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d83a3d71bc
Merge in RedPhone // FREEBIE
89 lines
3.4 KiB
C++
89 lines
3.4 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/neteq/accelerate.h"
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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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Accelerate::ReturnCodes Accelerate::Process(
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const int16_t* input,
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size_t input_length,
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AudioMultiVector* output,
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int16_t* length_change_samples) {
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// Input length must be (almost) 30 ms.
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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 || static_cast<int>(input_length) / num_channels_ <
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(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(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 Accelerate::SetParametersForPassiveSpeech(size_t /*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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}
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Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
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const int16_t* input, size_t input_length, size_t peak_index,
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int16_t best_correlation, bool active_speech,
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AudioMultiVector* output) const {
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// Check for strong correlation or passive speech.
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if ((best_correlation > kCorrelationThreshold) || !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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assert(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(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(&input[fs_mult_120 * 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[(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(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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