.num_channels_ < num_output_channels) { in ReMixFrame()
79 for (size_t j = 0; j < input.num_channels_; ++j) { in ReMixFrame()
82 for (size_t j = input.num_channels_; j < num_output_channels; ++j) { in ReMixFrame()
85 RTC_DCHECK_EQ(in_index, (k + 1) * input.num_channels_); in ReMixFrame()
88 RTC_DCHECK_EQ(in_index, input.samples_per_channel_ * input.num_channels_); in ReMixFrame()
95 if (input.num_channels_ == 2) { in ReMixFrame()
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/external/webrtc/audio/
Dremix_resample.cc25 src_frame.num_channels_, src_frame.sample_rate_hz_, in RemixAndResample()
44 if (num_channels > dst_frame->num_channels_) { in RemixAndResample()
47 RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2) in RemixAndResample()
48 << "dst_frame->num_channels_: " << dst_frame->num_channels_; in RemixAndResample()
51 src_data, num_channels, samples_per_channel, dst_frame->num_channels_, in RemixAndResample()
54 audio_ptr_num_channels = dst_frame->num_channels_; in RemixAndResample()
80 if (num_channels == 1 && dst_frame->num_channels_ == 2) { in RemixAndResample()
83 dst_frame->num_channels_ = 1; in RemixAndResample()
/external/webrtc/modules/audio_processing/ns/
Dnoise_suppressor.cc259 num_channels_(num_channels), in NoiseSuppressor()
261 filter_bank_states_heap_(NumChannelsOnHeap(num_channels_)), in NoiseSuppressor()
262 upper_band_gains_heap_(NumChannelsOnHeap(num_channels_)), in NoiseSuppressor()
263 energies_before_filtering_heap_(NumChannelsOnHeap(num_channels_)), in NoiseSuppressor()
264 gain_adjustments_heap_(NumChannelsOnHeap(num_channels_)), in NoiseSuppressor()
265 channels_(num_channels_) { in NoiseSuppressor()
266 for (size_t ch = 0; ch < num_channels_; ++ch) { in NoiseSuppressor()
278 for (size_t ch = 1; ch < num_channels_; ++ch) { in AggregateWienerFilters()
290 for (size_t ch = 0; ch < num_channels_; ++ch) { in Analyze()
296 for (size_t ch = 0; ch < num_channels_; ++ch) { in Analyze()
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/external/webrtc/modules/audio_processing/aec3/
Dalignment_mixer.cc53 : num_channels_(num_channels), in AlignmentMixer()
54 one_by_num_channels_(1.f / num_channels_), in AlignmentMixer()
58 ChooseMixingVariant(downmix, adaptive_selection, num_channels_)) { in AlignmentMixer()
61 cumulative_energies_.resize(num_channels_); in AlignmentMixer()
68 RTC_DCHECK_EQ(x.size(), num_channels_); in ProduceOutput()
82 RTC_DCHECK_EQ(x.size(), num_channels_); in Downmix()
83 RTC_DCHECK_GE(num_channels_, 2); in Downmix()
85 for (size_t ch = 1; ch < num_channels_; ++ch) { in Downmix()
97 RTC_DCHECK_EQ(x.size(), num_channels_); in SelectChannel()
98 RTC_DCHECK_GE(num_channels_, 2); in SelectChannel()
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/external/webrtc/modules/audio_processing/
Daudio_buffer.cc65 num_channels_(buffer_num_channels), in AudioBuffer()
119 const bool downmix_needed = input_num_channels_ > 1 && num_channels_ == 1; in CopyFrom()
150 for (size_t i = 0; i < num_channels_; ++i) { in CopyFrom()
158 for (size_t i = 0; i < num_channels_; ++i) { in CopyFrom()
172 for (size_t i = 0; i < num_channels_; ++i) { in CopyTo()
179 for (size_t i = 0; i < num_channels_; ++i) { in CopyTo()
185 for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) { in CopyTo()
196 for (size_t i = 0; i < num_channels_; ++i) { in CopyTo()
202 for (size_t i = 0; i < num_channels_; ++i) { in CopyTo()
208 for (size_t i = num_channels_; i < buffer->num_channels(); ++i) { in CopyTo()
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/external/tensorflow/tensorflow/lite/kernels/internal/
Dmfcc_mel_filterbank.cc44 num_channels_ = output_channel_count; in Initialize()
48 if (num_channels_ < 1) { in Initialize()
76 center_frequencies_.resize(num_channels_ + 1); in Initialize()
80 const double mel_spacing = mel_span / static_cast<double>(num_channels_ + 1); in Initialize()
81 for (int i = 0; i < num_channels_ + 1; ++i) { in Initialize()
102 while ((channel < num_channels_) && in Initialize()
135 for (int c = 0; c < num_channels_; ++c) { in Initialize()
185 output->assign(num_channels_, 0.0); in Compute()
194 if (channel < num_channels_) in Compute()
/external/tensorflow/tensorflow/core/kernels/
Dmfcc_mel_filterbank.cc45 num_channels_ = output_channel_count; in Initialize()
49 if (num_channels_ < 1) { in Initialize()
77 center_frequencies_.resize(num_channels_ + 1); in Initialize()
81 const double mel_spacing = mel_span / static_cast<double>(num_channels_ + 1); in Initialize()
82 for (int i = 0; i < num_channels_ + 1; ++i) { in Initialize()
103 while ((channel < num_channels_) && in Initialize()
136 for (int c = 0; c < num_channels_; ++c) { in Initialize()
184 output->assign(num_channels_, 0.0); in Compute()
193 if (channel < num_channels_) in Compute()
/external/webrtc/modules/audio_processing/include/
Daudio_frame_view.h29 num_channels_(num_channels), in AudioFrameView()
37 num_channels_(other.num_channels()), in AudioFrameView()
42 size_t num_channels() const { return num_channels_; } in num_channels()
48 RTC_DCHECK_LE(idx, num_channels_); in channel()
54 RTC_DCHECK_LE(idx, num_channels_); in channel()
62 size_t num_channels_; variable
Daudio_frame_proxies.cc23 StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_, in ProcessAudioFrame()
25 StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_, in ProcessAudioFrame()
56 if (frame->num_channels_ <= 0) { in ProcessReverseAudioFrame()
60 StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_, in ProcessReverseAudioFrame()
62 StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_, in ProcessReverseAudioFrame()
/external/webrtc/modules/audio_processing/transient/
Dtransient_suppressor_impl.cc55 num_channels_(0), in TransientSuppressorImpl()
112 num_channels_ = num_channels; in Initialize()
113 in_buffer_.reset(new float[analysis_length_ * num_channels_]); in Initialize()
115 analysis_length_ * num_channels_ * sizeof(in_buffer_[0])); in Initialize()
120 out_buffer_.reset(new float[analysis_length_ * num_channels_]); in Initialize()
122 analysis_length_ * num_channels_ * sizeof(out_buffer_[0])); in Initialize()
129 spectral_mean_.reset(new float[complex_analysis_length_ * num_channels_]); in Initialize()
131 complex_analysis_length_ * num_channels_ * sizeof(spectral_mean_[0])); in Initialize()
170 if (!data || data_length != data_length_ || num_channels != num_channels_ || in Suppress()
206 for (int i = 0; i < num_channels_; ++i) { in Suppress()
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/external/webrtc/api/audio/
Daudio_frame.cc34 swap(a.num_channels_, b.num_channels_); in swap()
40 const size_t length_a = a.samples_per_channel_ * a.num_channels_; in swap()
41 const size_t length_b = b.samples_per_channel_ * b.num_channels_; in swap()
62 num_channels_ = 0; in ResetWithoutMuting()
83 num_channels_ = num_channels; in UpdateFrame()
112 num_channels_ = src.num_channels_; in CopyFrom()
116 const size_t length = samples_per_channel_ * num_channels_; in CopyFrom()
/external/webrtc/modules/audio_device/include/
Dtest_audio_device.cc222 num_channels_(num_channels) { in PulsedNoiseCapturerImpl()
228 int NumChannels() const override { return num_channels_; } in NumChannels()
239 num_channels_, in Capture()
264 const int num_channels_; member in webrtc::__anon9ec0e82d0111::PulsedNoiseCapturerImpl
280 int NumChannels() const override { return num_channels_; } in NumChannels()
285 num_channels_, in Capture()
308 num_channels_(num_channels), in WavFileReader()
316 const int num_channels_; member in webrtc::__anon9ec0e82d0111::WavFileReader
334 int NumChannels() const override { return num_channels_; } in NumChannels()
347 num_channels_(num_channels) {} in WavFileWriter()
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/external/openscreen/cast/standalone_sender/
Dstreaming_opus_encoder.cc35 : num_channels_(num_channels), in StreamingOpusEncoder()
40 encoder_storage_(new uint8_t[opus_encoder_get_size(num_channels_)]), in StreamingOpusEncoder()
41 input_(new float[num_channels_ * samples_per_cast_frame_]), in StreamingOpusEncoder()
52 encoder(), sample_rate(), num_channels_, OPUS_APPLICATION_AUDIO); in StreamingOpusEncoder()
78 const opus_int32 bitrate = kTransparentBitrate * num_channels_ / 2; in UseHighQuality()
97 interleaved_samples += num_channels_ * samples_copied; in EncodeAndSend()
210 interleaved_samples + num_channels_ * samples_to_copy, in FillInputBuffer()
211 input_.get() + num_channels_ * num_samples_queued_); in FillInputBuffer()

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