dfd8e1aaed
Added support for dp6/dp7 protocols (ents are still broken). md3 tags should work properly (still suffer from origin-of-parent interpolation bugs) git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@1089 fc73d0e0-1445-4013-8a0c-d673dee63da5
428 lines
8.9 KiB
C
428 lines
8.9 KiB
C
/*
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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 the Free Software
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Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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// snd_mem.c: sound caching
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#include "quakedef.h"
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#include "winquake.h"
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int cache_full_cycle;
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qbyte *S_Alloc (int size);
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/*
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================
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ResampleSfx
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================
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*/
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void ResampleSfx (sfx_t *sfx, int inrate, int inwidth, qbyte *data)
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{
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int outcount;
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int srcsample;
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float stepscale;
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int i;
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int sample, fracstep;
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unsigned int samplefrac;
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sfxcache_t *sc;
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sc = Cache_Check (&sfx->cache);
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if (!sc)
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return;
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stepscale = (float)inrate / snd_speed; // this is usually 0.5, 1, or 2
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outcount = sc->length / stepscale;
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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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sc->speed = snd_speed;
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if (loadas8bit.value)
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sc->width = 1;
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else
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sc->width = inwidth;
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if (sc->stereo)
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{
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if (stepscale == 1 && inwidth == 1 && sc->width == 1)
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{
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outcount*=2;
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// fast special case
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for (i=0 ; i<outcount ; i++)
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((signed char *)sc->data)[i]
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= (int)( (unsigned char)(data[i]) - 128);
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}
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else if (stepscale == 1 && inwidth == 2 && sc->width == 2)
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{
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outcount*=2;
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// fast special case
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for (i=0 ; i<outcount ; i++)
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((short *)sc->data)[i] = LittleShort ( ((short *)data)[i] );
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}
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else
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{
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// general case
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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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{
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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<<1)] );
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else
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sample = (int)( (unsigned char)(data[(srcsample<<1)]) - 128) << 8;
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if (sc->width == 2)
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((short *)sc->data)[i<<1] = sample;
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else
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((signed char *)sc->data)[i<<1] = sample >> 8;
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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<<1)+1] );
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else
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sample = (int)( (unsigned char)(data[(srcsample<<1)+1]) - 128) << 8;
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if (sc->width == 2)
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((short *)sc->data)[(i<<1)+1] = sample;
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else
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((signed char *)sc->data)[(i<<1)+1] = sample >> 8;
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}
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}
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return;
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}
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// resample / decimate to the current source rate
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if (stepscale == 1 && inwidth == 1 && sc->width == 1)
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{
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// fast special case
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for (i=0 ; i<outcount ; i++)
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((signed char *)sc->data)[i]
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= (int)( (unsigned char)(data[i]) - 128);
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}
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else if (stepscale == 1 && inwidth == 2 && sc->width == 2)
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{
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// fast special case
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for (i=0 ; i<outcount ; i++)
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((short *)sc->data)[i] = LittleShort ( ((short *)data)[i] );
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}
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else
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{
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// general case
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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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{
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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 = (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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//=============================================================================
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/*
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==============
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S_LoadSound
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==============
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*/
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#ifdef AVAIL_MP3
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sfxcache_t *S_LoadMP3Sound (sfx_t *s);
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#endif
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sfxcache_t *S_LoadOVSound (sfx_t *s);
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sfxcache_t *S_LoadSound (sfx_t *s)
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{
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char namebuffer[256];
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qbyte *data;
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wavinfo_t info;
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int len;
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sfxcache_t *sc;
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qbyte stackbuf[1*1024]; // avoid dirtying the cache heap
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// see if still in memory
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sc = Cache_Check (&s->cache);
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if (sc)
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return sc;
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#ifdef AVAIL_OGGVORBIS
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//ogg vorbis support. The only bit actual code outside snd_ov.c (excluding def for the function call)
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sc = S_LoadOVSound(s); // try and load a replacement ov instead.
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if (sc)
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return sc;
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#endif
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#ifdef AVAIL_MP3
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//mp3 support. The only bit actual code outside snd_mp3.c (excluding def for the function call)
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sc = S_LoadMP3Sound(s); // try and load a replacement mp3 instead.
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if (sc)
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return sc;
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#endif
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//Con_Printf ("S_LoadSound: %x\n", (int)stackbuf);
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// load it in
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if (*s->name == '*')
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{
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Q_strcpy(namebuffer, "players/male/"); //q2
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Q_strcat(namebuffer, s->name+1); //q2
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}
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else if (s->name[0] == '.' && s->name[1] == '.' && s->name[2] == '/')
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Q_strcpy(namebuffer, s->name+3);
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else
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{
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Q_strcpy(namebuffer, "sound/");
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Q_strcat(namebuffer, s->name);
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}
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// Con_Printf ("loading %s\n",namebuffer);
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data = COM_LoadStackFile(namebuffer, stackbuf, sizeof(stackbuf));
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if (!data)
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{
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//FIXME: check to see if qued for download.
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Con_Printf ("Couldn't load %s\n", namebuffer);
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return NULL;
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}
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info = GetWavinfo (s->name, data, com_filesize);
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if (info.numchannels < 1 || info.numchannels > 2)
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{
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Con_Printf ("%s has an unsupported quantity of channels.\n",s->name);
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return NULL;
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}
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len = (int) ((double) info.samples * (double) snd_speed / (double) info.rate);
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len = len * info.width * info.numchannels;
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sc = Cache_Alloc ( &s->cache, len + sizeof(sfxcache_t), s->name);
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if (!sc)
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return NULL;
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sc->length = info.samples;
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sc->loopstart = info.loopstart;
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sc->speed = info.rate;
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sc->width = info.width;
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sc->stereo = info.numchannels-1;
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ResampleSfx (s, sc->speed, sc->width, data + info.dataofs);
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return sc;
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}
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/*
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===============================================================================
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WAV loading
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===============================================================================
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*/
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qbyte *data_p;
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qbyte *iff_end;
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qbyte *last_chunk;
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qbyte *iff_data;
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int iff_chunk_len;
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short 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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int 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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void FindNextChunk(char *name)
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{
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while (1)
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{
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data_p=last_chunk;
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data_p += 4;
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if (data_p >= iff_end)
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{ // 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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iff_chunk_len = GetLittleLong();
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if (iff_chunk_len < 0)
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{
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data_p = NULL;
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return;
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}
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// if (iff_chunk_len > 1024*1024)
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// Sys_Error ("FindNextChunk: %i length is past the 1 meg sanity limit", iff_chunk_len);
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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 (!Q_strncmp(data_p, name, 4))
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return;
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}
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}
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void FindChunk(char *name)
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{
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last_chunk = iff_data;
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FindNextChunk (name);
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}
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#if 0
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void 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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{
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memcpy (str, data_p, 4);
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data_p += 4;
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iff_chunk_len = GetLittleLong();
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Con_Printf ("0x%x : %s (%d)\n", (int)(data_p - 4), str, 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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#endif
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/*
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============
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GetWavinfo
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============
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*/
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wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength)
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{
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wavinfo_t info;
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int i;
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int format;
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int samples;
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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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FindChunk("RIFF");
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if (!(data_p && !Q_strncmp(data_p+8, "WAVE", 4)))
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{
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Con_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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// DumpChunks ();
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FindChunk("fmt ");
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if (!data_p)
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{
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Con_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 = GetLittleShort();
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if (format != 1)
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{
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Con_Printf("Microsoft PCM format only\n");
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return info;
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}
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info.numchannels = GetLittleShort();
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info.rate = GetLittleLong();
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data_p += 4+2;
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info.width = GetLittleShort() / 8;
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// get cue chunk
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FindChunk("cue ");
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if (data_p)
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{
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data_p += 32;
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info.loopstart = GetLittleLong();
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// Con_Printf("loopstart=%d\n", sfx->loopstart);
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// if the next chunk is a LIST chunk, look for a cue length marker
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FindNextChunk ("LIST");
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if (data_p)
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{
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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 = GetLittleLong (); // samples in loop
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info.samples = info.loopstart + i;
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// Con_Printf("looped length: %i\n", i);
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}
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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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FindChunk("data");
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if (!data_p)
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{
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Con_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 = GetLittleLong () / info.width /info.numchannels;
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if (info.samples)
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{
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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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}
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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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