/* snd_mem.c sound caching Copyright (C) 1996-1997 Id Software, Inc. This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. This program 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 General Public License for more details. You should have received a copy of the GNU General Public License along with this program; if not, write to: Free Software Foundation, Inc. 59 Temple Place - Suite 330 Boston, MA 02111-1307, USA */ #ifdef HAVE_CONFIG_H # include "config.h" #endif static __attribute__ ((unused)) const char rcsid[] = "$Id$"; #ifdef HAVE_STRING_H # include #endif #ifdef HAVE_STRINGS_H # include #endif #include "QF/cvar.h" #include "QF/dstring.h" #include "QF/sound.h" #include "QF/sys.h" #include "QF/qendian.h" #include "QF/quakefs.h" #include "snd_render.h" int cache_full_cycle; sfxcache_t * SND_GetCache (long samples, int rate, int inwidth, int channels, sfx_t *sfx, cache_allocator_t allocator) { int size; int width; float stepscale; sfxcache_t *sc; width = snd_loadas8bit->int_val ? 1 : 2; stepscale = (float) rate / shm->speed; // usually 0.5, 1, or 2 size = samples / stepscale; size *= width * channels; sc = allocator (&sfx->cache, size + sizeof (sfxcache_t), sfx->name); if (!sc) return 0; sc->length = samples; sc->speed = rate; sc->width = inwidth; sc->stereo = channels; sc->bytes = size; memcpy (sc->data + size, "\xde\xad\xbe\xef", 4); return sc; } void SND_ResampleSfx (sfxcache_t *sc, byte * data) { unsigned char *ib, *ob; int fracstep, outcount, sample, samplefrac, srcsample, i; float stepscale; short *is, *os; int inwidth = sc->width; int inrate = sc->speed; is = (short *) data; os = (short *) sc->data; ib = data; ob = sc->data; stepscale = (float) inrate / shm->speed; // usually 0.5, 1, or 2 outcount = sc->length / stepscale; sc->speed = shm->speed; if (snd_loadas8bit->int_val) sc->width = 1; else sc->width = 2; sc->stereo = 0; // resample / decimate to the current source rate if (stepscale == 1) { if (inwidth == 1 && sc->width == 1) { for (i = 0; i < outcount; i++) { *ob++ = *ib++ - 128; } } else if (inwidth == 1 && sc->width == 2) { for (i = 0; i < outcount; i++) { *os++ = (*ib++ - 128) << 8; } } else if (inwidth == 2 && sc->width == 1) { for (i = 0; i < outcount; i++) { *ob++ = LittleShort (*is++) >> 8; } } else if (inwidth == 2 && sc->width == 2) { for (i = 0; i < outcount; i++) { *os++ = LittleShort (*is++); } } } else { // general case if (snd_interp->int_val && stepscale < 1) { int j; int points = 1 / stepscale; for (i = 0; i < sc->length; i++) { int s1, s2; if (inwidth == 2) { s2 = s1 = LittleShort (is[0]); if (i < sc->length - 1) s2 = LittleShort (is[1]); is++; } else { s2 = s1 = (ib[0] - 128) << 8; if (i < sc->length - 1) s2 = (ib[1] - 128) << 8; ib++; } for (j = 0; j < points; j++) { sample = s1 + (s2 - s1) * ((float) j) / points; if (sc->width == 2) { os[j] = sample; } else { ob[j] = sample >> 8; } } if (sc->width == 2) { os += points; } else { ob += points; } } } else { samplefrac = 0; fracstep = stepscale * 256; for (i = 0; i < outcount; i++) { srcsample = samplefrac >> 8; samplefrac += fracstep; if (inwidth == 2) sample = LittleShort (((short *) data)[srcsample]); else sample = (int) ((unsigned char) (data[srcsample]) - 128) << 8; if (sc->width == 2) ((short *) sc->data)[i] = sample; else ((signed char *) sc->data)[i] = sample >> 8; } } } sc->length = outcount; if (sc->loopstart != -1) sc->loopstart = sc->loopstart / stepscale; if (memcmp (sc->data + sc->bytes, "\xde\xad\xbe\xef", 4)) Sys_Error ("SND_ResampleSfx screwed the pooch: %02x%02x%02x%02x", sc->data[sc->bytes + 0], sc->data[sc->bytes + 1], sc->data[sc->bytes + 2], sc->data[sc->bytes + 3]); } static sfxcache_t * SND_LoadSound (sfx_t *sfx, cache_allocator_t allocator) { char namebuffer[256]; dstring_t *foundname = dstring_new (); byte *data; wavinfo_t info; int len; float stepscale; sfxcache_t *sc; byte stackbuf[1 * 1024]; // avoid dirtying the cache heap QFile *file; // load it in strcpy (namebuffer, "sound/"); strncat (namebuffer, sfx->name, sizeof (namebuffer) - strlen (namebuffer)); _QFS_FOpenFile (namebuffer, &file, foundname, 1); if (!file) { dstring_delete (foundname); Sys_Printf ("Couldn't load %s\n", namebuffer); return 0; } if (strcmp (".ogg", QFS_FileExtension (foundname->str)) == 0) { dstring_delete (foundname); return SND_LoadOgg (file, sfx, allocator); } dstring_delete (foundname); Qclose (file); //FIXME this is a dumb way to do this data = QFS_LoadStackFile (namebuffer, stackbuf, sizeof (stackbuf)); if (!data) { Sys_Printf ("Couldn't load %s\n", namebuffer); return NULL; } info = SND_GetWavinfo (sfx->name, data, qfs_filesize); if (info.channels != 1) { Sys_Printf ("%s is a stereo sample\n", sfx->name); return NULL; } stepscale = (float) info.rate / shm->speed; len = info.samples / stepscale; if (snd_loadas8bit->int_val) { len = len * info.channels; } else { len = len * 2 * info.channels; } sc = SND_GetCache (info.samples, info.rate, info.width, info.channels, sfx, allocator); if (!sc) return NULL; sc->loopstart = info.loopstart; SND_ResampleSfx (sc, data + info.dataofs); return sc; } void SND_CallbackLoad (void *object, cache_allocator_t allocator) { SND_LoadSound (object, allocator); } /* WAV loading */ byte *data_p; byte *iff_end; byte *last_chunk; byte *iff_data; int iff_chunk_len; static short SND_GetLittleShort (void) { short val = 0; val = *data_p; val = val + (*(data_p + 1) << 8); data_p += 2; return val; } static int SND_GetLittleLong (void) { int val = 0; val = *data_p; val = val + (*(data_p + 1) << 8); val = val + (*(data_p + 2) << 16); val = val + (*(data_p + 3) << 24); data_p += 4; return val; } static void SND_FindNexctChunk (const char *name) { while (1) { data_p = last_chunk; if (data_p >= iff_end) { // didn't find the chunk data_p = NULL; return; } data_p += 4; iff_chunk_len = SND_GetLittleLong (); if (iff_chunk_len < 0) { data_p = NULL; return; } data_p -= 8; last_chunk = data_p + 8 + ((iff_chunk_len + 1) & ~1); if (!strncmp (data_p, name, 4)) return; } } static void SND_FindChunk (const char *name) { last_chunk = iff_data; SND_FindNexctChunk (name); } /* static void SND_DumpChunks (void) { char str[5]; str[4] = 0; data_p = iff_data; do { memcpy (str, data_p, 4); data_p += 4; iff_chunk_len = SND_GetLittleLong (); Sys_Printf ("0x%lx : %s (%d)\n", (long) (data_p - 4), str, iff_chunk_len); data_p += (iff_chunk_len + 1) & ~1; } while (data_p < iff_end); } */ wavinfo_t SND_GetWavinfo (const char *name, byte * wav, int wavlength) { int format, samples, i; wavinfo_t info; memset (&info, 0, sizeof (info)); if (!wav) return info; iff_data = wav; iff_end = wav + wavlength; // find "RIFF" chunk SND_FindChunk ("RIFF"); if (!(data_p && !strncmp (data_p + 8, "WAVE", 4))) { Sys_Printf ("Missing RIFF/WAVE chunks\n"); return info; } // get "fmt " chunk iff_data = data_p + 12; // SND_DumpChunks (); SND_FindChunk ("fmt "); if (!data_p) { Sys_Printf ("Missing fmt chunk\n"); return info; } data_p += 8; format = SND_GetLittleShort (); if (format != 1) { Sys_Printf ("Microsoft PCM format only\n"); return info; } info.channels = SND_GetLittleShort (); info.rate = SND_GetLittleLong (); data_p += 4 + 2; info.width = SND_GetLittleShort () / 8; // get cue chunk SND_FindChunk ("cue "); if (data_p) { data_p += 32; info.loopstart = SND_GetLittleLong (); // if the next chunk is a LIST chunk, look for a cue length marker SND_FindNexctChunk ("LIST"); if (data_p) { if (!strncmp (data_p + 28, "mark", 4)) { // this is not a proper parse, but it works with cooledit... data_p += 24; i = SND_GetLittleLong (); // samples in loop info.samples = info.loopstart + i; } } } else info.loopstart = -1; // find data chunk SND_FindChunk ("data"); if (!data_p) { Sys_Printf ("Missing data chunk\n"); return info; } data_p += 4; samples = SND_GetLittleLong () / info.width; if (info.samples) { if (samples < info.samples) Sys_Error ("Sound %s has a bad loop length", name); } else info.samples = samples; info.dataofs = data_p - wav; return info; }