/external/webrtc/webrtc/modules/audio_coding/neteq/ |
D | audio_multi_vector.cc | 27 num_channels_ = N; in AudioMultiVector() 36 num_channels_ = N; in AudioMultiVector() 48 for (size_t i = 0; i < num_channels_; ++i) { in Clear() 54 for (size_t i = 0; i < num_channels_; ++i) { in Zeros() 62 for (size_t i = 0; i < num_channels_; ++i) { in CopyTo() 70 assert(length % num_channels_ == 0); in PushBackInterleaved() 71 if (num_channels_ == 1) { in PushBackInterleaved() 76 size_t length_per_channel = length / num_channels_; in PushBackInterleaved() 78 for (size_t channel = 0; channel < num_channels_; ++channel) { in PushBackInterleaved() 84 source_ptr += num_channels_; // Jump to next element of this channel. in PushBackInterleaved() [all …]
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D | audio_multi_vector_unittest.cc | 34 : num_channels_(GetParam()), // Get the test parameter. in AudioMultiVectorTest() 35 interleaved_length_(num_channels_ * array_length()) { in AudioMultiVectorTest() 36 array_interleaved_ = new int16_t[num_channels_ * array_length()]; in AudioMultiVectorTest() 53 for (size_t j = 1; j <= num_channels_; ++j) { in SetUp() 64 const size_t num_channels_; member in webrtc::AudioMultiVectorTest 73 AudioMultiVector vec1(num_channels_); in TEST_P() 75 EXPECT_EQ(num_channels_, vec1.Channels()); in TEST_P() 79 AudioMultiVector vec2(num_channels_, initial_size); in TEST_P() 81 EXPECT_EQ(num_channels_, vec2.Channels()); in TEST_P() 87 AudioMultiVector vec(num_channels_, array_length()); in TEST_P() [all …]
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D | preemptive_expand.cc | 29 if (num_channels_ == 0 || in Process() 30 input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ || in Process() 31 old_data_length >= input_length / num_channels_ - overlap_samples_) { in Process() 80 input, (unmodified_length + peak_index) * num_channels_); in CheckCriteriaAndStretch() 82 AudioMultiVector temp_vector(num_channels_); in CheckCriteriaAndStretch() 84 &input[(unmodified_length - peak_index) * num_channels_], in CheckCriteriaAndStretch() 85 peak_index * num_channels_); in CheckCriteriaAndStretch() 90 &input[unmodified_length * num_channels_], in CheckCriteriaAndStretch() 91 input_length - unmodified_length * num_channels_); in CheckCriteriaAndStretch()
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D | background_noise.cc | 28 : num_channels_(num_channels), in BackgroundNoise() 29 channel_parameters_(new ChannelParameters[num_channels_]), in BackgroundNoise() 38 for (size_t channel = 0; channel < num_channels_; ++channel) { in Reset() 57 for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) { in Update() 129 assert(channel < num_channels_); in Energy() 134 assert(channel < num_channels_); in SetMuteFactor() 139 assert(channel < num_channels_); in MuteFactor() 144 assert(channel < num_channels_); in Filter() 149 assert(channel < num_channels_); in FilterState() 155 assert(channel < num_channels_); in SetFilterState() [all …]
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D | neteq_stereo_unittest.cc | 50 : num_channels_(GetParam().num_channels), in NetEqStereoTest() 69 input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_]; in NetEqStereoTest() 71 num_channels_]; in NetEqStereoTest() 72 output_multi_channel_ = new int16_t[kMaxBlockSize * num_channels_]; in NetEqStereoTest() 94 if (num_channels_ == 2) { in SetUp() 96 } else if (num_channels_ == 5) { in SetUp() 104 if (num_channels_ == 2) { in SetUp() 112 if (num_channels_ == 2) { in SetUp() 120 if (num_channels_ == 2) { in SetUp() 151 num_channels_, in GetNewPackets() [all …]
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D | accelerate.cc | 24 if (num_channels_ == 0 || in Process() 25 input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) { in Process() 70 output->PushBackInterleaved(input, fs_mult_120 * num_channels_); in CheckCriteriaAndStretch() 72 AudioMultiVector temp_vector(num_channels_); in CheckCriteriaAndStretch() 73 temp_vector.PushBackInterleaved(&input[fs_mult_120 * num_channels_], in CheckCriteriaAndStretch() 74 peak_index * num_channels_); in CheckCriteriaAndStretch() 79 &input[(fs_mult_120 + peak_index) * num_channels_], in CheckCriteriaAndStretch() 80 input_length - (fs_mult_120 + peak_index) * num_channels_); in CheckCriteriaAndStretch()
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D | expand_unittest.cc | 74 num_channels_(1), in ExpandTest() 75 background_noise_(num_channels_), in ExpandTest() 76 sync_buffer_(num_channels_, in ExpandTest() 83 num_channels_) { in ExpandTest() 98 ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels."; in SetUp() 103 size_t num_channels_; member in webrtc::ExpandTest 116 AudioMultiVector output(num_channels_); in TEST_F() 136 AudioMultiVector output(num_channels_); in TEST_F() 153 AudioMultiVector output(num_channels_); in TEST_F()
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D | time_stretch.h | 42 num_channels_(num_channels), in TimeStretch() 50 assert(num_channels_ > 0); in TimeStretch() 51 assert(master_channel_ < num_channels_); in TimeStretch() 94 const size_t num_channels_; variable
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D | merge.cc | 32 num_channels_(num_channels), in Merge() 37 expanded_(num_channels_) { in Merge() 38 assert(num_channels_ > 0); in Merge() 55 AudioMultiVector input_vector(num_channels_); in Process() 58 assert(input_length_per_channel == input_length / num_channels_); in Process() 63 for (size_t channel = 0; channel < num_channels_; ++channel) { in Process() 182 AudioMultiVector expanded_temp(num_channels_); in GetExpandedSignal() 379 return fs_hz_ / 100 * num_channels_; // 10 ms. in RequiredFutureSamples()
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/external/webrtc/webrtc/modules/utility/source/ |
D | audio_frame_operations_unittest.cc | 24 frame_.num_channels_ = 2; in AudioFrameOperationsTest() 44 EXPECT_EQ(frame1.num_channels_, frame2.num_channels_); in VerifyFramesAreEqual() 48 for (size_t i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_; in VerifyFramesAreEqual() 58 frame_.num_channels_ = 1; in TEST_F() 63 frame_.num_channels_ = 1; in TEST_F() 71 stereo_frame.num_channels_ = 2; in TEST_F() 79 frame_.num_channels_ = 2; // Need to set manually. in TEST_F() 84 frame_.num_channels_ = 1; in TEST_F() 96 mono_frame.num_channels_ = 1; in TEST_F() 104 frame_.num_channels_ = 1; // Need to set manually. in TEST_F() [all …]
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D | audio_frame_operations.cc | 26 if (frame->num_channels_ != 1) { in MonoToStereo() 38 frame->num_channels_ = 2; in MonoToStereo() 52 if (frame->num_channels_ != 2) { in StereoToMono() 57 frame->num_channels_ = 1; in StereoToMono() 63 if (frame->num_channels_ != 2) return; in SwapStereoChannels() 74 frame.samples_per_channel_ * frame.num_channels_); in Mute() 78 if (frame.num_channels_ != 2) { in Scale() 95 for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_; in ScaleWithSat()
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D | file_recorder_impl.cc | 144 if( incomingAudioFrame.num_channels_ == 2 && in RecordAudioToFile() 148 tempAudioFrame.num_channels_ = 1; in RecordAudioToFile() 162 else if( incomingAudioFrame.num_channels_ == 1 && in RecordAudioToFile() 166 tempAudioFrame.num_channels_ = 2; in RecordAudioToFile() 209 ptrAudioFrame->num_channels_); in RecordAudioToFile() 212 ptrAudioFrame->num_channels_, in RecordAudioToFile()
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/external/webrtc/webrtc/common_audio/resampler/ |
D | push_resampler.cc | 25 num_channels_(0) { in PushResampler() 38 num_channels == num_channels_) in InitializeIfNeeded() 48 num_channels_ = num_channels; in InitializeIfNeeded() 56 if (num_channels_ == 2) { in InitializeIfNeeded() 71 const size_t src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100; in Resample() 72 const size_t dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100; in Resample() 82 if (num_channels_ == 2) { in Resample() 83 const size_t src_length_mono = src_length / num_channels_; in Resample() 84 const size_t dst_capacity_mono = dst_capacity / num_channels_; in Resample() 86 Deinterleave(src, src_length_mono, num_channels_, deinterleaved); in Resample() [all …]
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/external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
D | audio_encoder_g722.cc | 40 : num_channels_(config.num_channels), in AudioEncoderG722() 46 encoders_(new EncoderState[num_channels_]), in AudioEncoderG722() 47 interleave_buffer_(2 * num_channels_) { in AudioEncoderG722() 51 for (size_t i = 0; i < num_channels_; ++i) { in AudioEncoderG722() 64 return SamplesPerChannel() / 2 * num_channels_; in MaxEncodedBytes() 72 return num_channels_; in NumChannels() 107 for (size_t j = 0; j < num_channels_; ++j) in EncodeInternal() 108 encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j]; in EncodeInternal() 119 for (size_t i = 0; i < num_channels_; ++i) { in EncodeInternal() 130 for (size_t j = 0; j < num_channels_; ++j) { in EncodeInternal() [all …]
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/external/webrtc/webrtc/modules/include/ |
D | module_common_types.h | 535 size_t num_channels_; variable 563 num_channels_ = 0; in Reset() 585 num_channels_ = num_channels; in UpdateFrame() 608 num_channels_ = src.num_channels_; in CopyFrom() 612 const size_t length = samples_per_channel_ * num_channels_; in CopyFrom() 618 memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t)); in Mute() 622 assert((num_channels_ > 0) && (num_channels_ < 3)); 623 if ((num_channels_ > 2) || (num_channels_ < 1)) return *this; 625 for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) { 633 assert((num_channels_ > 0) && (num_channels_ < 3)); in Append() [all …]
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/external/webrtc/webrtc/modules/audio_processing/ |
D | audio_buffer.cc | 56 num_channels_(num_process_channels), in AudioBuffer() 153 assert(stream_config.num_channels() == num_channels_ || num_channels_ == 1); in CopyTo() 161 for (size_t i = 0; i < num_channels_; ++i) { in CopyTo() 169 for (size_t i = 0; i < num_channels_; ++i) { in CopyTo() 178 for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) { in CopyTo() 188 num_channels_ = num_proc_channels_; in InitForNewData() 317 num_split_frames_, num_channels_, in mixed_low_pass_data() 345 return num_channels_; in num_channels() 349 num_channels_ = num_channels; in set_num_channels() 371 assert(frame->num_channels_ == num_input_channels_); in DeinterleaveFrom() [all …]
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D | audio_processing_impl_unittest.cc | 46 frame.num_channels_ = 1; in TEST() 60 frame.num_channels_ = 2; in TEST() 65 frame.num_channels_ = 2; in TEST()
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/external/webrtc/webrtc/common_audio/ |
D | channel_buffer.h | 50 num_channels_(num_channels), in data_() 52 for (size_t i = 0; i < num_channels_; ++i) { in data_() 54 channels_[j * num_channels_ + i] = in data_() 56 bands_[i * num_bands_ + j] = channels_[j * num_channels_ + i]; in data_() 79 return &channels_[band * num_channels_]; in channels() 94 RTC_DCHECK_LT(channel, num_channels_); in bands() 107 for (size_t i = 0; i < num_channels_; ++i) in Slice() 118 size_t num_channels() const { return num_channels_; } in num_channels() 120 size_t size() const {return num_frames_ * num_channels_; } in size() 133 const size_t num_channels_; variable
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D | wav_file.h | 54 size_t num_channels() const override { return num_channels_; } in num_channels() 60 const size_t num_channels_; variable 82 size_t num_channels() const override { return num_channels_; } in num_channels() 88 size_t num_channels_; variable
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/external/webrtc/webrtc/modules/audio_processing/transient/ |
D | transient_suppressor.cc | 53 num_channels_(0), in TransientSuppressor() 111 num_channels_ = num_channels; in Initialize() 112 in_buffer_.reset(new float[analysis_length_ * num_channels_]); in Initialize() 115 analysis_length_ * num_channels_ * sizeof(in_buffer_[0])); in Initialize() 121 out_buffer_.reset(new float[analysis_length_ * num_channels_]); in Initialize() 124 analysis_length_ * num_channels_ * sizeof(out_buffer_[0])); in Initialize() 131 spectral_mean_.reset(new float[complex_analysis_length_ * num_channels_]); in Initialize() 134 complex_analysis_length_ * num_channels_ * sizeof(spectral_mean_[0])); in Initialize() 174 if (!data || data_length != data_length_ || num_channels != num_channels_ || in Suppress() 210 for (int i = 0; i < num_channels_; ++i) { in Suppress() [all …]
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/external/tensorflow/tensorflow/core/kernels/ |
D | mfcc_mel_filterbank.cc | 45 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() 104 (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()
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
D | acm_send_test_oldapi.cc | 43 input_frame_.num_channels_ = 1; in AcmSendTestOldApi() 45 assert(input_block_size_samples_ * input_frame_.num_channels_ <= in AcmSendTestOldApi() 61 input_frame_.num_channels_ = channels; in RegisterCodec() 62 assert(input_block_size_samples_ * input_frame_.num_channels_ <= in RegisterCodec() 70 input_frame_.num_channels_ = external_speech_encoder->NumChannels(); in RegisterExternalCodec() 71 assert(input_block_size_samples_ * input_frame_.num_channels_ <= in RegisterExternalCodec() 89 if (input_frame_.num_channels_ > 1) { in NextPacket() 92 input_frame_.num_channels_, in NextPacket()
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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() 36 const size_t num_channels_; 42 explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) { in AudioDecoderPcmA() 57 const size_t num_channels_;
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/external/webrtc/webrtc/voice_engine/ |
D | utility_unittest.cc | 29 src_frame_.num_channels_ = 1; in UtilityTest() 50 frame->num_channels_ = 1; in SetMonoFrame() 68 frame->num_channels_ = 2; in SetStereoFrame() 83 EXPECT_EQ(ref_frame.num_channels_, test_frame.num_channels_); in VerifyParams() 100 ref_frame.num_channels_ - delay; i++) { in ComputeSNR() 121 i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) { in VerifyFramesAreEqual()
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D | utility.cc | 28 src_frame.num_channels_, src_frame.sample_rate_hz_, in RemixAndResample() 46 if (num_channels == 2 && dst_frame->num_channels_ == 1) { in RemixAndResample() 74 if (num_channels == 1 && dst_frame->num_channels_ == 2) { in RemixAndResample() 77 dst_frame->num_channels_ = 1; in RemixAndResample()
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