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theHobbit
Joined: 30 Sep 2006
Posts: 65
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 Posted: Thu Jul 17, 2008 5:43 am Post subject: Sample Rate Converter (From 44k to 22k, 11k or 8k) |
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Hi there, could someone point me to a fast and simple code to resample an audio buffer from 44k to 22k, 11k or 8k?, i have searched for a while and found libraries like libresample but i cant get to make it as fast as i would like. I'm not really interested in high quality so i thought it could be done with a simpler algorithm? Hope anyone can help.
Thanks in advance. |
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J.F.
Joined: 22 Feb 2004
Posts: 2906
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| Posted: Thu Jul 17, 2008 7:11 am Post subject: |
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Look at the way I modified the audio converter for SDL. In particular, look at the SLOW rate converter. It does point sample conversion using a scaled integer index. That's the fastest you're going to get, and it sounds reasonably good.
| Code: | /*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2004 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Library General Public License for more details.
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Sam Lantinga
slouken@libsdl.org
*/
#ifdef SAVE_RCSID
static char rcsid =
"@(#) $Id: SDL_audiocvt.c,v 1.7 2004/12/13 08:49:17 slouken Exp $";
#endif
/* Functions for audio drivers to perform runtime conversion of audio format */
#include <stdio.h>
#include "SDL_error.h"
#include "SDL_audio.h"
/* Effectively mix right and left channels into a single channel */
void SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Sint32 sample;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to mono\n");
#endif
switch (format&0x8018) {
case AUDIO_U8: {
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
for ( i=cvt->len_cvt/2; i; --i ) {
sample = src[0] + src[1];
if ( sample > 255 ) {
*dst = 255;
} else {
*dst = sample;
}
src += 2;
dst += 1;
}
}
break;
case AUDIO_S8: {
Sint8 *src, *dst;
src = (Sint8 *)cvt->buf;
dst = (Sint8 *)cvt->buf;
for ( i=cvt->len_cvt/2; i; --i ) {
sample = src[0] + src[1];
if ( sample > 127 ) {
*dst = 127;
} else
if ( sample < -128 ) {
*dst = -128;
} else {
*dst = sample;
}
src += 2;
dst += 1;
}
}
break;
case AUDIO_U16: {
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Uint16)((src[0]<<8)|src[1])+
(Uint16)((src[2]<<8)|src[3]);
if ( sample > 65535 ) {
dst[0] = 0xFF;
dst[1] = 0xFF;
} else {
dst[1] = (sample&0xFF);
sample >>= 8;
dst[0] = (sample&0xFF);
}
src += 4;
dst += 2;
}
} else {
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Uint16)((src[1]<<8)|src[0])+
(Uint16)((src[3]<<8)|src[2]);
if ( sample > 65535 ) {
dst[0] = 0xFF;
dst[1] = 0xFF;
} else {
dst[0] = (sample&0xFF);
sample >>= 8;
dst[1] = (sample&0xFF);
}
src += 4;
dst += 2;
}
}
}
break;
case AUDIO_S16: {
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Sint16)((src[0]<<8)|src[1])+
(Sint16)((src[2]<<8)|src[3]);
if ( sample > 32767 ) {
dst[0] = 0x7F;
dst[1] = 0xFF;
} else
if ( sample < -32768 ) {
dst[0] = 0x80;
dst[1] = 0x00;
} else {
dst[1] = (sample&0xFF);
sample >>= 8;
dst[0] = (sample&0xFF);
}
src += 4;
dst += 2;
}
} else {
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Sint16)((src[1]<<8)|src[0])+
(Sint16)((src[3]<<8)|src[2]);
if ( sample > 32767 ) {
dst[1] = 0x7F;
dst[0] = 0xFF;
} else
if ( sample < -32768 ) {
dst[1] = 0x80;
dst[0] = 0x00;
} else {
dst[0] = (sample&0xFF);
sample >>= 8;
dst[1] = (sample&0xFF);
}
src += 4;
dst += 2;
}
}
}
break;
}
cvt->len_cvt /= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Discard top 4 channels */
void SDL_ConvertStrip(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Sint32 lsample, rsample;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting down to stereo\n");
#endif
switch (format&0x8018) {
case AUDIO_U8: {
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
for ( i=cvt->len_cvt/6; i; --i ) {
lsample = src[0];
rsample = src[1];
dst[0] = lsample;
dst[1] = rsample;
src += 6;
dst += 2;
}
}
break;
case AUDIO_S8: {
Sint8 *src, *dst;
src = (Sint8 *)cvt->buf;
dst = (Sint8 *)cvt->buf;
for ( i=cvt->len_cvt/6; i; --i ) {
lsample = src[0];
rsample = src[1];
dst[0] = lsample;
dst[1] = rsample;
src += 6;
dst += 2;
}
}
break;
case AUDIO_U16: {
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/12; i; --i ) {
lsample = (Uint16)((src[0]<<8)|src[1]);
rsample = (Uint16)((src[2]<<8)|src[3]);
dst[1] = (lsample&0xFF);
lsample >>= 8;
dst[0] = (lsample&0xFF);
dst[3] = (rsample&0xFF);
rsample >>= 8;
dst[2] = (rsample&0xFF);
src += 12;
dst += 4;
}
} else {
for ( i=cvt->len_cvt/12; i; --i ) {
lsample = (Uint16)((src[1]<<8)|src[0]);
rsample = (Uint16)((src[3]<<8)|src[2]);
dst[0] = (lsample&0xFF);
lsample >>= 8;
dst[1] = (lsample&0xFF);
dst[2] = (rsample&0xFF);
rsample >>= 8;
dst[3] = (rsample&0xFF);
src += 12;
dst += 4;
}
}
}
break;
case AUDIO_S16: {
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/12; i; --i ) {
lsample = (Sint16)((src[0]<<8)|src[1]);
rsample = (Sint16)((src[2]<<8)|src[3]);
dst[1] = (lsample&0xFF);
lsample >>= 8;
dst[0] = (lsample&0xFF);
dst[3] = (rsample&0xFF);
rsample >>= 8;
dst[2] = (rsample&0xFF);
src += 12;
dst += 4;
}
} else {
for ( i=cvt->len_cvt/12; i; --i ) {
lsample = (Sint16)((src[1]<<8)|src[0]);
rsample = (Sint16)((src[3]<<8)|src[2]);
dst[0] = (lsample&0xFF);
lsample >>= 8;
dst[1] = (lsample&0xFF);
dst[2] = (rsample&0xFF);
rsample >>= 8;
dst[3] = (rsample&0xFF);
src += 12;
dst += 4;
}
}
}
break;
}
cvt->len_cvt /= 3;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Discard top 2 channels of 6 */
void SDL_ConvertStrip_2(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Sint32 lsample, rsample;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting 6 down to quad\n");
#endif
switch (format&0x8018) {
case AUDIO_U8: {
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
for ( i=cvt->len_cvt/4; i; --i ) {
lsample = src[0];
rsample = src[1];
dst[0] = lsample;
dst[1] = rsample;
src += 4;
dst += 2;
}
}
break;
case AUDIO_S8: {
Sint8 *src, *dst;
src = (Sint8 *)cvt->buf;
dst = (Sint8 *)cvt->buf;
for ( i=cvt->len_cvt/4; i; --i ) {
lsample = src[0];
rsample = src[1];
dst[0] = lsample;
dst[1] = rsample;
src += 4;
dst += 2;
}
}
break;
case AUDIO_U16: {
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/8; i; --i ) {
lsample = (Uint16)((src[0]<<8)|src[1]);
rsample = (Uint16)((src[2]<<8)|src[3]);
dst[1] = (lsample&0xFF);
lsample >>= 8;
dst[0] = (lsample&0xFF);
dst[3] = (rsample&0xFF);
rsample >>= 8;
dst[2] = (rsample&0xFF);
src += 8;
dst += 4;
}
} else {
for ( i=cvt->len_cvt/8; i; --i ) {
lsample = (Uint16)((src[1]<<8)|src[0]);
rsample = (Uint16)((src[3]<<8)|src[2]);
dst[0] = (lsample&0xFF);
lsample >>= 8;
dst[1] = (lsample&0xFF);
dst[2] = (rsample&0xFF);
rsample >>= 8;
dst[3] = (rsample&0xFF);
src += 8;
dst += 4;
}
}
}
break;
case AUDIO_S16: {
Uint8 *src, *dst;
src = cvt->buf;
dst = cvt->buf;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/8; i; --i ) {
lsample = (Sint16)((src[0]<<8)|src[1]);
rsample = (Sint16)((src[2]<<8)|src[3]);
dst[1] = (lsample&0xFF);
lsample >>= 8;
dst[0] = (lsample&0xFF);
dst[3] = (rsample&0xFF);
rsample >>= 8;
dst[2] = (rsample&0xFF);
src += 8;
dst += 4;
}
} else {
for ( i=cvt->len_cvt/8; i; --i ) {
lsample = (Sint16)((src[1]<<8)|src[0]);
rsample = (Sint16)((src[3]<<8)|src[2]);
dst[0] = (lsample&0xFF);
lsample >>= 8;
dst[1] = (lsample&0xFF);
dst[2] = (rsample&0xFF);
rsample >>= 8;
dst[3] = (rsample&0xFF);
src += 8;
dst += 4;
}
}
}
break;
}
cvt->len_cvt /= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Duplicate a mono channel to both stereo channels */
void SDL_ConvertStereo(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to stereo\n");
#endif
if ( (format & 0xFF) == 16 ) {
Uint16 *src, *dst;
src = (Uint16 *)(cvt->buf+cvt->len_cvt);
dst = (Uint16 *)(cvt->buf+cvt->len_cvt*2);
for ( i=cvt->len_cvt/2; i; --i ) {
dst -= 2;
src -= 1;
dst[0] = src[0];
dst[1] = src[0];
}
} else {
Uint8 *src, *dst;
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
for ( i=cvt->len_cvt; i; --i ) {
dst -= 2;
src -= 1;
dst[0] = src[0];
dst[1] = src[0];
}
}
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Duplicate a stereo channel to a pseudo-5.1 stream */
void SDL_ConvertSurround(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting stereo to surround\n");
#endif
switch (format&0x8018) {
case AUDIO_U8: {
Uint8 *src, *dst, lf, rf, ce;
src = (Uint8 *)(cvt->buf+cvt->len_cvt);
dst = (Uint8 *)(cvt->buf+cvt->len_cvt*3);
for ( i=cvt->len_cvt; i; --i ) {
dst -= 6;
src -= 2;
lf = src[0];
rf = src[1];
ce = (lf/2) + (rf/2);
dst[0] = lf;
dst[1] = rf;
dst[2] = lf - ce;
dst[3] = rf - ce;
dst[4] = ce;
dst[5] = ce;
}
}
break;
case AUDIO_S8: {
Sint8 *src, *dst, lf, rf, ce;
src = (Sint8 *)cvt->buf+cvt->len_cvt;
dst = (Sint8 *)cvt->buf+cvt->len_cvt*3;
for ( i=cvt->len_cvt; i; --i ) {
dst -= 6;
src -= 2;
lf = src[0];
rf = src[1];
ce = (lf/2) + (rf/2);
dst[0] = lf;
dst[1] = rf;
dst[2] = lf - ce;
dst[3] = rf - ce;
dst[4] = ce;
dst[5] = ce;
}
}
break;
case AUDIO_U16: {
Uint8 *src, *dst;
Uint16 lf, rf, ce, lr, rr;
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*3;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/4; i; --i ) {
dst -= 12;
src -= 4;
lf = (Uint16)((src[0]<<8)|src[1]);
rf = (Uint16)((src[2]<<8)|src[3]);
ce = (lf/2) + (rf/2);
rr = lf - ce;
lr = rf - ce;
dst[1] = (lf&0xFF);
dst[0] = ((lf>>8)&0xFF);
dst[3] = (rf&0xFF);
dst[2] = ((rf>>8)&0xFF);
dst[1+4] = (lr&0xFF);
dst[0+4] = ((lr>>8)&0xFF);
dst[3+4] = (rr&0xFF);
dst[2+4] = ((rr>>8)&0xFF);
dst[1+8] = (ce&0xFF);
dst[0+8] = ((ce>>8)&0xFF);
dst[3+8] = (ce&0xFF);
dst[2+8] = ((ce>>8)&0xFF);
}
} else {
for ( i=cvt->len_cvt/4; i; --i ) {
dst -= 12;
src -= 4;
lf = (Uint16)((src[1]<<8)|src[0]);
rf = (Uint16)((src[3]<<8)|src[2]);
ce = (lf/2) + (rf/2);
rr = lf - ce;
lr = rf - ce;
dst[0] = (lf&0xFF);
dst[1] = ((lf>>8)&0xFF);
dst[2] = (rf&0xFF);
dst[3] = ((rf>>8)&0xFF);
dst[0+4] = (lr&0xFF);
dst[1+4] = ((lr>>8)&0xFF);
dst[2+4] = (rr&0xFF);
dst[3+4] = ((rr>>8)&0xFF);
dst[0+8] = (ce&0xFF);
dst[1+8] = ((ce>>8)&0xFF);
dst[2+8] = (ce&0xFF);
dst[3+8] = ((ce>>8)&0xFF);
}
}
}
break;
case AUDIO_S16: {
Uint8 *src, *dst;
Sint16 lf, rf, ce, lr, rr;
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*3;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/4; i; --i ) {
dst -= 12;
src -= 4;
lf = (Sint16)((src[0]<<8)|src[1]);
rf = (Sint16)((src[2]<<8)|src[3]);
ce = (lf/2) + (rf/2);
rr = lf - ce;
lr = rf - ce;
dst[1] = (lf&0xFF);
dst[0] = ((lf>>8)&0xFF);
dst[3] = (rf&0xFF);
dst[2] = ((rf>>8)&0xFF);
dst[1+4] = (lr&0xFF);
dst[0+4] = ((lr>>8)&0xFF);
dst[3+4] = (rr&0xFF);
dst[2+4] = ((rr>>8)&0xFF);
dst[1+8] = (ce&0xFF);
dst[0+8] = ((ce>>8)&0xFF);
dst[3+8] = (ce&0xFF);
dst[2+8] = ((ce>>8)&0xFF);
}
} else {
for ( i=cvt->len_cvt/4; i; --i ) {
dst -= 12;
src -= 4;
lf = (Sint16)((src[1]<<8)|src[0]);
rf = (Sint16)((src[3]<<8)|src[2]);
ce = (lf/2) + (rf/2);
rr = lf - ce;
lr = rf - ce;
dst[0] = (lf&0xFF);
dst[1] = ((lf>>8)&0xFF);
dst[2] = (rf&0xFF);
dst[3] = ((rf>>8)&0xFF);
dst[0+4] = (lr&0xFF);
dst[1+4] = ((lr>>8)&0xFF);
dst[2+4] = (rr&0xFF);
dst[3+4] = ((rr>>8)&0xFF);
dst[0+8] = (ce&0xFF);
dst[1+8] = ((ce>>8)&0xFF);
dst[2+8] = (ce&0xFF);
dst[3+8] = ((ce>>8)&0xFF);
}
}
}
break;
}
cvt->len_cvt *= 3;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Duplicate a stereo channel to a pseudo-4.0 stream */
void SDL_ConvertSurround_4(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting stereo to quad\n");
#endif
switch (format&0x8018) {
case AUDIO_U8: {
Uint8 *src, *dst, lf, rf, ce;
src = (Uint8 *)(cvt->buf+cvt->len_cvt);
dst = (Uint8 *)(cvt->buf+cvt->len_cvt*2);
for ( i=cvt->len_cvt; i; --i ) {
dst -= 4;
src -= 2;
lf = src[0];
rf = src[1];
ce = (lf/2) + (rf/2);
dst[0] = lf;
dst[1] = rf;
dst[2] = lf - ce;
dst[3] = rf - ce;
}
}
break;
case AUDIO_S8: {
Sint8 *src, *dst, lf, rf, ce;
src = (Sint8 *)cvt->buf+cvt->len_cvt;
dst = (Sint8 *)cvt->buf+cvt->len_cvt*2;
for ( i=cvt->len_cvt; i; --i ) {
dst -= 4;
src -= 2;
lf = src[0];
rf = src[1];
ce = (lf/2) + (rf/2);
dst[0] = lf;
dst[1] = rf;
dst[2] = lf - ce;
dst[3] = rf - ce;
}
}
break;
case AUDIO_U16: {
Uint8 *src, *dst;
Uint16 lf, rf, ce, lr, rr;
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/4; i; --i ) {
dst -= 8;
src -= 4;
lf = (Uint16)((src[0]<<8)|src[1]);
rf = (Uint16)((src[2]<<8)|src[3]);
ce = (lf/2) + (rf/2);
rr = lf - ce;
lr = rf - ce;
dst[1] = (lf&0xFF);
dst[0] = ((lf>>8)&0xFF);
dst[3] = (rf&0xFF);
dst[2] = ((rf>>8)&0xFF);
dst[1+4] = (lr&0xFF);
dst[0+4] = ((lr>>8)&0xFF);
dst[3+4] = (rr&0xFF);
dst[2+4] = ((rr>>8)&0xFF);
}
} else {
for ( i=cvt->len_cvt/4; i; --i ) {
dst -= 8;
src -= 4;
lf = (Uint16)((src[1]<<8)|src[0]);
rf = (Uint16)((src[3]<<8)|src[2]);
ce = (lf/2) + (rf/2);
rr = lf - ce;
lr = rf - ce;
dst[0] = (lf&0xFF);
dst[1] = ((lf>>8)&0xFF);
dst[2] = (rf&0xFF);
dst[3] = ((rf>>8)&0xFF);
dst[0+4] = (lr&0xFF);
dst[1+4] = ((lr>>8)&0xFF);
dst[2+4] = (rr&0xFF);
dst[3+4] = ((rr>>8)&0xFF);
}
}
}
break;
case AUDIO_S16: {
Uint8 *src, *dst;
Sint16 lf, rf, ce, lr, rr;
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/4; i; --i ) {
dst -= 8;
src -= 4;
lf = (Sint16)((src[0]<<8)|src[1]);
rf = (Sint16)((src[2]<<8)|src[3]);
ce = (lf/2) + (rf/2);
rr = lf - ce;
lr = rf - ce;
dst[1] = (lf&0xFF);
dst[0] = ((lf>>8)&0xFF);
dst[3] = (rf&0xFF);
dst[2] = ((rf>>8)&0xFF);
dst[1+4] = (lr&0xFF);
dst[0+4] = ((lr>>8)&0xFF);
dst[3+4] = (rr&0xFF);
dst[2+4] = ((rr>>8)&0xFF);
}
} else {
for ( i=cvt->len_cvt/4; i; --i ) {
dst -= 8;
src -= 4;
lf = (Sint16)((src[1]<<8)|src[0]);
rf = (Sint16)((src[3]<<8)|src[2]);
ce = (lf/2) + (rf/2);
rr = lf - ce;
lr = rf - ce;
dst[0] = (lf&0xFF);
dst[1] = ((lf>>8)&0xFF);
dst[2] = (rf&0xFF);
dst[3] = ((rf>>8)&0xFF);
dst[0+4] = (lr&0xFF);
dst[1+4] = ((lr>>8)&0xFF);
dst[2+4] = (rr&0xFF);
dst[3+4] = ((rr>>8)&0xFF);
}
}
}
break;
}
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert 8-bit to 16-bit - LSB */
void SDL_Convert16LSB(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to 16-bit LSB\n");
#endif
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
for ( i=cvt->len_cvt; i; --i ) {
src -= 1;
dst -= 2;
dst[1] = *src;
dst[0] = 0;
}
format = ((format & ~0x0008) | AUDIO_U16LSB);
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert 8-bit to 16-bit - MSB */
void SDL_Convert16MSB(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to 16-bit MSB\n");
#endif
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
for ( i=cvt->len_cvt; i; --i ) {
src -= 1;
dst -= 2;
dst[0] = *src;
dst[1] = 0;
}
format = ((format & ~0x0008) | AUDIO_U16MSB);
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert 16-bit to 8-bit */
void SDL_Convert8(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to 8-bit\n");
#endif
src = cvt->buf;
dst = cvt->buf;
if ( (format & 0x1000) != 0x1000 ) { /* Little endian */
++src;
}
for ( i=cvt->len_cvt/2; i; --i ) {
*dst = *src;
src += 2;
dst += 1;
}
format = ((format & ~0x9010) | AUDIO_U8);
cvt->len_cvt /= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Toggle signed/unsigned */
void SDL_ConvertSign(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *data;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio signedness\n");
#endif
data = cvt->buf;
if ( (format & 0xFF) == 16 ) {
if ( (format & 0x1000) != 0x1000 ) { /* Little endian */
++data;
}
for ( i=cvt->len_cvt/2; i; --i ) {
*data ^= 0x80;
data += 2;
}
} else {
for ( i=cvt->len_cvt; i; --i ) {
*data++ ^= 0x80;
}
}
format = (format ^ 0x8000);
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Toggle endianness */
void SDL_ConvertEndian(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *data, tmp;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio endianness\n");
#endif
data = cvt->buf;
for ( i=cvt->len_cvt/2; i; --i ) {
tmp = data[0];
data[0] = data[1];
data[1] = tmp;
data += 2;
}
format = (format ^ 0x1000);
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert rate up by multiple of 2 */
void SDL_RateMUL2(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate * 2\n");
#endif
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
switch (format & 0xFF) {
case 8:
for ( i=cvt->len_cvt; i; --i ) {
src -= 1;
dst -= 2;
dst[0] = src[0];
dst[1] = src[0];
}
break;
case 16:
for ( i=cvt->len_cvt/2; i; --i ) {
src -= 2;
dst -= 4;
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[0];
dst[3] = src[1];
}
break;
}
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert rate up by multiple of 2, for stereo */
void SDL_RateMUL2_c2(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate * 2\n");
#endif
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
switch (format & 0xFF) {
case 8:
for ( i=cvt->len_cvt/2; i; --i ) {
src -= 2;
dst -= 4;
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[0];
dst[3] = src[1];
}
break;
case 16:
for ( i=cvt->len_cvt/4; i; --i ) {
src -= 4;
dst -= 8;
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[2];
dst[3] = src[3];
dst[4] = src[0];
dst[5] = src[1];
dst[6] = src[2];
dst[7] = src[3];
}
break;
}
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert rate up by multiple of 2, for quad */
void SDL_RateMUL2_c4(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate * 2\n");
#endif
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
switch (format & 0xFF) {
case 8:
for ( i=cvt->len_cvt/4; i; --i ) {
src -= 4;
dst -= 8;
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[2];
dst[3] = src[3];
dst[4] = src[0];
dst[5] = src[1];
dst[6] = src[2];
dst[7] = src[3];
}
break;
case 16:
for ( i=cvt->len_cvt/8; i; --i ) {
src -= 8;
dst -= 16;
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[2];
dst[3] = src[3];
dst[4] = src[4];
dst[5] = src[5];
dst[6] = src[6];
dst[7] = src[7];
dst[8] = src[0];
dst[9] = src[1];
dst[10] = src[2];
dst[11] = src[3];
dst[12] = src[4];
dst[13] = src[5];
dst[14] = src[6];
dst[15] = src[7];
}
break;
}
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert rate up by multiple of 2, for 5.1 */
void SDL_RateMUL2_c6(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate * 2\n");
#endif
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
switch (format & 0xFF) {
case 8:
for ( i=cvt->len_cvt/6; i; --i ) {
src -= 6;
dst -= 12;
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[2];
dst[3] = src[3];
dst[4] = src[4];
dst[5] = src[5];
dst[6] = src[0];
dst[7] = src[1];
dst[8] = src[2];
dst[9] = src[3];
dst[10] = src[4];
dst[11] = src[5];
}
break;
case 16:
for ( i=cvt->len_cvt/12; i; --i ) {
src -= 12;
dst -= 24;
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[2];
dst[3] = src[3];
dst[4] = src[4];
dst[5] = src[5];
dst[6] = src[6];
dst[7] = src[7];
dst[8] = src[8];
dst[9] = src[9];
dst[10] = src[10];
dst[11] = src[11];
dst[12] = src[0];
dst[13] = src[1];
dst[14] = src[2];
dst[15] = src[3];
dst[16] = src[4];
dst[17] = src[5];
dst[18] = src[6];
dst[19] = src[7];
dst[20] = src[8];
dst[21] = src[9];
dst[22] = src[10];
dst[23] = src[11];
}
break;
}
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert rate down by multiple of 2 */
void SDL_RateDIV2(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate / 2\n");
#endif
src = cvt->buf;
dst = cvt->buf;
switch (format & 0xFF) {
case 8:
for ( i=cvt->len_cvt/2; i; --i ) {
dst[0] = src[0];
src += 2;
dst += 1;
}
break;
case 16:
for ( i=cvt->len_cvt/4; i; --i ) {
dst[0] = src[0];
dst[1] = src[1];
src += 4;
dst += 2;
}
break;
}
cvt->len_cvt /= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert rate down by multiple of 2, for stereo */
void SDL_RateDIV2_c2(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate / 2\n");
#endif
src = cvt->buf;
dst = cvt->buf;
switch (format & 0xFF) {
case 8:
for ( i=cvt->len_cvt/4; i; --i ) {
dst[0] = src[0];
dst[1] = src[1];
src += 4;
dst += 2;
}
break;
case 16:
for ( i=cvt->len_cvt/8; i; --i ) {
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[2];
dst[3] = src[3];
src += 8;
dst += 4;
}
break;
}
cvt->len_cvt /= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert rate down by multiple of 2, for quad */
void SDL_RateDIV2_c4(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate / 2\n");
#endif
src = cvt->buf;
dst = cvt->buf;
switch (format & 0xFF) {
case 8:
for ( i=cvt->len_cvt/8; i; --i ) {
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[2];
dst[3] = src[3];
src += 8;
dst += 4;
}
break;
case 16:
for ( i=cvt->len_cvt/16; i; --i ) {
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[2];
dst[3] = src[3];
dst[4] = src[4];
dst[5] = src[5];
dst[6] = src[6];
dst[7] = src[7];
src += 16;
dst += 8;
}
break;
}
cvt->len_cvt /= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert rate down by multiple of 2, for 5.1 */
void SDL_RateDIV2_c6(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate / 2\n");
#endif
src = cvt->buf;
dst = cvt->buf;
switch (format & 0xFF) {
case 8:
for ( i=cvt->len_cvt/12; i; --i ) {
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[2];
dst[3] = src[3];
dst[4] = src[4];
dst[5] = src[5];
src += 12;
dst += 6;
}
break;
case 16:
for ( i=cvt->len_cvt/24; i; --i ) {
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[2];
dst[3] = src[3];
dst[4] = src[4];
dst[5] = src[5];
dst[6] = src[6];
dst[7] = src[7];
dst[8] = src[8];
dst[9] = src[9];
dst[10] = src[10];
dst[11] = src[11];
src += 24;
dst += 12;
}
break;
}
cvt->len_cvt /= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Very slow rate conversion routine */
void SDL_RateSLOW(SDL_AudioCVT *cvt, Uint16 format)
{
int ipos, incr;
int i, clen;
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr);
#endif
incr = (int)(cvt->rate_incr * 512.0f);
clen = (int)((float)cvt->len_cvt / cvt->rate_incr);
if ( cvt->rate_incr > 1.0 ) {
switch (format & 0xFF) {
case 8: {
Uint8 *output;
output = cvt->buf;
ipos = 0;
for ( i=clen; i; --i ) {
*output = cvt->buf[ipos>>9];
ipos += incr;
output += 1;
}
}
break;
case 16: {
Uint16 *output;
clen &= ~1;
output = (Uint16 *)cvt->buf;
ipos = 0;
for ( i=clen/2; i; --i ) {
*output=((Uint16 *)cvt->buf)[ipos>>9];
ipos += incr;
output += 1;
}
}
break;
}
} else {
switch (format & 0xFF) {
case 8: {
Uint8 *output;
output = cvt->buf+clen;
ipos = cvt->len_cvt<<9;
for ( i=clen; i; --i ) {
ipos -= incr;
output -= 1;
*output = cvt->buf[ipos>>9];
}
}
break;
case 16: {
Uint16 *output;
clen &= ~1;
output = (Uint16 *)(cvt->buf+clen);
ipos = cvt->len_cvt<<8;
for ( i=clen/2; i; --i ) {
ipos -= incr;
output -= 1;
*output=((Uint16 *)cvt->buf)[ipos>>9];
}
}
break;
}
}
cvt->len_cvt = clen;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
int SDL_ConvertAudio(SDL_AudioCVT *cvt)
{
/* Make sure there's data to convert */
if ( cvt->buf == NULL ) {
SDL_SetError("No buffer allocated for conversion");
return(-1);
}
/* Return okay if no conversion is necessary */
cvt->len_cvt = cvt->len;
if ( cvt->filters[0] == NULL ) {
return(0);
}
/* Set up the conversion and go! */
cvt->filter_index = 0;
cvt->filters[0](cvt, cvt->src_format);
return(0);
}
/* Creates a set of audio filters to convert from one format to another.
Returns -1 if the format conversion is not supported, or 1 if the
audio filter is set up.
*/
int SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
Uint16 src_format, Uint8 src_channels, int src_rate,
Uint16 dst_format, Uint8 dst_channels, int dst_rate)
{
/*printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
src_format, dst_format, src_channels, dst_channels, src_rate, dst_rate);*/
/* Start off with no conversion necessary */
cvt->needed = 0;
cvt->filter_index = 0;
cvt->filters[0] = NULL;
cvt->len_mult = 1;
cvt->len_ratio = 1.0;
/* First filter: Endian conversion from src to dst */
if ( (src_format & 0x1000) != (dst_format & 0x1000)
&& ((src_format & 0xff) != 8) ) {
cvt->filters[cvt->filter_index++] = SDL_ConvertEndian;
}
/* Second filter: Sign conversion -- signed/unsigned */
if ( (src_format & 0x8000) != (dst_format & 0x8000) ) {
cvt->filters[cvt->filter_index++] = SDL_ConvertSign;
}
/* Next filter: Convert 16 bit <--> 8 bit PCM */
if ( (src_format & 0xFF) != (dst_format & 0xFF) ) {
switch (dst_format&0x10FF) {
case AUDIO_U8:
cvt->filters[cvt->filter_index++] =
SDL_Convert8;
cvt->len_ratio /= 2;
break;
case AUDIO_U16LSB:
cvt->filters[cvt->filter_index++] =
SDL_Convert16LSB;
cvt->len_mult *= 2;
cvt->len_ratio *= 2;
break;
case AUDIO_U16MSB:
cvt->filters[cvt->filter_index++] =
SDL_Convert16MSB;
cvt->len_mult *= 2;
cvt->len_ratio *= 2;
break;
}
}
/* Last filter: Mono/Stereo conversion */
if ( src_channels != dst_channels ) {
if ( (src_channels == 1) && (dst_channels > 1) ) {
cvt->filters[cvt->filter_index++] =
SDL_ConvertStereo;
cvt->len_mult *= 2;
src_channels = 2;
cvt->len_ratio *= 2;
}
if ( (src_channels == 2) &&
(dst_channels == 6) ) {
cvt->filters[cvt->filter_index++] =
SDL_ConvertSurround;
src_channels = 6;
cvt->len_mult *= 3;
cvt->len_ratio *= 3;
}
if ( (src_channels == 2) &&
(dst_channels == 4) ) {
cvt->filters[cvt->filter_index++] =
SDL_ConvertSurround_4;
src_channels = 4;
cvt->len_mult *= 2;
cvt->len_ratio *= 2;
}
while ( (src_channels*2) <= dst_channels ) {
cvt->filters[cvt->filter_index++] =
SDL_ConvertStereo;
cvt->len_mult *= 2;
src_channels *= 2;
cvt->len_ratio *= 2;
}
if ( (src_channels == 6) &&
(dst_channels <= 2) ) {
cvt->filters[cvt->filter_index++] =
SDL_ConvertStrip;
src_channels = 2;
cvt->len_ratio /= 3;
}
if ( (src_channels == 6) &&
(dst_channels == 4) ) {
cvt->filters[cvt->filter_index++] =
SDL_ConvertStrip_2;
src_channels = 4;
cvt->len_ratio /= 2;
}
/* This assumes that 4 channel audio is in the format:
Left {front/back} + Right {front/back}
so converting to L/R stereo works properly.
*/
while ( ((src_channels%2) == 0) &&
((src_channels/2) >= dst_channels) ) {
cvt->filters[cvt->filter_index++] =
SDL_ConvertMono;
src_channels /= 2;
cvt->len_ratio /= 2;
}
if ( src_channels != dst_channels ) {
/* Uh oh.. */;
}
}
/* Do rate conversion */
cvt->rate_incr = 0.0;
if ( (src_rate/100) != (dst_rate/100) ) {
Uint32 hi_rate, lo_rate;
int len_mult;
float len_ratio;
void (*rate_cvt)(SDL_AudioCVT *cvt, Uint16 format);
if ( src_rate > dst_rate ) {
hi_rate = src_rate;
lo_rate = dst_rate;
switch (src_channels) {
case 1: rate_cvt = SDL_RateDIV2; break;
case 2: rate_cvt = SDL_RateDIV2_c2; break;
case 4: rate_cvt = SDL_RateDIV2_c4; break;
case 6: rate_cvt = SDL_RateDIV2_c6; break;
default: return -1;
}
len_mult = 1;
len_ratio = 0.5;
} else {
hi_rate = dst_rate;
lo_rate = src_rate;
switch (src_channels) {
case 1: rate_cvt = SDL_RateMUL2; break;
case 2: rate_cvt = SDL_RateMUL2_c2; break;
case 4: rate_cvt = SDL_RateMUL2_c4; break;
case 6: rate_cvt = SDL_RateMUL2_c6; break;
default: return -1;
}
len_mult = 2;
len_ratio = 2.0;
}
/* If hi_rate = lo_rate*2^x then conversion is easy */
while ( ((lo_rate*2)/100) <= (hi_rate/100) ) {
cvt->filters[cvt->filter_index++] = rate_cvt;
cvt->len_mult *= len_mult;
lo_rate *= 2;
cvt->len_ratio *= len_ratio;
}
/* We may need a slow conversion here to finish up */
if ( (lo_rate/100) != (hi_rate/100) ) {
#if 0
/* The problem with this is that if the input buffer is
say 1K, and the conversion rate is say 1.1, then the
output buffer is 1.1K, which may not be an acceptable
buffer size for the audio driver (not a power of 2)
*/
/* For now, punt and hope the rate distortion isn't great.
*/
#else
if ( src_rate < dst_rate ) {
cvt->rate_incr = (float)lo_rate/hi_rate;
cvt->len_mult *= 2;
cvt->len_ratio /= cvt->rate_incr;
} else {
cvt->rate_incr = (float)hi_rate/lo_rate;
cvt->len_ratio *= cvt->rate_incr;
}
cvt->filters[cvt->filter_index++] = SDL_RateSLOW;
#endif
}
}
/* Set up the filter information */
if ( cvt->filter_index != 0 ) {
cvt->needed = 1;
cvt->src_format = src_format;
cvt->dst_format = dst_format;
cvt->len = 0;
cvt->buf = NULL;
cvt->filters[cvt->filter_index] = NULL;
}
return(cvt->needed);
}
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theHobbit
Joined: 30 Sep 2006
Posts: 65
|
| Posted: Fri Jul 18, 2008 7:33 am Post subject: |
|
|
Thanks, in fact i'm using SDL in my app, but i thought the SDL audio converters werent working because the audio was playing like 2x faster.
Have you managed to get the SDL audio converter to work fine?. I´m trying to play audio in 22k using the adplug engine, but i cant get it to sound right. |
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J.F.
Joined: 22 Feb 2004
Posts: 2906
|
| Posted: Fri Jul 18, 2008 8:39 am Post subject: |
|
|
| theHobbit wrote: | Thanks, in fact i'm using SDL in my app, but i thought the SDL audio converters werent working because the audio was playing like 2x faster.
Have you managed to get the SDL audio converter to work fine?. I´m trying to play audio in 22k using the adplug engine, but i cant get it to sound right. |
Yes, just substitute the above for audio/SDL_audiocvt.c and recompile. |
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