blob: 8b8bdb1af5eb390a27a5144362dd169a496c2030 [file] [log] [blame] [edit]
/*
* 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 "stddef.h"
#include "rtc_base/checks.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
// TODO(bjornv): Change the return type to report errors.
void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
int16_t* data_out,
const int16_t* __restrict coefficients,
size_t coefficients_length,
size_t data_length) {
size_t i = 0;
size_t j = 0;
RTC_DCHECK_GT(data_length, 0);
RTC_DCHECK_GT(coefficients_length, 1);
for (i = 0; i < data_length; i++) {
int64_t output = 0;
int64_t sum = 0;
for (j = coefficients_length - 1; j > 0; j--) {
// Negative overflow is permitted here, because this is
// auto-regressive filters, and the state for each batch run is
// stored in the "negative" positions of the output vector.
sum += coefficients[j] * data_out[(ptrdiff_t) i - (ptrdiff_t) j];
}
output = coefficients[0] * data_in[i];
output -= sum;
// Saturate and store the output.
output = WEBRTC_SPL_SAT(134215679, output, -134217728);
data_out[i] = (int16_t)((output + 2048) >> 12);
}
}