/external/webrtc/webrtc/modules/audio_coding/neteq/ |
H A D | audio_multi_vector.cc | 27 num_channels_ = N; 36 num_channels_ = N; 48 for (size_t i = 0; i < num_channels_; ++i) { 54 for (size_t i = 0; i < num_channels_; ++i) { 62 for (size_t i = 0; i < num_channels_; ++i) { 70 assert(length % num_channels_ == 0); 71 if (num_channels_ == 1) { 76 size_t length_per_channel = length / num_channels_; 78 for (size_t channel = 0; channel < num_channels_; ++channel) { 84 source_ptr += num_channels_; // Jum [all...] |
H A D | audio_multi_vector_unittest.cc | 34 : num_channels_(GetParam()), // Get the test parameter. 35 interleaved_length_(num_channels_ * array_length()) { 36 array_interleaved_ = new int16_t[num_channels_ * array_length()]; 53 for (size_t j = 1; j <= num_channels_; ++j) { 64 const size_t num_channels_; member in class:webrtc::AudioMultiVectorTest 73 AudioMultiVector vec1(num_channels_); 75 EXPECT_EQ(num_channels_, vec1.Channels()); 79 AudioMultiVector vec2(num_channels_, initial_size); 81 EXPECT_EQ(num_channels_, vec2.Channels()); 87 AudioMultiVector vec(num_channels_, array_lengt [all...] |
H A D | accelerate.cc | 24 if (num_channels_ == 0 || 25 input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) { 70 output->PushBackInterleaved(input, fs_mult_120 * num_channels_); 72 AudioMultiVector temp_vector(num_channels_); 73 temp_vector.PushBackInterleaved(&input[fs_mult_120 * num_channels_], 74 peak_index * num_channels_); 79 &input[(fs_mult_120 + peak_index) * num_channels_], 80 input_length - (fs_mult_120 + peak_index) * num_channels_);
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H A D | preemptive_expand.cc | 29 if (num_channels_ == 0 || 30 input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ || 31 old_data_length >= input_length / num_channels_ - overlap_samples_) { 80 input, (unmodified_length + peak_index) * num_channels_); 82 AudioMultiVector temp_vector(num_channels_); 84 &input[(unmodified_length - peak_index) * num_channels_], 85 peak_index * num_channels_); 90 &input[unmodified_length * num_channels_], 91 input_length - unmodified_length * num_channels_);
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H A D | time_stretch.h | 42 num_channels_(num_channels), 50 assert(num_channels_ > 0); 51 assert(master_channel_ < num_channels_); 94 const size_t num_channels_;
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H A D | expand_unittest.cc | 74 num_channels_(1), 75 background_noise_(num_channels_), 76 sync_buffer_(num_channels_, 83 num_channels_) { 98 ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels."; 103 size_t num_channels_; member in class:webrtc::ExpandTest 116 AudioMultiVector output(num_channels_); 136 AudioMultiVector output(num_channels_); 153 AudioMultiVector output(num_channels_);
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H A D | background_noise.cc | 28 : num_channels_(num_channels), 29 channel_parameters_(new ChannelParameters[num_channels_]), 38 for (size_t channel = 0; channel < num_channels_; ++channel) { 57 for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) { 129 assert(channel < num_channels_); 134 assert(channel < num_channels_); 139 assert(channel < num_channels_); 144 assert(channel < num_channels_); 149 assert(channel < num_channels_); 155 assert(channel < num_channels_); [all...] |
H A D | neteq_stereo_unittest.cc | 50 : num_channels_(GetParam().num_channels), 69 input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_]; 71 num_channels_]; 72 output_multi_channel_ = new int16_t[kMaxBlockSize * num_channels_]; 94 if (num_channels_ == 2) { 96 } else if (num_channels_ == 5) { 104 if (num_channels_ == 2) { 112 if (num_channels_ == 2) { 120 if (num_channels_ == 2) { 151 num_channels_, 242 const size_t num_channels_; member in class:webrtc::NetEqStereoTest [all...] |
/external/webrtc/webrtc/common_audio/resampler/ |
H A D | push_resampler.cc | 25 num_channels_(0) { 38 num_channels == num_channels_) 48 num_channels_ = num_channels; 56 if (num_channels_ == 2) { 71 const size_t src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100; 72 const size_t dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100; 82 if (num_channels_ == 2) { 83 const size_t src_length_mono = src_length / num_channels_; 84 const size_t dst_capacity_mono = dst_capacity / num_channels_; 86 Deinterleave(src, src_length_mono, num_channels_, deinterleave [all...] |
/external/webrtc/webrtc/modules/utility/source/ |
H A D | audio_frame_operations_unittest.cc | 24 frame_.num_channels_ = 2; 44 EXPECT_EQ(frame1.num_channels_, frame2.num_channels_); 48 for (size_t i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_; 58 frame_.num_channels_ = 1; 63 frame_.num_channels_ = 1; 71 stereo_frame.num_channels_ = 2; 79 frame_.num_channels_ = 2; // Need to set manually. 84 frame_.num_channels_ = 1; 96 mono_frame.num_channels_ [all...] |
H A D | audio_frame_operations.cc | 26 if (frame->num_channels_ != 1) { 38 frame->num_channels_ = 2; 52 if (frame->num_channels_ != 2) { 57 frame->num_channels_ = 1; 63 if (frame->num_channels_ != 2) return; 74 frame.samples_per_channel_ * frame.num_channels_); 78 if (frame.num_channels_ != 2) { 95 for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
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H A D | file_recorder_impl.cc | 144 if( incomingAudioFrame.num_channels_ == 2 && 148 tempAudioFrame.num_channels_ = 1; 162 else if( incomingAudioFrame.num_channels_ == 1 && 166 tempAudioFrame.num_channels_ = 2; 209 ptrAudioFrame->num_channels_); 212 ptrAudioFrame->num_channels_,
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/external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
H A D | audio_encoder_g722.cc | 40 : num_channels_(config.num_channels), 46 encoders_(new EncoderState[num_channels_]), 47 interleave_buffer_(2 * num_channels_) { 51 for (size_t i = 0; i < num_channels_; ++i) { 64 return SamplesPerChannel() / 2 * num_channels_; 72 return num_channels_; 107 for (size_t j = 0; j < num_channels_; ++j) 108 encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j]; 119 for (size_t i = 0; i < num_channels_; ++i) { 130 for (size_t j = 0; j < num_channels_; [all...] |
/external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
H A D | audio_decoder_pcm16b.cc | 19 : num_channels_(num_channels) { 26 return num_channels_;
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H A D | audio_decoder_pcm16b.h | 34 const size_t num_channels_; member in class:webrtc::final
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/external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
H A D | audio_decoder_pcm.h | 21 explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) { 36 const size_t num_channels_; member in class:webrtc::final 42 explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) { 57 const size_t num_channels_; member in class:webrtc::final
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H A D | audio_decoder_pcm.cc | 20 return num_channels_; 44 return num_channels_;
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/external/webrtc/webrtc/common_audio/ |
H A D | channel_buffer.h | 50 num_channels_(num_channels), 52 for (size_t i = 0; i < num_channels_; ++i) { 54 channels_[j * num_channels_ + i] = 56 bands_[i * num_bands_ + j] = channels_[j * num_channels_ + i]; 65 // 0 <= channel < |num_channels_| 75 // 0 <= channel < |num_channels_| 79 return &channels_[band * num_channels_]; 90 // 0 <= channel < |num_channels_| 94 RTC_DCHECK_LT(channel, num_channels_); 107 for (size_t i = 0; i < num_channels_; 133 const size_t num_channels_; member in class:webrtc::ChannelBuffer [all...] |
H A D | wav_file.h | 54 size_t num_channels() const override { return num_channels_; } 60 const size_t num_channels_; member in class:webrtc::final 82 size_t num_channels() const override { return num_channels_; } 88 size_t num_channels_; member in class:webrtc::final
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/external/webrtc/webrtc/modules/include/ |
H A D | module_common_types.h | 475 * samples_per_channel_ * num_channels_ 535 size_t num_channels_; member in class:webrtc::AudioFrame 563 num_channels_ = 0; 585 num_channels_ = num_channels; 608 num_channels_ = src.num_channels_; 612 const size_t length = samples_per_channel_ * num_channels_; 618 memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t)); 622 assert((num_channels_ > 0) && (num_channels_ < [all...] |
/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | acm_send_test_oldapi.cc | 43 input_frame_.num_channels_ = 1; 45 assert(input_block_size_samples_ * input_frame_.num_channels_ <= 61 input_frame_.num_channels_ = channels; 62 assert(input_block_size_samples_ * input_frame_.num_channels_ <= 70 input_frame_.num_channels_ = external_speech_encoder->NumChannels(); 71 assert(input_block_size_samples_ * input_frame_.num_channels_ <= 89 if (input_frame_.num_channels_ > 1) { 92 input_frame_.num_channels_,
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/external/webrtc/webrtc/modules/audio_processing/ |
H A D | audio_processing_impl_unittest.cc | 46 frame.num_channels_ = 1; 60 frame.num_channels_ = 2; 64 // ProcessStream sets num_channels_ == num_output_channels. 65 frame.num_channels_ = 2;
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H A D | audio_buffer.cc | 56 num_channels_(num_process_channels), 153 assert(stream_config.num_channels() == num_channels_ || num_channels_ == 1); 161 for (size_t i = 0; i < num_channels_; ++i) { 169 for (size_t i = 0; i < num_channels_; ++i) { 178 for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) { 188 num_channels_ = num_proc_channels_; 317 num_split_frames_, num_channels_, local 345 return num_channels_; 349 num_channels_ [all...] |
/external/webrtc/webrtc/modules/audio_processing/transient/ |
H A D | transient_suppressor.cc | 53 num_channels_(0), 111 num_channels_ = num_channels; 112 in_buffer_.reset(new float[analysis_length_ * num_channels_]); 115 analysis_length_ * num_channels_ * sizeof(in_buffer_[0])); 121 out_buffer_.reset(new float[analysis_length_ * num_channels_]); 124 analysis_length_ * num_channels_ * sizeof(out_buffer_[0])); 131 spectral_mean_.reset(new float[complex_analysis_length_ * num_channels_]); 134 complex_analysis_length_ * num_channels_ * sizeof(spectral_mean_[0])); 174 if (!data || data_length != data_length_ || num_channels != num_channels_ || 210 for (int i = 0; i < num_channels_; [all...] |
/external/webrtc/webrtc/voice_engine/ |
H A D | utility.cc | 28 src_frame.num_channels_, src_frame.sample_rate_hz_, 46 if (num_channels == 2 && dst_frame->num_channels_ == 1) { 74 if (num_channels == 1 && dst_frame->num_channels_ == 2) { 77 dst_frame->num_channels_ = 1;
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