mirror of
https://git.code.sf.net/p/quake/quakeforge
synced 2024-11-06 13:11:20 +00:00
405 lines
8.6 KiB
C
405 lines
8.6 KiB
C
/*
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snd_mem.c
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sound caching
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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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*/
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static const char rcsid[] =
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"$Id$";
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#ifdef HAVE_STRING_H
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# include <string.h>
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#endif
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#ifdef HAVE_STRINGS_H
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# include <strings.h>
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#endif
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#include "QF/cvar.h"
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#include "QF/sound.h"
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#include "QF/sys.h"
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#include "QF/qendian.h"
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#include "QF/quakefs.h"
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int cache_full_cycle;
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sfxcache_t *
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SND_GetCache (long samples, int rate, int inwidth, int channels,
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sfx_t *sfx, cache_allocator_t allocator)
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{
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int size;
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int width;
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float stepscale;
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sfxcache_t *sc;
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width = snd_loadas8bit->int_val ? 1 : 2;
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stepscale = (float) rate / shm->speed; // usually 0.5, 1, or 2
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size = samples / stepscale;
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size *= width * channels;
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sc = allocator (&sfx->cache, size + sizeof (sfxcache_t), sfx->name);
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if (!sc)
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return 0;
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sc->length = samples;
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sc->speed = rate;
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sc->width = inwidth;
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sc->stereo = channels;
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sc->bytes = size;
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memcpy (sc->data + size, "\xde\xad\xbe\xef", 4);
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return sc;
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}
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void
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SND_ResampleSfx (sfxcache_t *sc, byte * data)
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{
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unsigned char *ib, *ob;
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int fracstep, outcount, sample, samplefrac, srcsample, i;
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float stepscale;
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short *is, *os;
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int inwidth = sc->width;
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int inrate = sc->speed;
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is = (short *) data;
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os = (short *) sc->data;
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ib = data;
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ob = sc->data;
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stepscale = (float) inrate / shm->speed; // usually 0.5, 1, or 2
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outcount = sc->length / stepscale;
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sc->speed = shm->speed;
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if (snd_loadas8bit->int_val)
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sc->width = 1;
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else
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sc->width = 2;
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sc->stereo = 0;
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// resample / decimate to the current source rate
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if (stepscale == 1) {
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if (inwidth == 1 && sc->width == 1) {
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for (i = 0; i < outcount; i++) {
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*ob++ = *ib++ - 128;
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}
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} else if (inwidth == 1 && sc->width == 2) {
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for (i = 0; i < outcount; i++) {
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*os++ = (*ib++ - 128) << 8;
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}
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} else if (inwidth == 2 && sc->width == 1) {
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for (i = 0; i < outcount; i++) {
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*ob++ = LittleShort (*is++) >> 8;
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}
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} else if (inwidth == 2 && sc->width == 2) {
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for (i = 0; i < outcount; i++) {
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*os++ = LittleShort (*is++);
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}
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}
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} else {
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// general case
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if (snd_interp->int_val && stepscale < 1) {
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int j;
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int points = 1 / stepscale;
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for (i = 0; i < sc->length; i++) {
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int s1, s2;
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if (inwidth == 2) {
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s2 = s1 = LittleShort (is[0]);
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if (i < sc->length - 1)
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s2 = LittleShort (is[1]);
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is++;
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} else {
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s2 = s1 = (ib[0] - 128) << 8;
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if (i < sc->length - 1)
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s2 = (ib[1] - 128) << 8;
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ib++;
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}
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for (j = 0; j < points; j++) {
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sample = s1 + (s2 - s1) * ((float) j) / points;
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if (sc->width == 2) {
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os[j] = sample;
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} else {
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ob[j] = sample >> 8;
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}
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}
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if (sc->width == 2) {
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os += points;
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} else {
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ob += points;
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}
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}
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} else {
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samplefrac = 0;
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fracstep = stepscale * 256;
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for (i = 0; i < outcount; i++) {
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srcsample = samplefrac >> 8;
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samplefrac += fracstep;
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if (inwidth == 2)
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sample = LittleShort (((short *) data)[srcsample]);
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else
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sample =
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(int) ((unsigned char) (data[srcsample]) - 128) << 8;
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if (sc->width == 2)
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((short *) sc->data)[i] = sample;
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else
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((signed char *) sc->data)[i] = sample >> 8;
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}
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}
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}
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sc->length = outcount;
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if (sc->loopstart != -1)
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sc->loopstart = sc->loopstart / stepscale;
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if (memcmp (sc->data + sc->bytes, "\xde\xad\xbe\xef", 4))
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Sys_Error ("SND_ResampleSfx screwed the pooch: %02x%02x%02x%02x",
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sc->data[sc->bytes + 0], sc->data[sc->bytes + 1],
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sc->data[sc->bytes + 2], sc->data[sc->bytes + 3]);
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}
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static sfxcache_t *
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SND_LoadSound (sfx_t *sfx, cache_allocator_t allocator)
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{
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char namebuffer[256];
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char foundname[256];
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byte *data;
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wavinfo_t info;
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int len;
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float stepscale;
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sfxcache_t *sc;
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byte stackbuf[1 * 1024]; // avoid dirtying the cache heap
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QFile *file;
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// load it in
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strcpy (namebuffer, "sound/");
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strncat (namebuffer, sfx->name, sizeof (namebuffer) - strlen (namebuffer));
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_COM_FOpenFile (namebuffer, &file, foundname, 1);
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if (!file) {
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Sys_Printf ("Couldn't load %s\n", namebuffer);
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return 0;
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}
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if (strcmp (".ogg", COM_FileExtension (foundname)) == 0) {
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return SND_LoadOgg (file, sfx, allocator);
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}
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Qclose (file); //FIXME this is a dumb way to do this
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data = COM_LoadStackFile (namebuffer, stackbuf, sizeof (stackbuf));
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if (!data) {
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Sys_Printf ("Couldn't load %s\n", namebuffer);
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return NULL;
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}
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info = SND_GetWavinfo (sfx->name, data, com_filesize);
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if (info.channels != 1) {
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Sys_Printf ("%s is a stereo sample\n", sfx->name);
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return NULL;
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}
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stepscale = (float) info.rate / shm->speed;
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len = info.samples / stepscale;
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if (snd_loadas8bit->int_val) {
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len = len * info.channels;
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} else {
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len = len * 2 * info.channels;
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}
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sc = SND_GetCache (info.samples, info.rate, info.width, info.channels,
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sfx, allocator);
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if (!sc)
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return NULL;
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sc->loopstart = info.loopstart;
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SND_ResampleSfx (sc, data + info.dataofs);
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return sc;
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}
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void
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SND_CallbackLoad (void *object, cache_allocator_t allocator)
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{
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SND_LoadSound (object, allocator);
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}
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/* WAV loading */
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byte *data_p;
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byte *iff_end;
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byte *last_chunk;
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byte *iff_data;
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int iff_chunk_len;
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static short
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SND_GetLittleShort (void)
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{
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short val = 0;
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val = *data_p;
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val = val + (*(data_p + 1) << 8);
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data_p += 2;
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return val;
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}
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static int
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SND_GetLittleLong (void)
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{
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int val = 0;
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val = *data_p;
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val = val + (*(data_p + 1) << 8);
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val = val + (*(data_p + 2) << 16);
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val = val + (*(data_p + 3) << 24);
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data_p += 4;
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return val;
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}
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static void
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SND_FindNexctChunk (const char *name)
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{
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while (1) {
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data_p = last_chunk;
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if (data_p >= iff_end) { // didn't find the chunk
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data_p = NULL;
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return;
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}
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data_p += 4;
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iff_chunk_len = SND_GetLittleLong ();
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if (iff_chunk_len < 0) {
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data_p = NULL;
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return;
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}
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data_p -= 8;
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last_chunk = data_p + 8 + ((iff_chunk_len + 1) & ~1);
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if (!strncmp (data_p, name, 4))
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return;
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}
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}
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static void
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SND_FindChunk (const char *name)
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{
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last_chunk = iff_data;
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SND_FindNexctChunk (name);
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}
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/*
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static void
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SND_DumpChunks (void)
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{
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char str[5];
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str[4] = 0;
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data_p = iff_data;
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do {
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memcpy (str, data_p, 4);
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data_p += 4;
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iff_chunk_len = SND_GetLittleLong ();
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Sys_Printf ("0x%lx : %s (%d)\n", (long) (data_p - 4), str,
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iff_chunk_len);
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data_p += (iff_chunk_len + 1) & ~1;
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} while (data_p < iff_end);
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}
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*/
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wavinfo_t
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SND_GetWavinfo (const char *name, byte * wav, int wavlength)
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{
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int format, samples, i;
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wavinfo_t info;
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memset (&info, 0, sizeof (info));
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if (!wav)
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return info;
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iff_data = wav;
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iff_end = wav + wavlength;
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// find "RIFF" chunk
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SND_FindChunk ("RIFF");
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if (!(data_p && !strncmp (data_p + 8, "WAVE", 4))) {
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Sys_Printf ("Missing RIFF/WAVE chunks\n");
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return info;
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}
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// get "fmt " chunk
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iff_data = data_p + 12;
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// SND_DumpChunks ();
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SND_FindChunk ("fmt ");
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if (!data_p) {
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Sys_Printf ("Missing fmt chunk\n");
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return info;
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}
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data_p += 8;
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format = SND_GetLittleShort ();
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if (format != 1) {
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Sys_Printf ("Microsoft PCM format only\n");
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return info;
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}
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info.channels = SND_GetLittleShort ();
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info.rate = SND_GetLittleLong ();
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data_p += 4 + 2;
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info.width = SND_GetLittleShort () / 8;
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// get cue chunk
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SND_FindChunk ("cue ");
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if (data_p) {
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data_p += 32;
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info.loopstart = SND_GetLittleLong ();
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// if the next chunk is a LIST chunk, look for a cue length marker
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SND_FindNexctChunk ("LIST");
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if (data_p) {
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if (!strncmp (data_p + 28, "mark", 4)) {
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// this is not a proper parse, but it works with cooledit...
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data_p += 24;
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i = SND_GetLittleLong (); // samples in loop
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info.samples = info.loopstart + i;
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}
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}
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} else
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info.loopstart = -1;
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// find data chunk
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SND_FindChunk ("data");
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if (!data_p) {
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Sys_Printf ("Missing data chunk\n");
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return info;
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}
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data_p += 4;
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samples = SND_GetLittleLong () / info.width;
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if (info.samples) {
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if (samples < info.samples)
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Sys_Error ("Sound %s has a bad loop length", name);
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} else
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info.samples = samples;
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info.dataofs = data_p - wav;
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return info;
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}
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