/* * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #include "modules/audio_coding/neteq/preemptive_expand.h" #include #include "api/array_view.h" #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/time_stretch.h" namespace webrtc { PreemptiveExpand::ReturnCodes PreemptiveExpand::Process( const int16_t* input, size_t input_length, size_t old_data_length, AudioMultiVector* output, size_t* length_change_samples) { old_data_length_per_channel_ = old_data_length; // Input length must be (almost) 30 ms. // Also, the new part must be at least |overlap_samples_| elements. static const size_t k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate. if (num_channels_ == 0 || input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ || old_data_length >= input_length / num_channels_ - overlap_samples_) { // Length of input data too short to do preemptive expand. Simply move all // data from input to output. output->PushBackInterleaved( rtc::ArrayView(input, input_length)); return kError; } const bool kFastMode = false; // Fast mode is not available for PE Expand. return TimeStretch::Process(input, input_length, kFastMode, output, length_change_samples); } void PreemptiveExpand::SetParametersForPassiveSpeech(size_t len, int16_t* best_correlation, size_t* peak_index) const { // When the signal does not contain any active speech, the correlation does // not matter. Simply set it to zero. *best_correlation = 0; // For low energy expansion, the new data can be less than 15 ms, // but we must ensure that best_correlation is not larger than the length of // the new data. // but we must ensure that best_correlation is not larger than the new data. *peak_index = std::min(*peak_index, len - old_data_length_per_channel_); } PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch( const int16_t* input, size_t input_length, size_t peak_index, int16_t best_correlation, bool active_speech, bool /*fast_mode*/, AudioMultiVector* output) const { // Pre-calculate common multiplication with |fs_mult_|. // 120 corresponds to 15 ms. size_t fs_mult_120 = static_cast(fs_mult_ * 120); // Check for strong correlation (>0.9 in Q14) and at least 15 ms new data, // or passive speech. if (((best_correlation > kCorrelationThreshold) && (old_data_length_per_channel_ <= fs_mult_120)) || !active_speech) { // Do accelerate operation by overlap add. // Set length of the first part, not to be modified. size_t unmodified_length = std::max(old_data_length_per_channel_, fs_mult_120); // Copy first part, including cross-fade region. output->PushBackInterleaved(rtc::ArrayView( input, (unmodified_length + peak_index) * num_channels_)); // Copy the last |peak_index| samples up to 15 ms to |temp_vector|. AudioMultiVector temp_vector(num_channels_); temp_vector.PushBackInterleaved(rtc::ArrayView( &input[(unmodified_length - peak_index) * num_channels_], peak_index * num_channels_)); // Cross-fade |temp_vector| onto the end of |output|. output->CrossFade(temp_vector, peak_index); // Copy the last unmodified part, 15 ms + pitch period until the end. output->PushBackInterleaved(rtc::ArrayView( &input[unmodified_length * num_channels_], input_length - unmodified_length * num_channels_)); if (active_speech) { return kSuccess; } else { return kSuccessLowEnergy; } } else { // Accelerate not allowed. Simply move all data from decoded to outData. output->PushBackInterleaved( rtc::ArrayView(input, input_length)); return kNoStretch; } } PreemptiveExpand* PreemptiveExpandFactory::Create( int sample_rate_hz, size_t num_channels, const BackgroundNoise& background_noise, size_t overlap_samples) const { return new PreemptiveExpand(sample_rate_hz, num_channels, background_noise, overlap_samples); } } // namespace webrtc