d94659dc27
This is the first public upload of the new NetEq, version 4. It has been through extensive internal review during the course of the project. TEST=trybots Review URL: https://webrtc-codereview.appspot.com/1073005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
101 lines
4.1 KiB
C++
101 lines
4.1 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/neteq4/preemptive_expand.h"
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#include <algorithm> // min, max
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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namespace webrtc {
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PreemptiveExpand::ReturnCodes PreemptiveExpand::Process(
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const int16_t* input,
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int input_length,
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int old_data_length,
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AudioMultiVector<int16_t>* output,
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int16_t* length_change_samples) {
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old_data_length_per_channel_ = old_data_length;
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// Input length must be (almost) 30 ms.
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// Also, the new part must be at least |overlap_samples_| elements.
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static const int k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate.
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if (num_channels_ == 0 ||
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input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ ||
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old_data_length >= input_length / num_channels_ - overlap_samples_) {
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// Length of input data too short to do preemptive expand. Simply move all
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// data from input to output.
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output->PushBackInterleaved(input, input_length);
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return kError;
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}
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return TimeStretch::Process(input, input_length, output,
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length_change_samples);
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}
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void PreemptiveExpand::SetParametersForPassiveSpeech(int len,
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int16_t* best_correlation,
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int* peak_index) const {
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// When the signal does not contain any active speech, the correlation does
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// not matter. Simply set it to zero.
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*best_correlation = 0;
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// For low energy expansion, the new data can be less than 15 ms,
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// but we must ensure that best_correlation is not larger than the length of
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// the new data.
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// but we must ensure that best_correlation is not larger than the new data.
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*peak_index = std::min(*peak_index, len - old_data_length_per_channel_);
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}
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PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
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const WebRtc_Word16 *input, int input_length, size_t peak_index,
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int16_t best_correlation, bool active_speech,
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AudioMultiVector<int16_t>* output) const {
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// Pre-calculate common multiplication with |fs_mult_|.
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// 120 corresponds to 15 ms.
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int fs_mult_120 = fs_mult_ * 120;
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assert(old_data_length_per_channel_ >= 0); // Make sure it's been set.
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// Check for strong correlation (>0.9 in Q14) and at least 15 ms new data,
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// or passive speech.
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if (((best_correlation > kCorrelationThreshold) &&
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(old_data_length_per_channel_ <= fs_mult_120)) ||
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!active_speech) {
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// Do accelerate operation by overlap add.
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// Set length of the first part, not to be modified.
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size_t unmodified_length = std::max(old_data_length_per_channel_,
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fs_mult_120);
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// Copy first part, including cross-fade region.
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output->PushBackInterleaved(
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input, (unmodified_length + peak_index) * num_channels_);
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// Copy the last |peak_index| samples up to 15 ms to |temp_vector|.
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AudioMultiVector<int16_t> temp_vector(num_channels_);
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temp_vector.PushBackInterleaved(
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&input[(unmodified_length - peak_index) * num_channels_],
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peak_index * num_channels_);
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// Cross-fade |temp_vector| onto the end of |output|.
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output->CrossFade(temp_vector, peak_index);
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// Copy the last unmodified part, 15 ms + pitch period until the end.
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output->PushBackInterleaved(
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&input[unmodified_length * num_channels_],
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input_length - unmodified_length * num_channels_);
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if (active_speech) {
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return kSuccess;
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} else {
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return kSuccessLowEnergy;
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}
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} else {
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// Accelerate not allowed. Simply move all data from decoded to outData.
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output->PushBackInterleaved(input, input_length);
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return kNoStretch;
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}
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}
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} // namespace webrtc
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