mirror of
https://git.code.sf.net/p/quake/newtree
synced 2024-11-22 20:21:38 +00:00
new sound resampling from nuq. If 16 bit internal samples is too much for your
system, set loadas8bit.
This commit is contained in:
parent
21787a552f
commit
23f5c58362
1 changed files with 97 additions and 43 deletions
140
source/snd_mem.c
140
source/snd_mem.c
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@ -1,7 +1,7 @@
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/*
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snd_mem.c
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sound caching
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@description@
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Copyright (C) 1996-1997 Id Software, Inc.
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@ -27,16 +27,17 @@
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*/
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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# include "config.h"
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#endif
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#include <string.h>
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#include "sys.h"
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#include "sound.h"
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#include "qendian.h"
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#include "quakefs.h"
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#include "console.h"
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#include <string.h>
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int cache_full_cycle;
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byte *S_Alloc (int size);
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@ -54,53 +55,104 @@ void ResampleSfx (sfx_t *sfx, int inrate, int inwidth, byte *data)
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int i;
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int sample, samplefrac, fracstep;
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sfxcache_t *sc;
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short *is, *os;
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unsigned char *ib, *ob;
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sc = Cache_Check (&sfx->cache);
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if (!sc)
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return;
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is = (short*)data;
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os = (short*)sc->data;
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ib = data;
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ob = sc->data;
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stepscale = (float)inrate / shm->speed; // this is usually 0.5, 1, or 2
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outcount = sc->length / stepscale;
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sc->speed = shm->speed;
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if (loadas8bit->int_val)
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sc->width = 1;
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else
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sc->width = 2;
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sc->stereo = 0;
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// resample / decimate to the current source rate
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if (stepscale == 1) {
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if (inwidth == 1 && sc->width == 1) {
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for (i=0 ; i<outcount ; i++) {
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*ob++ = *ib++ - 128;
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}
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} else if (inwidth == 1 && sc->width == 2) {
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for (i=0 ; i<outcount ; i++) {
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*os++ = (*ib++ - 128) << 8;
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}
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} else if (inwidth == 2 && sc->width == 1) {
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for (i=0 ; i<outcount ; i++) {
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*ob++ = LittleShort (*is++) >> 8;
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}
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} else if (inwidth == 2 && sc->width == 2) {
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for (i=0 ; i<outcount ; i++) {
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*os++ = LittleShort (*is++);
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}
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}
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} else {
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// general case
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if (stepscale < 1) {
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int points = 1/stepscale;
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int j;
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for (i = 0; i < sc->length; i++) {
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int s1, s2;
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if (inwidth == 2) {
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s2 = s1 = LittleShort (is[0]);
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if (i < sc->length - 1)
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s2 = LittleShort (is[1]);
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is++;
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} else {
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s2 = s1 = (ib[0] - 128) << 8;
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if (i < sc->length - 1)
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s2 = (ib[1] - 128) << 8;
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ib++;
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}
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for (j = 0; j < points; j++) {
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sample = s1 + (s2 - s1) * ((float)j) / points;
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if (sc->width == 2) {
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os[j] = sample;
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} else {
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ob[j] = sample >> 8;
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}
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}
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if (sc->width == 2) {
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os += points;
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} else {
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ob += points;
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}
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}
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} else {
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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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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 = shm->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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sc->stereo = 0;
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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
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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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@ -150,7 +202,11 @@ sfxcache_t *S_LoadSound (sfx_t *s)
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stepscale = (float)info.rate / shm->speed;
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len = info.samples / stepscale;
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len = len * info.width * info.channels;
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if (loadas8bit->int_val) {
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len = len * info.channels;
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} else {
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len = len * 2 * info.channels;
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}
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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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}
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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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@ -256,7 +311,6 @@ void DumpChunks(void)
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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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