/* 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" #include "winquake.h" int cache_full_cycle; qbyte *S_Alloc (int size); /* ================ ResampleSfx ================ */ void ResampleSfx (sfx_t *sfx, int inrate, int inwidth, qbyte *data) { int incount; int outcount; int srcsample; float stepscale; int i; int sample, fracstep; unsigned int samplefrac; sfxcache_t *sc; extern cvar_t snd_linearresample; qboolean linearsampling = !!(snd_linearresample.value); sc = Cache_Check (&sfx->cache); if (!sc) return; stepscale = (float)inrate / snd_speed; // this is usually 0.5, 1, or 2 incount = sc->length; outcount = sc->length / stepscale; sc->length = outcount; if (sc->loopstart != -1) sc->loopstart = sc->loopstart / stepscale; sc->speed = snd_speed; if (loadas8bit.value) sc->width = 1; else sc->width = inwidth; if (sc->numchannels==2) { if (stepscale == 1 && inwidth == 1 && sc->width == 1) { outcount*=2; // fast special case for (i=0 ; idata)[i] = (int)( (unsigned char)(data[i]) - 128); } else if (stepscale == 1 && inwidth == 2 && sc->width == 2) { outcount*=2; // fast special case for (i=0 ; idata)[i] = LittleShort ( ((short *)data)[i] ); } else { // general case samplefrac = 0; fracstep = stepscale*256; for (i=0 ; i> 8; linearsampling = linearsampling && ((srcsample+1) < incount); if (linearsampling) { if (inwidth == 2) sample = ((255 - (samplefrac & 0xFF)) * LittleShort( ((short *)data)[ srcsample<<1 ] ) + (samplefrac & 0xFF) * LittleShort( ((short *)data)[ (srcsample+1)<<1 ] )) >> 8; else sample = ((255-(samplefrac & 0xFF)) * (int)( (unsigned char)(data[ srcsample<<1 ]) - 128 ) + (samplefrac & 0xFF) * (int)( (unsigned char)(data[ (srcsample+1)<<1 ]) - 128 )); } else { if (inwidth == 2) sample = LittleShort ( ((short *)data)[(srcsample<<1)] ); else sample = (int)( (unsigned char)(data[(srcsample<<1)]) - 128) << 8; } if (sc->width == 2) ((short *)sc->data)[i<<1] = sample; else ((signed char *)sc->data)[i<<1] = sample >> 8; // srcsample = samplefrac >> 8; // samplefrac += fracstep; if (linearsampling) { if (inwidth == 2) sample = ((255 - (samplefrac & 0xFF)) * LittleShort( ((short *)data)[ (srcsample<<1)+1 ] ) + (samplefrac & 0xFF) * LittleShort( ((short *)data)[ ((srcsample+1)<<1)+1 ] )) >> 8; else sample = ((255-(samplefrac & 0xFF)) * (int)( (unsigned char)(data[ (srcsample<<1)+1 ]) - 128 ) + (samplefrac & 0xFF) * (int)( (unsigned char)(data[ ((srcsample+1)<<1)+1 ]) - 128 )); } else { if (inwidth == 2) sample = LittleShort ( ((short *)data)[(srcsample<<1)+1] ); else sample = (int)( (unsigned char)(data[(srcsample<<1)+1]) - 128) << 8; } if (sc->width == 2) ((short *)sc->data)[(i<<1)+1] = sample; else ((signed char *)sc->data)[(i<<1)+1] = sample >> 8; samplefrac += fracstep; } } return; } // 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 if (stepscale == 1 && inwidth == 2 && sc->width == 2) { // fast special case for (i=0 ; idata)[i] = LittleShort ( ((short *)data)[i] ); } else { // general case samplefrac = 0; fracstep = stepscale*256; for (i=0 ; i> 8; linearsampling = linearsampling && ((srcsample+1) < incount); if (linearsampling) { if (inwidth == 2) sample = ((255 - (samplefrac & 0xFF)) * LittleShort( ((short *)data)[srcsample] ) + (samplefrac & 0xFF) * LittleShort( ((short *)data)[srcsample+1] )) >> 8; else sample = ((255-(samplefrac & 0xFF)) * (int)( (unsigned char)(data[srcsample]) - 128 ) + (samplefrac & 0xFF) * (int)( (unsigned char)(data[srcsample+1]) - 128 )); } else { 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; samplefrac += fracstep; } } } //============================================================================= #ifdef DOOMWADS // needs fine tuning.. educated guesses #define DSPK_RATE 128 #define DSPK_FREQ 31 sfxcache_t *S_LoadDoomSpeakerSound (sfx_t *s, qbyte *data, int datalen, int sndspeed) { sfxcache_t *sc; // format data from Unofficial Doom Specs v1.6 unsigned short *dataus; int samples, len, timeraccum, inrate, inaccum; qbyte *outdata; qbyte towrite; if (datalen < 4) return NULL; dataus = (unsigned short*)data; if (LittleShort(dataus[0]) != 0) return NULL; samples = LittleShort(dataus[1]); data += 4; datalen -= 4; if (datalen != samples) return NULL; len = (int)((double)samples * (double)snd_speed / DSPK_RATE); sc = Cache_Alloc (&s->cache, len + sizeof(sfxcache_t), s->name); if (!sc) { return NULL; } sc->length = len; sc->loopstart = -1; sc->numchannels = 1; sc->width = 1; sc->speed = snd_speed; timeraccum = 0; outdata = sc->data; towrite = 0x40; inrate = (int)((double)snd_speed / DSPK_RATE); inaccum = inrate; while (len > 0) { timeraccum += *data * DSPK_FREQ; if (timeraccum > snd_speed) { towrite ^= 0xFF; // swap speaker component timeraccum -= snd_speed; } inaccum--; if (!inaccum) { data++; inaccum = inrate; } *outdata = towrite; outdata++; len--; } return sc; } sfxcache_t *S_LoadDoomSound (sfx_t *s, qbyte *data, int datalen, int sndspeed) { sfxcache_t *sc; // format data from Unofficial Doom Specs v1.6 unsigned short *dataus; int samples, rate, len; if (datalen < 8) return NULL; dataus = (unsigned short*)data; if (LittleShort(dataus[0]) != 3) return NULL; rate = LittleShort(dataus[1]); samples = LittleShort(dataus[2]); data += 8; datalen -= 8; if (datalen != samples) return NULL; len = (int)((double)samples * (double)snd_speed / (double)rate); sc = Cache_Alloc (&s->cache, len + sizeof(sfxcache_t), s->name); if (!sc) { return NULL; } sc->length = samples; sc->loopstart = -1; sc->numchannels = 1; sc->width = 1; sc->speed = rate; ResampleSfx (s, sc->speed, sc->width, data); return sc; } #endif sfxcache_t *S_LoadWavSound (sfx_t *s, qbyte *data, int datalen, int sndspeed) { wavinfo_t info; int len; sfxcache_t *sc; info = GetWavinfo (s->name, data, datalen); if (info.numchannels < 1 || info.numchannels > 2) { Con_Printf ("%s has an unsupported quantity of channels.\n",s->name); return NULL; } len = (int) ((double) info.samples * (double) snd_speed / (double) info.rate); len = len * info.width * info.numchannels; 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->numchannels = info.numchannels; ResampleSfx (s, sc->speed, sc->width, data + info.dataofs); return sc; } sfxcache_t *S_LoadOVSound (sfx_t *s, qbyte *data, int datalen, int sndspeed); S_LoadSound_t AudioInputPlugins[10] = { #ifdef AVAIL_OGGVORBIS S_LoadOVSound, #endif S_LoadWavSound, #ifdef DOOMWADS S_LoadDoomSound, S_LoadDoomSpeakerSound, #endif }; qboolean S_RegisterSoundInputPlugin(S_LoadSound_t loadfnc) { int i; for (i = 0; i < sizeof(AudioInputPlugins)/sizeof(AudioInputPlugins[0]); i++) { if (!AudioInputPlugins[i]) { AudioInputPlugins[i] = loadfnc; return true; } } return false; } /* ============== S_LoadSound ============== */ sfxcache_t *S_LoadSound (sfx_t *s) { char stackbuf[65536]; char namebuffer[256]; qbyte *data; sfxcache_t *sc; int i; char *name = s->name; // see if still in memory sc = Cache_Check (&s->cache); if (sc) return sc; s->decoder = NULL; if (name[1] == ':' && name[2] == '\\') { FILE *f; #ifndef _WIN32 //convert from windows to a suitable alternative. char unixname[128]; sprintf(unixname, "/mnt/%c/%s", name[0]-'A'+'a', name+3); name = unixname; while (*name) { if (*name == '\\') *name = '/'; name++; } name = unixname; #endif if ((f = fopen(name, "rb"))) { com_filesize = COM_filelength(f); data = Hunk_TempAlloc (com_filesize); fread(data, 1, com_filesize, f); fclose(f); } else { Con_SafePrintf ("Couldn't load %s\n", namebuffer); return NULL; } } else { //Con_Printf ("S_LoadSound: %x\n", (int)stackbuf); // load it in data = NULL; if (*name == '*') { Q_strcpy(namebuffer, "players/male/"); //q2 Q_strcat(namebuffer, name+1); //q2 } else if (name[0] == '.' && name[1] == '.' && name[2] == '/') Q_strcpy(namebuffer, name+3); else { Q_strcpy(namebuffer, "sound/"); Q_strcat(namebuffer, name); data = COM_LoadStackFile(name, stackbuf, sizeof(stackbuf)); } // Con_Printf ("loading %s\n",namebuffer); if (!data) data = COM_LoadStackFile(namebuffer, stackbuf, sizeof(stackbuf)); } if (!data) { //FIXME: check to see if qued for download. Con_DPrintf ("Couldn't load %s\n", namebuffer); return NULL; } for (i = sizeof(AudioInputPlugins)/sizeof(AudioInputPlugins[0])-1; i >= 0; i--) { if (AudioInputPlugins[i]) { sc = AudioInputPlugins[i](s, data, com_filesize, snd_speed); if (sc) return sc; } } return NULL; } /* =============================================================================== WAV loading =============================================================================== */ qbyte *data_p; qbyte *iff_end; qbyte *last_chunk; qbyte *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; data_p += 4; if (data_p >= iff_end) { // didn't find the chunk data_p = NULL; return; } 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(data_p, name, 4)) return; } } void FindChunk(char *name) { last_chunk = iff_data; FindNextChunk (name); } #if 0 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); } #endif /* ============ GetWavinfo ============ */ wavinfo_t GetWavinfo (char *name, qbyte *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(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.numchannels = 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 (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 /info.numchannels; 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; }