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fteqw/engine/client/snd_mem.c

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/*
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"
#include "errno.h"
int cache_full_cycle;
qbyte *S_Alloc (int size);
#define LINEARUPSCALE(in, inrate, insamps, out, outrate, outlshift, outrshift) \
{ \
scale = inrate / (double)outrate; \
infrac = floor(scale * 65536); \
outsamps = insamps / scale; \
inaccum = 0; \
outnlsamps = floor(1.0 / scale); \
outsamps -= outnlsamps; \
\
while (outsamps) \
{ \
*out = ((0xFFFF - inaccum)*in[0] + inaccum*in[1]) >> (16 - outlshift + outrshift); \
inaccum += infrac; \
in += (inaccum >> 16); \
inaccum &= 0xFFFF; \
out++; \
outsamps--; \
} \
while (outnlsamps) \
{ \
*out = (*in >> outrshift) << outlshift; \
out++; \
outnlsamps--; \
} \
}
#define LINEARUPSCALESTEREO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
{ \
scale = inrate / (double)outrate; \
infrac = floor(scale * 65536); \
outsamps = insamps / scale; \
inaccum = 0; \
outnlsamps = floor(1.0 / scale); \
outsamps -= outnlsamps; \
\
while (outsamps) \
{ \
out[0] = ((0xFFFF - inaccum)*in[0] + inaccum*in[2]) >> (16 - outlshift + outrshift); \
out[1] = ((0xFFFF - inaccum)*in[1] + inaccum*in[3]) >> (16 - outlshift + outrshift); \
inaccum += infrac; \
in += (inaccum >> 16) * 2; \
inaccum &= 0xFFFF; \
out += 2; \
outsamps--; \
} \
while (outnlsamps) \
{ \
out[0] = (in[0] >> outrshift) << outlshift; \
out[1] = (in[1] >> outrshift) << outlshift; \
out += 2; \
outnlsamps--; \
} \
}
#define LINEARUPSCALESTEREOTOMONO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
{ \
scale = inrate / (double)outrate; \
infrac = floor(scale * 65536); \
outsamps = insamps / scale; \
inaccum = 0; \
outnlsamps = floor(1.0 / scale); \
outsamps -= outnlsamps; \
\
while (outsamps) \
{ \
*out = ((((0xFFFF - inaccum)*in[0] + inaccum*in[2]) >> (16 - outlshift + outrshift)) + \
(((0xFFFF - inaccum)*in[1] + inaccum*in[3]) >> (16 - outlshift + outrshift))) >> 1; \
inaccum += infrac; \
in += (inaccum >> 16) * 2; \
inaccum &= 0xFFFF; \
out++; \
outsamps--; \
} \
while (outnlsamps) \
{ \
out[0] = (((in[0] >> outrshift) << outlshift) + ((in[1] >> outrshift) << outlshift)) >> 1; \
out++; \
outnlsamps--; \
} \
}
#define LINEARDOWNSCALE(in, inrate, insamps, out, outrate, outlshift, outrshift) \
{ \
scale = outrate / (double)inrate; \
infrac = floor(scale * 65536); \
inaccum = 0; \
insamps--; \
outsampleft = 0; \
\
while (insamps) \
{ \
inaccum += infrac; \
if (inaccum >> 16) \
{ \
inaccum &= 0xFFFF; \
outsampleft += (infrac - inaccum) * (*in); \
*out = outsampleft >> (16 - outlshift + outrshift); \
out++; \
outsampleft = inaccum * (*in); \
} \
else \
outsampleft += infrac * (*in); \
in++; \
insamps--; \
} \
outsampleft += (0xFFFF - inaccum) * (*in);\
*out = outsampleft >> (16 - outlshift + outrshift); \
}
#define LINEARDOWNSCALESTEREO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
{ \
scale = outrate / (double)inrate; \
infrac = floor(scale * 65536); \
inaccum = 0; \
insamps--; \
outsampleft = 0; \
outsampright = 0; \
\
while (insamps) \
{ \
inaccum += infrac; \
if (inaccum >> 16) \
{ \
inaccum &= 0xFFFF; \
outsampleft += (infrac - inaccum) * in[0]; \
outsampright += (infrac - inaccum) * in[1]; \
out[0] = outsampleft >> (16 - outlshift + outrshift); \
out[1] = outsampright >> (16 - outlshift + outrshift); \
out += 2; \
outsampleft = inaccum * in[0]; \
outsampright = inaccum * in[1]; \
} \
else \
{ \
outsampleft += infrac * in[0]; \
outsampright += infrac * in[1]; \
} \
in += 2; \
insamps--; \
} \
outsampleft += (0xFFFF - inaccum) * in[0];\
outsampright += (0xFFFF - inaccum) * in[1];\
out[0] = outsampleft >> (16 - outlshift + outrshift); \
out[1] = outsampright >> (16 - outlshift + outrshift); \
}
#define LINEARDOWNSCALESTEREOTOMONO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
{ \
scale = outrate / (double)inrate; \
infrac = floor(scale * 65536); \
inaccum = 0; \
insamps--; \
outsampleft = 0; \
\
while (insamps) \
{ \
inaccum += infrac; \
if (inaccum >> 16) \
{ \
inaccum &= 0xFFFF; \
outsampleft += (infrac - inaccum) * ((in[0] + in[1]) >> 1); \
*out = outsampleft >> (16 - outlshift + outrshift); \
out++; \
outsampleft = inaccum * ((in[0] + in[1]) >> 1); \
} \
else \
outsampleft += infrac * ((in[0] + in[1]) >> 1); \
in += 2; \
insamps--; \
} \
outsampleft += (0xFFFF - inaccum) * ((in[0] + in[1]) >> 1);\
*out = outsampleft >> (16 - outlshift + outrshift); \
}
#define STANDARDRESCALE(in, inrate, insamps, out, outrate, outlshift, outrshift) \
{ \
scale = inrate / (double)outrate; \
infrac = floor(scale * 65536); \
outsamps = insamps / scale; \
inaccum = 0; \
\
while (outsamps) \
{ \
*out = (*in >> outrshift) << outlshift; \
inaccum += infrac; \
in += (inaccum >> 16); \
inaccum &= 0xFFFF; \
out++; \
outsamps--; \
} \
}
#define STANDARDRESCALESTEREO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
{ \
scale = inrate / (double)outrate; \
infrac = floor(scale * 65536); \
outsamps = insamps / scale; \
inaccum = 0; \
\
while (outsamps) \
{ \
out[0] = (in[0] >> outrshift) << outlshift; \
out[1] = (in[1] >> outrshift) << outlshift; \
inaccum += infrac; \
in += (inaccum >> 16) * 2; \
inaccum &= 0xFFFF; \
out += 2; \
outsamps--; \
} \
}
#define STANDARDRESCALESTEREOTOMONO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
{ \
scale = inrate / (double)outrate; \
infrac = floor(scale * 65536); \
outsamps = insamps / scale; \
inaccum = 0; \
\
while (outsamps) \
{ \
out[0] = (((in[0] >> outrshift) << outlshift) + ((in[1] >> outrshift) << outlshift)) >> 1; \
inaccum += infrac; \
in += (inaccum >> 16) * 2; \
inaccum &= 0xFFFF; \
out++; \
outsamps--; \
} \
}
#define QUICKCONVERT(in, insamps, out, outlshift, outrshift) \
{ \
while (insamps) \
{ \
*out = (*in >> outrshift) << outlshift; \
out++; \
in++; \
insamps--; \
} \
}
#define QUICKCONVERTSTEREOTOMONO(in, insamps, out, outlshift, outrshift) \
{ \
while (insamps) \
{ \
*out = (((in[0] >> outrshift) << outlshift) + ((in[1] >> outrshift) << outlshift)) >> 1; \
out++; \
in += 2; \
insamps--; \
} \
}
// SND_ResampleStream: takes a sound stream and converts with given parameters. Limited to
// 8-16-bit signed conversions and mono-to-mono/stereo-to-stereo conversions.
// Not an in-place algorithm.
void SND_ResampleStream (void *in, int inrate, int inwidth, int inchannels, int insamps, void *out, int outrate, int outwidth, int outchannels, int resampstyle)
{
double scale;
signed char *in8 = (signed char *)in;
short *in16 = (short *)in;
signed char *out8 = (signed char *)out;
short *out16 = (short *)out;
int outsamps, outnlsamps, outsampleft, outsampright;
int infrac, inaccum;
if (insamps <= 0)
return;
if (inchannels == outchannels && inwidth == outwidth && inrate == outrate)
{
memcpy(out, in, inwidth*insamps*inchannels);
return;
}
if (inchannels == 1 && outchannels == 1)
{
if (inwidth == 1)
{
if (outwidth == 1)
{
if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALE(in8, inrate, insamps, out8, outrate, 0, 0)
else
STANDARDRESCALE(in8, inrate, insamps, out8, outrate, 0, 0)
}
else // downsample
{
if (resampstyle > 1)
LINEARDOWNSCALE(in8, inrate, insamps, out8, outrate, 0, 0)
else
STANDARDRESCALE(in8, inrate, insamps, out8, outrate, 0, 0)
}
return;
}
else
{
if (inrate == outrate) // quick convert
QUICKCONVERT(in8, insamps, out16, 8, 0)
else if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALE(in8, inrate, insamps, out16, outrate, 8, 0)
else
STANDARDRESCALE(in8, inrate, insamps, out16, outrate, 8, 0)
}
else // downsample
{
if (resampstyle > 1)
LINEARDOWNSCALE(in8, inrate, insamps, out16, outrate, 8, 0)
else
STANDARDRESCALE(in8, inrate, insamps, out16, outrate, 8, 0)
}
return;
}
}
else // 16-bit
{
if (outwidth == 2)
{
if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALE(in16, inrate, insamps, out16, outrate, 0, 0)
else
STANDARDRESCALE(in16, inrate, insamps, out16, outrate, 0, 0)
}
else // downsample
{
if (resampstyle > 1)
LINEARDOWNSCALE(in16, inrate, insamps, out16, outrate, 0, 0)
else
STANDARDRESCALE(in16, inrate, insamps, out16, outrate, 0, 0)
}
return;
}
else
{
if (inrate == outrate) // quick convert
QUICKCONVERT(in16, insamps, out8, 0, 8)
else if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALE(in16, inrate, insamps, out8, outrate, 0, 8)
else
STANDARDRESCALE(in16, inrate, insamps, out8, outrate, 0, 8)
}
else // downsample
{
if (resampstyle > 1)
LINEARDOWNSCALE(in16, inrate, insamps, out8, outrate, 0, 8)
else
STANDARDRESCALE(in16, inrate, insamps, out8, outrate, 0, 8)
}
return;
}
}
}
else if (outchannels == 2 && inchannels == 2)
{
if (inwidth == 1)
{
if (outwidth == 1)
{
if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALESTEREO(in8, inrate, insamps, out8, outrate, 0, 0)
else
STANDARDRESCALESTEREO(in8, inrate, insamps, out8, outrate, 0, 0)
}
else // downsample
{
if (resampstyle > 1)
LINEARDOWNSCALESTEREO(in8, inrate, insamps, out8, outrate, 0, 0)
else
STANDARDRESCALESTEREO(in8, inrate, insamps, out8, outrate, 0, 0)
}
}
else
{
if (inrate == outrate) // quick convert
{
insamps *= 2;
QUICKCONVERT(in8, insamps, out16, 8, 0)
}
else if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALESTEREO(in8, inrate, insamps, out16, outrate, 8, 0)
else
STANDARDRESCALESTEREO(in8, inrate, insamps, out16, outrate, 8, 0)
}
else // downsample
{
if (resampstyle > 1)
LINEARDOWNSCALESTEREO(in8, inrate, insamps, out16, outrate, 8, 0)
else
STANDARDRESCALESTEREO(in8, inrate, insamps, out16, outrate, 8, 0)
}
}
}
else // 16-bit
{
if (outwidth == 2)
{
if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
else
STANDARDRESCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
}
else // downsample
{
if (resampstyle > 1)
LINEARDOWNSCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
else
STANDARDRESCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
}
}
else
{
if (inrate == outrate) // quick convert
{
insamps *= 2;
QUICKCONVERT(in16, insamps, out8, 0, 8)
}
else if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
else
STANDARDRESCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
}
else // downsample
{
if (resampstyle > 1)
LINEARDOWNSCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
else
STANDARDRESCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
}
}
}
}
#if 0
else if (outchannels == 1 && inchannels == 2)
{
if (inwidth == 1)
{
if (outwidth == 1)
{
if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALESTEREOTOMONO(in8, inrate, insamps, out8, outrate, 0, 0)
else
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out8, outrate, 0, 0)
}
else // downsample
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out8, outrate, 0, 0)
}
else
{
if (inrate == outrate) // quick convert
QUICKCONVERTSTEREOTOMONO(in8, insamps, out16, 8, 0)
else if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALESTEREOTOMONO(in8, inrate, insamps, out16, outrate, 8, 0)
else
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out16, outrate, 8, 0)
}
else // downsample
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out16, outrate, 8, 0)
}
}
else // 16-bit
{
if (outwidth == 2)
{
if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALESTEREOTOMONO(in16, inrate, insamps, out16, outrate, 0, 0)
else
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out16, outrate, 0, 0)
}
else // downsample
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out16, outrate, 0, 0)
}
else
{
if (inrate == outrate) // quick convert
QUICKCONVERTSTEREOTOMONO(in16, insamps, out8, 0, 8)
else if (inrate < outrate) // upsample
{
if (resampstyle)
LINEARUPSCALESTEREOTOMONO(in16, inrate, insamps, out8, outrate, 0, 8)
else
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out8, outrate, 0, 8)
}
else // downsample
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out8, outrate, 0, 8)
}
}
}
#endif
}
/*
================
ResampleSfx
================
*/
void ResampleSfx (sfx_t *sfx, int inrate, int inchannels, int inwidth, int insamps, int inloopstart, qbyte *data)
{
extern cvar_t snd_linearresample;
double scale;
sfxcache_t *sc;
int outsamps;
int len;
int outwidth;
scale = snd_speed / (double)inrate;
outsamps = insamps * scale;
if (loadas8bit.ival < 0)
outwidth = 2;
else if (loadas8bit.ival)
outwidth = 1;
else
outwidth = inwidth;
len = outsamps * outwidth * inchannels;
sc = Cache_Alloc (&sfx->cache, len + sizeof(sfxcache_t), sfx->name);
if (!sc)
{
return;
}
sc->numchannels = inchannels;
sc->width = outwidth;
sc->speed = snd_speed;
sc->length = outsamps;
if (inloopstart == -1)
sc->loopstart = inloopstart;
else
sc->loopstart = inloopstart * scale;
SND_ResampleStream (data,
inrate,
inwidth,
inchannels,
insamps,
sc->data,
sc->speed,
sc->width,
sc->numchannels,
snd_linearresample.ival);
}
//=============================================================================
#ifdef DOOMWADS
#define DSPK_RATE 140
#define DSPK_BASE 170.0
#define DSPK_EXP 0.0433
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, inrate, inaccum;
qbyte *outdata;
qbyte towrite;
double timeraccum, timerfreq;
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;
if (*data)
timerfreq = DSPK_BASE * pow((double)2.0, DSPK_EXP * (*data));
else
timerfreq = 0;
while (len > 0)
{
timeraccum += timerfreq;
if (timeraccum > (float)snd_speed)
{
towrite ^= 0xFF; // swap speaker component
timeraccum -= (float)snd_speed;
}
inaccum--;
if (!inaccum)
{
data++;
if (*data)
timerfreq = DSPK_BASE * pow((double)2.0, DSPK_EXP * (*data));
inaccum = inrate;
}
*outdata = towrite;
outdata++;
len--;
}
return sc;
}
sfxcache_t *S_LoadDoomSound (sfx_t *s, qbyte *data, int datalen, int sndspeed)
{
// 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;
COM_CharBias(data, sc->length);
ResampleSfx (s, rate, 1, 1, samples, -1, data);
return Cache_Check(&s->cache);
}
#endif
sfxcache_t *S_LoadWavSound (sfx_t *s, qbyte *data, int datalen, int sndspeed)
{
wavinfo_t info;
if (datalen < 4 || strncmp(data, "RIFF", 4))
return NULL;
info = GetWavinfo (s->name, data, datalen);
if (info.numchannels < 1 || info.numchannels > 2)
{
s->failedload = true;
Con_Printf ("%s has an unsupported quantity of channels.\n",s->name);
return NULL;
}
if (info.width == 1)
COM_CharBias(data + info.dataofs, info.samples*info.numchannels);
else if (info.width == 2)
COM_SwapLittleShortBlock((short *)(data + info.dataofs), info.samples*info.numchannels);
ResampleSfx (s, info.rate, info.numchannels, info.width, info.samples, info.loopstart, data + info.dataofs);
return Cache_Check(&s->cache);
}
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;
size_t result;
char *name = s->name;
if (s->failedload)
return NULL; //it failed to load once before, don't bother trying again.
// 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);
result = fread(data, 1, com_filesize, f); //do something with result
if (result != com_filesize)
Con_SafePrintf("S_LoadSound() fread: Filename: %s, expected %i, result was %u (%s)\n",name,com_filesize,(unsigned int)result,strerror(errno));
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 == '*') //q2 sexed sounds
{
//clq2_parsestartsound detects this also
//here we just precache the male sound name, which provides us with our default
Q_strcpy(namebuffer, "players/male/"); //q2
Q_strcat(namebuffer, name+1); //q2
}
else if (name[0] == '.' && name[1] == '.' && name[2] == '/')
{
//not relative to sound/
Q_strcpy(namebuffer, name+3);
}
else
{
//q1 behaviour, relative to sound/
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)
{
char altname[sizeof(namebuffer)];
COM_StripExtension(namebuffer, altname, sizeof(altname));
COM_DefaultExtension(altname, ".ogg", sizeof(altname));
data = COM_LoadStackFile(altname, stackbuf, sizeof(stackbuf));
if (data)
Con_DPrintf("found a mangled name\n");
}
}
if (!data)
{
//FIXME: check to see if queued for download.
Con_DPrintf ("Couldn't load %s\n", namebuffer);
s->failedload = true;
return NULL;
}
s->failedload = false;
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;
}
}
}
if (!s->failedload)
Con_Printf ("Format not recognised: %s\n", namebuffer);
s->failedload = true;
return NULL;
}
/*
===============================================================================
WAV loading
===============================================================================
*/
char *wavname;
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;
}
unsigned int FindNextChunk(char *name)
{
unsigned int dataleft;
while (1)
{
dataleft = iff_end - last_chunk;
if (dataleft < 8)
{ // didn't find the chunk
data_p = NULL;
return 0;
}
data_p=last_chunk;
data_p += 4;
dataleft-= 8;
iff_chunk_len = GetLittleLong();
if (iff_chunk_len < 0)
{
data_p = NULL;
return 0;
}
if (iff_chunk_len > dataleft)
{
Con_DPrintf ("\"%s\" seems truncated by %i bytes\n", wavname, iff_chunk_len-dataleft);
#if 1
iff_chunk_len = dataleft;
#else
data_p = NULL;
return 0;
#endif
}
dataleft-= iff_chunk_len;
// 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;
if ((iff_chunk_len&1) && dataleft)
last_chunk++;
if (!Q_strncmp(data_p, name, 4))
return iff_chunk_len;
}
}
unsigned int FindChunk(char *name)
{
last_chunk = iff_data;
return 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;
int chunklen;
memset (&info, 0, sizeof(info));
if (!wav)
return info;
iff_data = wav;
iff_end = wav + wavlength;
wavname = name;
// find "RIFF" chunk
chunklen = FindChunk("RIFF");
if (chunklen < 4 || Q_strncmp(data_p+8, "WAVE", 4))
{
Con_Printf("Missing RIFF/WAVE chunks in %s\n", name);
return info;
}
// get "fmt " chunk
iff_data = data_p + 12;
// DumpChunks ();
chunklen = FindChunk("fmt ");
if (chunklen < 24-8)
{
Con_Printf("Missing/truncated 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
chunklen = FindChunk("cue ");
if (chunklen >= 36-8)
{
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
chunklen = FindNextChunk ("LIST");
if (chunklen >= 32-8)
{
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
chunklen = FindChunk("data");
if (!chunklen)
{
Con_Printf("Missing data chunk in %s\n", name);
return info;
}
data_p += 8;
samples = chunklen / info.width /info.numchannels;
if (info.samples)
{
if (samples < info.samples)
{
info.samples = samples;
Con_Printf ("Sound %s has a bad loop length\n", name);
}
}
else
info.samples = samples;
if (info.loopstart > info.samples)
{
Con_Printf ("Sound %s has a bad loop start\n", name);
info.loopstart = info.samples;
}
info.dataofs = data_p - wav;
return info;
}
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