forked from KolibriOS/kolibrios
643 lines
14 KiB
C
643 lines
14 KiB
C
/*
|
|
SDL - Simple DirectMedia Layer
|
|
Copyright (C) 1997, 1998, 1999, 2000, 2001 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@devolution.com
|
|
*/
|
|
|
|
#ifdef SAVE_RCSID
|
|
static char rcsid =
|
|
"@(#) $Id: SDL_audiocvt.c,v 1.2 2001/04/26 16:50:17 hercules 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);
|
|
}
|
|
}
|
|
|
|
|
|
/* 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);
|
|
}
|
|
}
|
|
|
|
/* 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 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);
|
|
}
|
|
}
|
|
|
|
/* Very slow rate conversion routine */
|
|
void SDL_RateSLOW(SDL_AudioCVT *cvt, Uint16 format)
|
|
{
|
|
double ipos;
|
|
int i, clen;
|
|
|
|
#ifdef DEBUG_CONVERT
|
|
fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr);
|
|
#endif
|
|
clen = (int)((double)cvt->len_cvt / cvt->rate_incr);
|
|
if ( cvt->rate_incr > 1.0 ) {
|
|
switch (format & 0xFF) {
|
|
case 8: {
|
|
Uint8 *output;
|
|
|
|
output = cvt->buf;
|
|
ipos = 0.0;
|
|
for ( i=clen; i; --i ) {
|
|
*output = cvt->buf[(int)ipos];
|
|
ipos += cvt->rate_incr;
|
|
output += 1;
|
|
}
|
|
}
|
|
break;
|
|
|
|
case 16: {
|
|
Uint16 *output;
|
|
|
|
clen &= ~1;
|
|
output = (Uint16 *)cvt->buf;
|
|
ipos = 0.0;
|
|
for ( i=clen/2; i; --i ) {
|
|
*output=((Uint16 *)cvt->buf)[(int)ipos];
|
|
ipos += cvt->rate_incr;
|
|
output += 1;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
} else {
|
|
switch (format & 0xFF) {
|
|
case 8: {
|
|
Uint8 *output;
|
|
|
|
output = cvt->buf+clen;
|
|
ipos = (double)cvt->len_cvt;
|
|
for ( i=clen; i; --i ) {
|
|
ipos -= cvt->rate_incr;
|
|
output -= 1;
|
|
*output = cvt->buf[(int)ipos];
|
|
}
|
|
}
|
|
break;
|
|
|
|
case 16: {
|
|
Uint16 *output;
|
|
|
|
clen &= ~1;
|
|
output = (Uint16 *)(cvt->buf+clen);
|
|
ipos = (double)cvt->len_cvt/2;
|
|
for ( i=clen/2; i; --i ) {
|
|
ipos -= cvt->rate_incr;
|
|
output -= 1;
|
|
*output=((Uint16 *)cvt->buf)[(int)ipos];
|
|
}
|
|
}
|
|
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)
|
|
{
|
|
/* 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 ) {
|
|
while ( (src_channels*2) <= dst_channels ) {
|
|
cvt->filters[cvt->filter_index++] =
|
|
SDL_ConvertStereo;
|
|
cvt->len_mult *= 2;
|
|
src_channels *= 2;
|
|
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;
|
|
double len_ratio;
|
|
void (*rate_cvt)(SDL_AudioCVT *cvt, Uint16 format);
|
|
|
|
if ( src_rate > dst_rate ) {
|
|
hi_rate = src_rate;
|
|
lo_rate = dst_rate;
|
|
rate_cvt = SDL_RateDIV2;
|
|
len_mult = 1;
|
|
len_ratio = 0.5;
|
|
} else {
|
|
hi_rate = dst_rate;
|
|
lo_rate = src_rate;
|
|
rate_cvt = SDL_RateMUL2;
|
|
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 1
|
|
/* 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 = (double)lo_rate/hi_rate;
|
|
cvt->len_mult *= 2;
|
|
cvt->len_ratio /= cvt->rate_incr;
|
|
} else {
|
|
cvt->rate_incr = (double)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);
|
|
}
|