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

/* WAVE files are little-endian */

/*******************************************/
/* Define values for Microsoft WAVE format */
/*******************************************/
#define RIFF		0x46464952		/* "RIFF" */
#define WAVE		0x45564157		/* "WAVE" */
#define FACT		0x74636166		/* "fact" */
#define LIST		0x5453494c		/* "LIST" */
#define FMT		0x20746D66		/* "fmt " */
#define DATA		0x61746164		/* "data" */
#define PCM_CODE	0x0001
#define MS_ADPCM_CODE	0x0002
#define IMA_ADPCM_CODE	0x0011
#define WAVE_MONO	1
#define WAVE_STEREO	2

/* Normally, these three chunks come consecutively in a WAVE file */
typedef struct WaveFMT {
/* Not saved in the chunk we read:
	Uint32	FMTchunk;
	Uint32	fmtlen;
*/
	Uint16	encoding;	
	Uint16	channels;		/* 1 = mono, 2 = stereo */
	Uint32	frequency;		/* One of 11025, 22050, or 44100 Hz */
	Uint32	byterate;		/* Average bytes per second */
	Uint16	blockalign;		/* Bytes per sample block */
	Uint16	bitspersample;		/* One of 8, 12, 16, or 4 for ADPCM */
} WaveFMT;

/* The general chunk found in the WAVE file */
typedef struct Chunk {
	Uint32 magic;
	Uint32 length;
	Uint8 *data;
} Chunk;