mirror of
https://github.com/oxen-io/session-android.git
synced 2024-12-21 07:27:30 +00:00
89 lines
3.4 KiB
C++
89 lines
3.4 KiB
C++
/*
|
|
* 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 "webrtc/modules/audio_coding/neteq/accelerate.h"
|
|
|
|
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
|
|
|
|
namespace webrtc {
|
|
|
|
Accelerate::ReturnCodes Accelerate::Process(
|
|
const int16_t* input,
|
|
size_t input_length,
|
|
AudioMultiVector* output,
|
|
int16_t* length_change_samples) {
|
|
// Input length must be (almost) 30 ms.
|
|
static const int k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate.
|
|
if (num_channels_ == 0 || static_cast<int>(input_length) / num_channels_ <
|
|
(2 * k15ms - 1) * fs_mult_) {
|
|
// Length of input data too short to do accelerate. Simply move all data
|
|
// from input to output.
|
|
output->PushBackInterleaved(input, input_length);
|
|
return kError;
|
|
}
|
|
return TimeStretch::Process(input, input_length, output,
|
|
length_change_samples);
|
|
}
|
|
|
|
void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/,
|
|
int16_t* best_correlation,
|
|
int* /*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;
|
|
}
|
|
|
|
Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
|
|
const int16_t* input, size_t input_length, size_t peak_index,
|
|
int16_t best_correlation, bool active_speech,
|
|
AudioMultiVector* output) const {
|
|
// Check for strong correlation or passive speech.
|
|
if ((best_correlation > kCorrelationThreshold) || !active_speech) {
|
|
// Do accelerate operation by overlap add.
|
|
|
|
// Pre-calculate common multiplication with |fs_mult_|.
|
|
// 120 corresponds to 15 ms.
|
|
size_t fs_mult_120 = fs_mult_ * 120;
|
|
|
|
assert(fs_mult_120 >= peak_index); // Should be handled in Process().
|
|
// Copy first part; 0 to 15 ms.
|
|
output->PushBackInterleaved(input, fs_mult_120 * num_channels_);
|
|
// Copy the |peak_index| starting at 15 ms to |temp_vector|.
|
|
AudioMultiVector temp_vector(num_channels_);
|
|
temp_vector.PushBackInterleaved(&input[fs_mult_120 * 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(
|
|
&input[(fs_mult_120 + peak_index) * num_channels_],
|
|
input_length - (fs_mult_120 + peak_index) * num_channels_);
|
|
|
|
if (active_speech) {
|
|
return kSuccess;
|
|
} else {
|
|
return kSuccessLowEnergy;
|
|
}
|
|
} else {
|
|
// Accelerate not allowed. Simply move all data from decoded to outData.
|
|
output->PushBackInterleaved(input, input_length);
|
|
return kNoStretch;
|
|
}
|
|
}
|
|
|
|
Accelerate* AccelerateFactory::Create(
|
|
int sample_rate_hz,
|
|
size_t num_channels,
|
|
const BackgroundNoise& background_noise) const {
|
|
return new Accelerate(sample_rate_hz, num_channels, background_noise);
|
|
}
|
|
|
|
} // namespace webrtc
|