mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00

WebRTC internal code should always use include paths that start from the root of the project and that clearly identify the header file. This allows 'gn check' to actually keep dependencies under control because 'gn check' cannot enforce anything if the include path is not fully qualified (starting from the root of the project). Bug: webrtc:8815 Change-Id: I36f01784fa5f5b77eefc02db479b1f7f6ee1a8c3 Reviewed-on: https://webrtc-review.googlesource.com/46263 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21871}
89 lines
2.8 KiB
C
89 lines
2.8 KiB
C
/*
|
|
* Copyright (c) 2011 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.
|
|
*/
|
|
|
|
/******************************************************************
|
|
|
|
iLBC Speech Coder ANSI-C Source Code
|
|
|
|
WebRtcIlbcfix_HpOutput.c
|
|
|
|
******************************************************************/
|
|
|
|
#include "modules/audio_coding/codecs/ilbc/defines.h"
|
|
|
|
/*----------------------------------------------------------------*
|
|
* high-pass filter of output and *2 with saturation
|
|
*---------------------------------------------------------------*/
|
|
|
|
void WebRtcIlbcfix_HpOutput(
|
|
int16_t *signal, /* (i/o) signal vector */
|
|
int16_t *ba, /* (i) B- and A-coefficients (2:nd order)
|
|
{b[0] b[1] b[2] -a[1] -a[2]} a[0]
|
|
is assumed to be 1.0 */
|
|
int16_t *y, /* (i/o) Filter state yhi[n-1] ylow[n-1]
|
|
yhi[n-2] ylow[n-2] */
|
|
int16_t *x, /* (i/o) Filter state x[n-1] x[n-2] */
|
|
size_t len) /* (i) Number of samples to filter */
|
|
{
|
|
size_t i;
|
|
int32_t tmpW32;
|
|
int32_t tmpW32b;
|
|
|
|
for (i=0; i<len; i++) {
|
|
|
|
/*
|
|
y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
|
|
+ (-a[1])*y[i-1] + (-a[2])*y[i-2];
|
|
*/
|
|
|
|
tmpW32 = y[1] * ba[3]; /* (-a[1])*y[i-1] (low part) */
|
|
tmpW32 += y[3] * ba[4]; /* (-a[2])*y[i-2] (low part) */
|
|
tmpW32 = (tmpW32>>15);
|
|
tmpW32 += y[0] * ba[3]; /* (-a[1])*y[i-1] (high part) */
|
|
tmpW32 += y[2] * ba[4]; /* (-a[2])*y[i-2] (high part) */
|
|
tmpW32 *= 2;
|
|
|
|
tmpW32 += signal[i] * ba[0]; /* b[0]*x[0] */
|
|
tmpW32 += x[0] * ba[1]; /* b[1]*x[i-1] */
|
|
tmpW32 += x[1] * ba[2]; /* b[2]*x[i-2] */
|
|
|
|
/* Update state (input part) */
|
|
x[1] = x[0];
|
|
x[0] = signal[i];
|
|
|
|
/* Rounding in Q(12-1), i.e. add 2^10 */
|
|
tmpW32b = tmpW32 + 1024;
|
|
|
|
/* Saturate (to 2^26) so that the HP filtered signal does not overflow */
|
|
tmpW32b = WEBRTC_SPL_SAT((int32_t)67108863, tmpW32b, (int32_t)-67108864);
|
|
|
|
/* Convert back to Q0 and multiply with 2 */
|
|
signal[i] = (int16_t)(tmpW32b >> 11);
|
|
|
|
/* Update state (filtered part) */
|
|
y[2] = y[0];
|
|
y[3] = y[1];
|
|
|
|
/* upshift tmpW32 by 3 with saturation */
|
|
if (tmpW32>268435455) {
|
|
tmpW32 = WEBRTC_SPL_WORD32_MAX;
|
|
} else if (tmpW32<-268435456) {
|
|
tmpW32 = WEBRTC_SPL_WORD32_MIN;
|
|
} else {
|
|
tmpW32 *= 8;
|
|
}
|
|
|
|
y[0] = (int16_t)(tmpW32 >> 16);
|
|
y[1] = (int16_t)((tmpW32 & 0xffff) >> 1);
|
|
|
|
}
|
|
|
|
return;
|
|
}
|