/* =========================================================================== Copyright (C) 1997-2001 Id Software, Inc. This file is part of Quake 2 source code. Quake 2 source code 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. Quake 2 source code 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 Quake 2 source code; if not, write to the Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA =========================================================================== */ // snd_mem.c: sound caching #include "client.h" #include "snd_loc.h" int cache_full_cycle; byte *S_Alloc (int size); /* ================ ResampleSfx ================ */ // Knightmare /*void New_ResampleSfx (sfxcache_t *sc, byte *data, char *name) { int i, outcount, srcsample, srclength, samplefrac, fracstep; float scale = (float) sc->speed / (float) dma.speed; // this is usually 0.5 (128), 1 (256), or 2 (512) fracstep = ((double) sc->speed / (double) dma.speed) * 256.0; srclength = sc->length << sc->stereo; outcount = (double) sc->length * (double) dma.speed / (double) sc->speed; sc->length = outcount; if (sc->loopstart != -1) sc->loopstart = (double) sc->loopstart * (double) dma.speed / (double) sc->speed; sc->speed = dma.speed; // resample / decimate to the current source rate if (fracstep == 256) { if (sc->width == 1) // 8bit for (i = 0;i < srclength;i++) ((signed char *)sc->data)[i] = ((unsigned char *)data)[i] - 128; else for (i = 0;i < srclength;i++) ((short *)sc->data)[i] = LittleShort (((short *)data)[i]); //crashes here } else { // general case Com_DPrintf("ResampleSfx: resampling sound %s\n", name); samplefrac = 0; if ((fracstep & 255) == 0) // skipping points on perfect multiple { srcsample = 0; fracstep >>= 8; if (sc->width == 2) { short *out = (void *)sc->data, *in = (void *)data; if (sc->stereo) { fracstep <<= 1; for (i=0 ; idata; unsigned char *in = (void *)data; if (sc->stereo) { fracstep <<= 1; for (i=0 ; iwidth == 2) { short *out = (void *)sc->data, *in = (void *)data; if (sc->stereo) { for (i=0 ; i> 8) << 1; a = in[srcsample ]; if (srcsample+2 >= srclength) b = 0; else b = in[srcsample+2]; sample = (((b - a) * (samplefrac & 255)) >> 8) + a; *out++ = (short) sample; a = in[srcsample+1]; if (srcsample+2 >= srclength) b = 0; else b = in[srcsample+3]; sample = (((b - a) * (samplefrac & 255)) >> 8) + a; *out++ = (short) sample; samplefrac += fracstep; } } else { for (i=0 ; i> 8; a = in[srcsample ]; if (srcsample+1 >= srclength) b = 0; else b = in[srcsample+1]; sample = (((b - a) * (samplefrac & 255)) >> 8) + a; *out++ = (short) sample; samplefrac += fracstep; } } } else { signed char *out = (void *)sc->data; unsigned char *in = (void *)data; if (sc->stereo) // LordHavoc: stereo sound support { for (i=0 ; i> 8) << 1; a = (int) in[srcsample ] - 128; if (srcsample+2 >= srclength) b = 0; else b = (int) in[srcsample+2] - 128; sample = (((b - a) * (samplefrac & 255)) >> 8) + a; *out++ = (signed char) sample; a = (int) in[srcsample+1] - 128; if (srcsample+2 >= srclength) b = 0; else b = (int) in[srcsample+3] - 128; sample = (((b - a) * (samplefrac & 255)) >> 8) + a; *out++ = (signed char) sample; samplefrac += fracstep; } } else { for (i=0 ; i> 8; a = (int) in[srcsample ] - 128; if (srcsample+1 >= srclength) b = 0; else b = (int) in[srcsample+1] - 128; sample = (((b - a) * (samplefrac & 255)) >> 8) + a; *out++ = (signed char) sample; samplefrac += fracstep; } } } } } }*/ 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 = sfx->cache; if (!sc) return; stepscale = (float)inrate / dma.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 = dma.speed; if (s_loadas8bit->value) sc->width = 1; else sc->width = inwidth; // Knightmare //sc->stereo = 0; // resample / decimate to the current source rate if (stepscale == 1 && inwidth == 1 && sc->width == 1) { // fast special case for (i = 0; i < outcount; i++) ((signed char *)sc->data)[i] = (int)( (unsigned char)(data[i]) - 128); } else { // general case 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; } } } //============================================================================= /* ============== S_LoadSound ============== */ sfxcache_t *S_LoadSound (sfx_t *s) { char namebuffer[MAX_QPATH]; byte *data; wavinfo_t info; int len; float stepscale; sfxcache_t *sc; int size; char *name; if (s->name[0] == '*') return NULL; // see if still in memory sc = s->cache; if (sc) return sc; // load it in if (s->truename) name = s->truename; else name = s->name; if (name[0] == '#') // strncpy(namebuffer, &name[1]); Q_strncpyz (namebuffer, sizeof(namebuffer), &name[1]); else Com_sprintf (namebuffer, sizeof(namebuffer), "sound/%s", name); // Com_Printf ("loading %s\n",namebuffer); size = FS_LoadFile (namebuffer, (void **)&data); if (!data) { Com_DPrintf ("Couldn't load %s\n", namebuffer); return NULL; } info = GetWavinfo (s->name, data, size); /*if (info.channels != 1) { Com_Printf ("%s is a stereo sample\n",s->name); FS_FreeFile (data); return NULL; }*/ if (info.channels < 1 || info.channels > 2) //CDawg changed { Com_Printf ("%s has an invalid number of channels\n", s->name); FS_FreeFile (data); return NULL; } // calculate resampled length stepscale = (float)info.rate / dma.speed; len = info.samples / stepscale; len = len * info.width * info.channels; sc = s->cache = Z_Malloc (len + sizeof(sfxcache_t)); if (!sc) { FS_FreeFile (data); return NULL; } sc->length = info.samples; sc->loopstart = info.loopstart; //sc->speed = info.rate; sc->speed = info.rate * info.channels; //CDawg changed sc->width = info.width; sc->stereo = info.channels; sc->music = !strncmp (namebuffer, "music/", 6); // force loopstart if it's a music file if ( sc->music && (sc->loopstart == -1) ) sc->loopstart = 0; ResampleSfx (s, sc->speed, sc->width, data + info.dataofs); FS_FreeFile (data); 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 (!strncmp(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(); // Com_Printf ("0x%x : %s (%d)\n", (int)(data_p - 4), str, iff_chunk_len); Com_Printf ("0x%x : %s (%d)\n", (size_t)(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 && !strncmp(data_p+8, "WAVE", 4))) { Com_Printf("Missing RIFF/WAVE chunks\n"); return info; } // get "fmt " chunk iff_data = data_p + 12; // DumpChunks (); FindChunk("fmt "); if (!data_p) { Com_Printf("Missing fmt chunk\n"); return info; } data_p += 8; format = GetLittleShort(); if (format != 1) { Com_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(); // Com_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 (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; // Com_Printf("looped length: %i\n", i); } } } else info.loopstart = -1; // find data chunk FindChunk("data"); if (!data_p) { Com_Printf("Missing data chunk\n"); return info; } data_p += 4; samples = GetLittleLong () / info.width; if (info.samples) { if (samples < info.samples) Com_Error (ERR_DROP, "Sound %s has a bad loop length", name); } else info.samples = samples; info.dataofs = data_p - wav; return info; }