1 /* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #include "webrtc/modules/audio_coding/neteq/accelerate.h" 12 13 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" 14 15 namespace webrtc { 16 17 Accelerate::ReturnCodes Accelerate::Process( 18 const int16_t* input, 19 size_t input_length, 20 AudioMultiVector* output, 21 int16_t* length_change_samples) { 22 // Input length must be (almost) 30 ms. 23 static const int k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate. 24 if (num_channels_ == 0 || static_cast<int>(input_length) / num_channels_ < 25 (2 * k15ms - 1) * fs_mult_) { 26 // Length of input data too short to do accelerate. Simply move all data 27 // from input to output. 28 output->PushBackInterleaved(input, input_length); 29 return kError; 30 } 31 return TimeStretch::Process(input, input_length, output, 32 length_change_samples); 33 } 34 35 void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/, 36 int16_t* best_correlation, 37 int* /*peak_index*/) const { 38 // When the signal does not contain any active speech, the correlation does 39 // not matter. Simply set it to zero. 40 *best_correlation = 0; 41 } 42 43 Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch( 44 const int16_t* input, size_t input_length, size_t peak_index, 45 int16_t best_correlation, bool active_speech, 46 AudioMultiVector* output) const { 47 // Check for strong correlation or passive speech. 48 if ((best_correlation > kCorrelationThreshold) || !active_speech) { 49 // Do accelerate operation by overlap add. 50 51 // Pre-calculate common multiplication with |fs_mult_|. 52 // 120 corresponds to 15 ms. 53 size_t fs_mult_120 = fs_mult_ * 120; 54 55 assert(fs_mult_120 >= peak_index); // Should be handled in Process(). 56 // Copy first part; 0 to 15 ms. 57 output->PushBackInterleaved(input, fs_mult_120 * num_channels_); 58 // Copy the |peak_index| starting at 15 ms to |temp_vector|. 59 AudioMultiVector temp_vector(num_channels_); 60 temp_vector.PushBackInterleaved(&input[fs_mult_120 * num_channels_], 61 peak_index * num_channels_); 62 // Cross-fade |temp_vector| onto the end of |output|. 63 output->CrossFade(temp_vector, peak_index); 64 // Copy the last unmodified part, 15 ms + pitch period until the end. 65 output->PushBackInterleaved( 66 &input[(fs_mult_120 + peak_index) * num_channels_], 67 input_length - (fs_mult_120 + peak_index) * num_channels_); 68 69 if (active_speech) { 70 return kSuccess; 71 } else { 72 return kSuccessLowEnergy; 73 } 74 } else { 75 // Accelerate not allowed. Simply move all data from decoded to outData. 76 output->PushBackInterleaved(input, input_length); 77 return kNoStretch; 78 } 79 } 80 81 Accelerate* AccelerateFactory::Create( 82 int sample_rate_hz, 83 size_t num_channels, 84 const BackgroundNoise& background_noise) const { 85 return new Accelerate(sample_rate_hz, num_channels, background_noise); 86 } 87 88 } // namespace webrtc 89
