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