/* 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 the Free Software Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ // snd_mem.c: sound caching #include "quakedef.h" int cache_full_cycle; byte *S_Alloc (int size); /* ================ ResampleSfx ================ */ void ResampleSfx (sfx_t *sfx, int inrate, int inwidth, byte *data) { int outcount; int srcsample; float stepscale; int i; int sample, samplefrac, fracstep; sfxcache_t *sc; sc = Cache_Check (&sfx->cache); if (!sc) return; stepscale = (float)inrate / shm->speed; // this is usually 0.5, 1, or 2 outcount = sc->length / stepscale; sc->length = outcount; if (sc->loopstart != -1) sc->loopstart = sc->loopstart / stepscale; sc->speed = shm->speed; if (loadas8bit.value) sc->width = 1; else sc->width = inwidth; sc->stereo = 0; // resample / decimate to the current source rate if (stepscale == 1 && inwidth == 1 && sc->width == 1) { // fast special case for (i=0 ; idata)[i] = (int)( (unsigned char)(data[i]) - 128); } else { // general case samplefrac = 0; fracstep = stepscale*256; for (i=0 ; i> 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; } } } //============================================================================= /* ============== S_LoadSound ============== */ sfxcache_t *S_LoadSound (sfx_t *s) { char namebuffer[256]; byte *data; wavinfo_t info; int len; float stepscale; sfxcache_t *sc; byte stackbuf[1*1024]; // avoid dirtying the cache heap // see if still in memory sc = Cache_Check (&s->cache); if (sc) return sc; //Con_Printf ("S_LoadSound: %x\n", (int)stackbuf); // load it in Q_strcpy(namebuffer, ""); Q_strcat(namebuffer, s->name); // Con_Printf ("loading %s\n",namebuffer); data = COM_LoadStackFile(namebuffer, stackbuf, sizeof(stackbuf)); if (!data) { Con_Printf ("Couldn't load %s\n", namebuffer); return NULL; } info = GetWavinfo (s->name, data, com_filesize); if (info.channels != 1) { Con_Printf ("%s is a stereo sample\n",s->name); return NULL; } stepscale = (float)info.rate / shm->speed; len = info.samples / stepscale; len = len * info.width * info.channels; sc = Cache_Alloc ( &s->cache, len + sizeof(sfxcache_t), s->name); if (!sc) return NULL; sc->length = info.samples; sc->loopstart = info.loopstart; sc->speed = info.rate; sc->width = info.width; sc->stereo = info.channels; ResampleSfx (s, sc->speed, sc->width, data + info.dataofs); return sc; } /* =============================================================================== WAV loading =============================================================================== */ byte *data_p; byte *iff_end; byte *last_chunk; byte *iff_data; int iff_chunk_len; short GetLittleShort(void) { short val = 0; val = *data_p; val = val + (*(data_p+1)<<8); data_p += 2; return val; } int 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; } void FindNextChunk(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 = GetLittleLong(); if (iff_chunk_len < 0) { data_p = NULL; return; } // if (iff_chunk_len > 1024*1024) // Sys_Error ("FindNextChunk: %i length is past the 1 meg sanity limit", iff_chunk_len); data_p -= 8; last_chunk = data_p + 8 + ( (iff_chunk_len + 1) & ~1 ); if (!Q_strncmp((char *)data_p, name, 4)) return; } } void FindChunk(char *name) { last_chunk = iff_data; FindNextChunk (name); } void DumpChunks(void) { char str[5]; str[4] = 0; data_p=iff_data; do { memcpy (str, data_p, 4); data_p += 4; iff_chunk_len = GetLittleLong(); Con_Printf ("0x%x : %s (%d)\n", (int)(data_p - 4), str, iff_chunk_len); data_p += (iff_chunk_len + 1) & ~1; } while (data_p < iff_end); } /* ============ GetWavinfo ============ */ wavinfo_t GetWavinfo (char *name, byte *wav, int wavlength) { wavinfo_t info; int i; int format; int samples; memset (&info, 0, sizeof(info)); if (!wav) return info; iff_data = wav; iff_end = wav + wavlength; // find "RIFF" chunk FindChunk("RIFF"); if (!(data_p && !Q_strncmp((char *)data_p+8, "WAVE", 4))) { Con_Printf("Missing RIFF/WAVE chunks\n"); return info; } // get "fmt " chunk iff_data = data_p + 12; // DumpChunks (); FindChunk("fmt "); if (!data_p) { Con_Printf("Missing fmt chunk\n"); return info; } data_p += 8; format = GetLittleShort(); if (format != 1) { Con_Printf("Microsoft PCM format only\n"); return info; } info.channels = GetLittleShort(); info.rate = GetLittleLong(); data_p += 4+2; info.width = GetLittleShort() / 8; // get cue chunk FindChunk("cue "); if (data_p) { data_p += 32; info.loopstart = GetLittleLong(); // Con_Printf("loopstart=%d\n", sfx->loopstart); // if the next chunk is a LIST chunk, look for a cue length marker FindNextChunk ("LIST"); if (data_p) { if (!strncmp ((char *)data_p + 28, "mark", 4)) { // this is not a proper parse, but it works with cooledit... data_p += 24; i = GetLittleLong (); // samples in loop info.samples = info.loopstart + i; // Con_Printf("looped length: %i\n", i); } } } else info.loopstart = -1; // find data chunk FindChunk("data"); if (!data_p) { Con_Printf("Missing data chunk\n"); return info; } data_p += 4; samples = 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; }