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3111783aa1
Prior to this commit, most .c files in modules/audio_coding/codecs/ilbc don't include their corresponding headers, nor do they order #includes as per the Google Style Guide [1]. The former is especially harmful, since in C it can silently allow the function signature to diverge from its prototype, thus causing disaster at runtime. This CL fixes both issues. In effect, this allows the common_audio and modules/audio_coding:ilbc targets to be compiled with Clang's -Wmissing-prototypes, though this CL does not add that change. [1]: https://google.github.io/styleguide/cppguide.html#Names_and_Order_of_Includes Bug: webrtc:12314 Change-Id: I8299968ed3cc86ff35d9de045072b846298043af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198362 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Timothy Gu <timothygu@chromium.org> Cr-Commit-Position: refs/heads/master@{#32896}
91 lines
2.9 KiB
C
91 lines
2.9 KiB
C
/*
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* Copyright (c) 2011 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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/******************************************************************
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iLBC Speech Coder ANSI-C Source Code
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WebRtcIlbcfix_HpOutput.c
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******************************************************************/
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#include "modules/audio_coding/codecs/ilbc/hp_output.h"
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#include "modules/audio_coding/codecs/ilbc/defines.h"
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/*----------------------------------------------------------------*
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* high-pass filter of output and *2 with saturation
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*---------------------------------------------------------------*/
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void WebRtcIlbcfix_HpOutput(
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int16_t *signal, /* (i/o) signal vector */
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int16_t *ba, /* (i) B- and A-coefficients (2:nd order)
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{b[0] b[1] b[2] -a[1] -a[2]} a[0]
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is assumed to be 1.0 */
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int16_t *y, /* (i/o) Filter state yhi[n-1] ylow[n-1]
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yhi[n-2] ylow[n-2] */
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int16_t *x, /* (i/o) Filter state x[n-1] x[n-2] */
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size_t len) /* (i) Number of samples to filter */
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{
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size_t i;
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int32_t tmpW32;
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int32_t tmpW32b;
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for (i=0; i<len; i++) {
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/*
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y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
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+ (-a[1])*y[i-1] + (-a[2])*y[i-2];
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*/
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tmpW32 = y[1] * ba[3]; /* (-a[1])*y[i-1] (low part) */
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tmpW32 += y[3] * ba[4]; /* (-a[2])*y[i-2] (low part) */
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tmpW32 = (tmpW32>>15);
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tmpW32 += y[0] * ba[3]; /* (-a[1])*y[i-1] (high part) */
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tmpW32 += y[2] * ba[4]; /* (-a[2])*y[i-2] (high part) */
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tmpW32 *= 2;
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tmpW32 += signal[i] * ba[0]; /* b[0]*x[0] */
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tmpW32 += x[0] * ba[1]; /* b[1]*x[i-1] */
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tmpW32 += x[1] * ba[2]; /* b[2]*x[i-2] */
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/* Update state (input part) */
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x[1] = x[0];
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x[0] = signal[i];
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/* Rounding in Q(12-1), i.e. add 2^10 */
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tmpW32b = tmpW32 + 1024;
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/* Saturate (to 2^26) so that the HP filtered signal does not overflow */
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tmpW32b = WEBRTC_SPL_SAT((int32_t)67108863, tmpW32b, (int32_t)-67108864);
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/* Convert back to Q0 and multiply with 2 */
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signal[i] = (int16_t)(tmpW32b >> 11);
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/* Update state (filtered part) */
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y[2] = y[0];
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y[3] = y[1];
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/* upshift tmpW32 by 3 with saturation */
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if (tmpW32>268435455) {
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tmpW32 = WEBRTC_SPL_WORD32_MAX;
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} else if (tmpW32<-268435456) {
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tmpW32 = WEBRTC_SPL_WORD32_MIN;
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} else {
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tmpW32 *= 8;
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}
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y[0] = (int16_t)(tmpW32 >> 16);
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y[1] = (int16_t)((tmpW32 & 0xffff) >> 1);
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}
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return;
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}
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