Files
platform-external-webrtc/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c
bjornv@webrtc.org 4ab23d0e8f Refactor audio_coding/ilbc: removes usage of macro WEBRTC_SPL_LSHIFT_W32
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.

Also removes unnecessary casts to int32_t from int16_t.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48519004

Cr-Commit-Position: refs/heads/master@{#8800}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8800 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 06:01:43 +00:00

90 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 "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] */
int16_t len) /* (i) Number of samples to filter */
{
int 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 = (tmpW32<<1);
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 <<= 3;
}
y[0] = (int16_t)(tmpW32 >> 16);
y[1] = (int16_t)((tmpW32 - (y[0] << 16)) >> 1);
}
return;
}