Here are patches for SDL12 and SDL_mixer for 4 or 6 channel
surround sound on Linux using the Alsa driver. To use them, naturally
you need a sound card that will do 4 or 6 channels and probably also a
recent version of the Alsa drivers and library. Since the only SDL
output driver that knows about surround sound is the Alsa driver,
you���ll want to choose it, using:
export SDL_AUDIODRIVER=alsa
There are no syntactic changes to the programming API. No new
library calls, no differences in arguments.
There are two semantic changes:
(1) For library calls with number of channels as an argument, formerly
you could use only 1 or 2 for the number of channels. Now you
can also use 4 or 6.
(2) The two "left" and "right" arguments to Mix_SetPanning, for the
case of 4 or 6 channels, no longer simply control the volumes of
the left and right channels. Now the "left" argument is converted
to an angle and Mix_SetPosition is called, and the "right" argu-
ment is ignored.
With two exceptions, so far as I know, the modified SDL12 and
SDL_mixer work the same way as the original versions, when opened for
1 or 2 channel output. The two exceptions are bugs which I fixed.
Well, the first, anyway, is a bug for sure. When rate conversions up
or down by a factor of two are applied (in src/audio/SDL_audiocvt.c),
streams with different numbers of channels (that is, mono and stereo)
are treated the same way: either each sample is copied or every other
sample is omitted. This is ok for mono, but for stereo, it is frames
that should be copied or omitted, where by "frame" I mean a portion of
the stream containing one sample for each channel. (In the SDL source,
confusingly, sometimes frames are called "samples".) So for these
rate conversions, stereo streams have to be treated differently, and
they are, in my modified version.
The other problem that might be characterized as a bug arises
when SDL_mixer is passed a multichannel chunk which does not have an
integral number of frames. Due to the way the effect_position code
loops over frames, when the chunk ends with a partial frame, memory
outside the chunk buffer will be accessed. In the case of stereo,
it���s possible that because malloc may give more memory than requested,
this potential problem never actually causes a segment fault. I don���t
know. For 6 channel chunks, I do know, and it does cause segment
faults.
If SDL_mixer is passed defective chunks and this causes a segment
fault, arguably, that���s not a bug in SDL_mixer. Still, whether or not
it counts as a bug, it���s easy to protect against, so why not? I added
code in mixer.c to discard any partial frame at the end of a chunk.
Then what about when SDL or SDL_mixer is opened for 4 or 6 chan-
nel output? What happens with the parts of the current library
designed for stereo? I don���t know whether I���ve covered all the bases,
but I���ve tried:
(1) For playing 2 channel waves, or other cases where SDL knows it has
to match up a 2 channel source with a 4 or 6 channel output, I���ve
added code in SDL_audiocvt.c to make the necessary conversions.
(2) For playing midis using timidity, I���ve converted timidity to do 4
or 6 channel output, upon request.
(3) For playing mods using mikmod, I put ad hoc code in music.c to
convert the stereo output that mikmod produces to 4 or 6 chan-
nels. Obviously it would be better to change the mikmod code to
mix down into 4 or 6 channels, but I have a hard time following
the code in mikmod, so I didn���t do that.
(4) For playing mp3s, I put ad hoc code in smpeg to copy channels in
the case when 4 or 6 channel output is needed.
(5) There seems to be no problem with .ogg files - stereo .oggs can be
up converted as .wavs are.
(6) The effect_position code in SDL_mixer is now generalized to in-
clude the cases of 4 and 6 channel streams.
I���ve done a very limited amount of compatibility testing for some
of the games using SDL I happen to have. For details, see the file
TESTS.
I���ve put into a separate archive, Surround-SDL-testfiles.tgz, a
couple of 6 channel wave files for testing and a 6 channel ogg file.
If you have the right hardware and version of Alsa, you should be able
to play the wave files with the Alsa utility aplay (and hear all
channels, except maybe lfe, for chan-id.wav, since it���s rather faint).
Don���t expect aplay to give good sound, though. There���s something
wrong with the current version of aplay.
The canyon.ogg file is to test loading of 6 channel oggs. After
patching and compiling, you can play it with playmus. (My version of
ogg123 will not play it, and I had to patch mplayer to get it to play
6 channel oggs.)
Greg Lee <greg@ling.lll.hawaii.edu>
Thus, July 1, 2004
/*
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 =
"@(#) $Id$";
#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 = cvt->buf+cvt->len_cvt;
dst = 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 = cvt->buf+cvt->len_cvt;
dst = 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)
{
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)
{
/*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;
double 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 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);
}