107 lines
4.1 KiB
C++
107 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 "modules/audio_coding/neteq/accelerate.h"
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#include "api/array_view.h"
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#include "modules/audio_coding/neteq/audio_multi_vector.h"
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namespace webrtc {
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Accelerate::ReturnCodes Accelerate::Process(const int16_t* input,
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size_t input_length,
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bool fast_accelerate,
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AudioMultiVector* output,
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size_t* length_change_samples) {
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// Input length must be (almost) 30 ms.
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static const size_t 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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// Length of input data too short to do accelerate. Simply move all data
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// from input to output.
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output->PushBackInterleaved(
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rtc::ArrayView<const int16_t>(input, input_length));
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return kError;
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}
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return TimeStretch::Process(input, input_length, fast_accelerate, output,
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length_change_samples);
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}
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void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/,
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int16_t* best_correlation,
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size_t* /*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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}
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Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
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const int16_t* input,
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size_t input_length,
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size_t peak_index,
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int16_t best_correlation,
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bool active_speech,
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bool fast_mode,
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AudioMultiVector* output) const {
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// Check for strong correlation or passive speech.
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// Use 8192 (0.5 in Q14) in fast mode.
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const int correlation_threshold = fast_mode ? 8192 : kCorrelationThreshold;
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if ((best_correlation > correlation_threshold) || !active_speech) {
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// Do accelerate operation by overlap add.
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// Pre-calculate common multiplication with `fs_mult_`.
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// 120 corresponds to 15 ms.
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size_t fs_mult_120 = fs_mult_ * 120;
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if (fast_mode) {
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// Fit as many multiples of `peak_index` as possible in fs_mult_120.
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// TODO(henrik.lundin) Consider finding multiple correlation peaks and
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// pick the one with the longest correlation lag in this case.
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peak_index = (fs_mult_120 / peak_index) * peak_index;
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}
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RTC_DCHECK_GE(fs_mult_120, peak_index); // Should be handled in Process().
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// Copy first part; 0 to 15 ms.
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output->PushBackInterleaved(
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rtc::ArrayView<const int16_t>(input, fs_mult_120 * num_channels_));
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// Copy the `peak_index` starting at 15 ms to `temp_vector`.
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AudioMultiVector temp_vector(num_channels_);
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temp_vector.PushBackInterleaved(rtc::ArrayView<const int16_t>(
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&input[fs_mult_120 * num_channels_], 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(rtc::ArrayView<const int16_t>(
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&input[(fs_mult_120 + peak_index) * num_channels_],
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input_length - (fs_mult_120 + peak_index) * 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(
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rtc::ArrayView<const int16_t>(input, input_length));
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return kNoStretch;
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}
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}
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Accelerate* AccelerateFactory::Create(
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int sample_rate_hz,
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size_t num_channels,
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const BackgroundNoise& background_noise) const {
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return new Accelerate(sample_rate_hz, num_channels, background_noise);
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}
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} // namespace webrtc
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