mirror of
https://git.code.sf.net/p/quake/newtree
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313 lines
6.6 KiB
C
313 lines
6.6 KiB
C
/*
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snd_sgi.c
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sound support for sgi
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Copyright (C) 1996-1997 Id Software, Inc.
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This program is free software; you can redistribute it and/or
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modify it under the terms of the GNU General Public License
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as published by the Free Software Foundation; either version 2
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of the License, or (at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
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See the GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to:
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Free Software Foundation, Inc.
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59 Temple Place - Suite 330
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Boston, MA 02111-1307, USA
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$Id$
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <errno.h>
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#include <limits.h>
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#include <dmedia/audio.h>
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#include "qtypes.h"
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#include "qargs.h"
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#include "sound.h"
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#include "console.h"
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static int snd_inited = 0;
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static ALconfig alc;
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static ALport alp;
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static int tryrates[] = { 11025, 22050, 44100, 8000 };
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static unsigned char *dma_buffer, *write_buffer;
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static int bufsize;
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static int wbufp;
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static int framecount;
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qboolean
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SNDDMA_Init (void)
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{
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ALpv alpv;
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int i;
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char *s;
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alc = alNewConfig ();
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if (!alc) {
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Con_Printf ("Could not make an new sound config: %s\n",
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alGetErrorString (oserror ()));
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return 0;
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}
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shm = &sn;
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shm->splitbuffer = 0;
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/* get & probe settings */
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/* sample format */
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if (alSetSampFmt (alc, AL_SAMPFMT_TWOSCOMP) < 0) {
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Con_Printf ("Could not sample format of default output to two's "
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"complement\n");
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alFreeConfig (alc);
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return 0;
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}
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/* sample bits */
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s = getenv ("QUAKE_SOUND_SAMPLEBITS");
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if (s)
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shm->samplebits = atoi (s);
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else if ((i = COM_CheckParm ("-sndbits")) != 0)
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shm->samplebits = atoi (com_argv[i + 1]);
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if (shm->samplebits != 16 && shm->samplebits != 8) {
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alpv.param = AL_WORDSIZE;
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if (alGetParams (AL_DEFAULT_OUTPUT, &alpv, 1) < 0) {
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Con_Printf ("Could not get supported wordsize of default "
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"output: %s\n", alGetErrorString (oserror ()));
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return 0;
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}
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if (alpv.value.i >= 16) {
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shm->samplebits = 16;
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} else {
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if (alpv.value.i >= 8)
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shm->samplebits = 8;
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else {
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Con_Printf ("Sound disabled since interface "
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"doesn't even support 8 bit.");
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alFreeConfig (alc);
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return 0;
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}
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}
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}
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/* sample rate */
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s = getenv ("QUAKE_SOUND_SPEED");
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if (s)
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shm->speed = atoi (s);
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else if ((i = COM_CheckParm ("-sndspeed")) != 0)
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shm->speed = atoi (com_argv[i + 1]);
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else {
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alpv.param = AL_RATE;
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for (i = 0; i < sizeof (tryrates) / sizeof (int); i++) {
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alpv.value.ll = alDoubleToFixed (tryrates[i]);
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if (alSetParams (AL_DEFAULT_OUTPUT, &alpv, 1) >= 0)
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break;
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}
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if (i >= sizeof (tryrates) / sizeof (int)) {
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Con_Printf ("Sound disabled since interface doesn't even "
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"support a sample rate of %d\n", tryrates[i - 1]);
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alFreeConfig (alc);
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return 0;
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}
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shm->speed = tryrates[i];
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}
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/* channels */
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s = getenv ("QUAKE_SOUND_CHANNELS");
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if (s)
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shm->channels = atoi (s);
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else if ((i = COM_CheckParm ("-sndmono")) != 0)
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shm->channels = 1;
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else if ((i = COM_CheckParm ("-sndstereo")) != 0)
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shm->channels = 2;
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else
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shm->channels = 2;
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/* set 'em */
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/* channels */
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while (shm->channels > 0) {
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if (alSetChannels (alc, shm->channels) < 0) {
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Con_Printf ("Unable to set number of channels to %d, trying half\n",
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shm->channels);
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shm->channels /= 2;
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} else
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break;
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}
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if (shm->channels <= 0) {
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Con_Printf ("Sound disabled since interface doesn't even support 1 "
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"channel\n");
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alFreeConfig (alc);
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return 0;
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}
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/* sample rate */
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alpv.param = AL_RATE;
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alpv.value.ll = alDoubleToFixed (shm->speed);
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if (alSetParams (AL_DEFAULT_OUTPUT, &alpv, 1) < 0) {
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Con_Printf ("Could not set samplerate of default output to %d: %s\n",
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shm->speed, alGetErrorString (oserror ()));
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alFreeConfig (alc);
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return 0;
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}
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/* set sizes of buffers relative to sizes of those for ** the 'standard'
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frequency of 11025 ** ** use *huge* buffers since at least my indigo2
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has enough ** to do to get sound on the way anyway */
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bufsize = 32768 * (int) ((double) shm->speed / 11025.0);
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dma_buffer = malloc (bufsize);
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if (dma_buffer == NULL) {
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Con_Printf ("Could not get %d bytes of memory for audio dma buffer\n",
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bufsize);
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alFreeConfig (alc);
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return 0;
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}
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write_buffer = malloc (bufsize);
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if (write_buffer == NULL) {
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Con_Printf ("Could not get %d bytes of memory for audio write buffer\n",
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bufsize);
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free (dma_buffer);
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alFreeConfig (alc);
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return 0;
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}
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/* sample bits */
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switch (shm->samplebits) {
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case 24:
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i = AL_SAMPLE_24;
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break;
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case 16:
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i = AL_SAMPLE_16;
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break;
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default:
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i = AL_SAMPLE_8;
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break;
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}
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if (alSetWidth (alc, i) < 0) {
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Con_Printf ("Could not set wordsize of default output to %d: %s\n",
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shm->samplebits, alGetErrorString (oserror ()));
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free (write_buffer);
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free (dma_buffer);
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alFreeConfig (alc);
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return 0;
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}
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alp = alOpenPort ("quakeforge", "w", alc);
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if (!alp) {
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Con_Printf ("Could not open sound port: %s\n",
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alGetErrorString (oserror ()));
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free (write_buffer);
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free (dma_buffer);
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alFreeConfig (alc);
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return 0;
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}
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shm->soundalive = true;
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shm->samples = bufsize / (shm->samplebits / 8);
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shm->samplepos = 0;
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shm->submission_chunk = 1;
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shm->buffer = dma_buffer;
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framecount = 0;
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snd_inited = 1;
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return 1;
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}
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int
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SNDDMA_GetDMAPos (void)
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{
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/* Con_Printf("framecount: %d %d\n", (framecount * shm->channels) %
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shm->samples, alGetFilled(alp)); */
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shm->samplepos = ((framecount - alGetFilled (alp))
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* shm->channels) % shm->samples;
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return shm->samplepos;
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}
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void
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SNDDMA_Shutdown (void)
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{
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if (snd_inited) {
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free (write_buffer);
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free (dma_buffer);
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alClosePort (alp);
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alFreeConfig (alc);
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snd_inited = 0;
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}
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}
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/*
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==============
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SNDDMA_Submit
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Send sound to device if buffer isn't really the dma buffer
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===============
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*/
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void
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SNDDMA_Submit (void)
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{
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int bsize;
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int bytes, b;
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unsigned char *p;
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int idx;
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int stop = paintedtime;
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if (paintedtime < wbufp)
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wbufp = 0; // reset
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bsize = shm->channels * (shm->samplebits / 8);
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bytes = (paintedtime - wbufp) * bsize;
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if (!bytes)
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return;
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if (bytes > bufsize) {
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bytes = bufsize;
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stop = wbufp + bytes / bsize;
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}
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p = write_buffer;
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idx = (wbufp * bsize) & (bufsize - 1);
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for (b = bytes; b; b--) {
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*p++ = dma_buffer[idx];
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idx = (idx + 1) & (bufsize - 1);
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}
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wbufp = stop;
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alWriteFrames (alp, write_buffer, bytes / bsize);
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framecount += bytes / bsize;
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}
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/* end of file */
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