d1d0d86fea
Fix sound source issues in Q3. Fix q2 air acceleration/prediction omission. Don't change console completion while typing (while that option is still possible). Shift+tab now cycles completion backwards (now ctrl+shift for cycle subconsoles). Allow a few things to ignore sv_pure - including csprogs files (which is useful for all the mods that come with the csprogs.dat distributed separately). clamp pitch values to the range documented by openal, to hopefully avoid error spam. add some colour coding to the text editor when shader files are being edited/viewed. Changed how overbrights are clamped on q3bsp. Added portalfboscale for explicit texture scales on portal/refract/reflect fbos. qc decompiler can now at least attempt to decompile qtest's qc. fteqccgui can now be pointed at a .pak file, and decompile the progs.dat inside. git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@5269 fc73d0e0-1445-4013-8a0c-d673dee63da5
1191 lines
27 KiB
C
1191 lines
27 KiB
C
/*
|
|
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 "fs.h"
|
|
|
|
typedef struct
|
|
{
|
|
int format;
|
|
int rate;
|
|
int width;
|
|
int numchannels;
|
|
int loopstart;
|
|
int samples;
|
|
int dataofs; // chunk starts this many bytes from file start
|
|
} wavinfo_t;
|
|
|
|
static wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength);
|
|
|
|
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
|
|
================
|
|
*/
|
|
static qboolean 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;
|
|
|
|
sfx->decoder.buf = sc = BZ_Malloc(len + sizeof(sfxcache_t));
|
|
if (!sc)
|
|
{
|
|
return false;
|
|
}
|
|
|
|
sc->numchannels = inchannels;
|
|
sc->width = outwidth;
|
|
sc->speed = snd_speed;
|
|
sc->length = outsamps;
|
|
sc->soundoffset = 0;
|
|
sc->data = (qbyte*)(sc+1);
|
|
if (inloopstart == -1)
|
|
sfx->loopstart = inloopstart;
|
|
else
|
|
sfx->loopstart = inloopstart * scale;
|
|
|
|
SND_ResampleStream (data,
|
|
inrate,
|
|
inwidth,
|
|
inchannels,
|
|
insamps,
|
|
sc->data,
|
|
sc->speed,
|
|
sc->width,
|
|
sc->numchannels,
|
|
snd_linearresample.ival);
|
|
|
|
return true;
|
|
}
|
|
|
|
//=============================================================================
|
|
#ifdef PACKAGE_DOOMWAD
|
|
#define DSPK_RATE 140
|
|
#define DSPK_BASE 170.0
|
|
#define DSPK_EXP 0.0433
|
|
|
|
/*
|
|
qboolean QDECL S_LoadDoomSpeakerSound (sfx_t *s, qbyte *data, size_t 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;
|
|
s->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;
|
|
}
|
|
*/
|
|
static qboolean QDECL S_LoadDoomSound (sfx_t *s, qbyte *data, size_t datalen, int sndspeed)
|
|
{
|
|
// format data from Unofficial Doom Specs v1.6
|
|
unsigned short *dataus;
|
|
int samples, rate;
|
|
|
|
if (datalen < 8)
|
|
return false;
|
|
|
|
dataus = (unsigned short*)data;
|
|
|
|
if (LittleShort(dataus[0]) != 3)
|
|
return false;
|
|
|
|
rate = LittleShort(dataus[1]);
|
|
samples = LittleShort(dataus[2]);
|
|
|
|
data += 8;
|
|
datalen -= 8;
|
|
|
|
if (datalen != samples)
|
|
return false;
|
|
|
|
COM_CharBias(data, datalen);
|
|
|
|
return ResampleSfx (s, rate, 1, 1, samples, -1, data);
|
|
}
|
|
#endif
|
|
|
|
void S_ShortedLittleFloats(void *p, size_t samples)
|
|
{
|
|
short *out = p;
|
|
float *in = p;
|
|
int t;
|
|
while(samples --> 0)
|
|
{
|
|
t = LittleFloat(*in++) * 32767;
|
|
t = bound(-32768, t, 32767);
|
|
*out++ = t;
|
|
}
|
|
}
|
|
|
|
static qboolean QDECL S_LoadWavSound (sfx_t *s, qbyte *data, size_t datalen, int sndspeed)
|
|
{
|
|
wavinfo_t info;
|
|
|
|
if (datalen < 4 || strncmp(data, "RIFF", 4))
|
|
return false;
|
|
|
|
info = GetWavinfo (s->name, data, datalen);
|
|
if (info.numchannels < 1 || info.numchannels > 2)
|
|
{
|
|
s->loadstate = SLS_FAILED;
|
|
Con_Printf ("%s has an unsupported quantity of channels.\n",s->name);
|
|
return false;
|
|
}
|
|
|
|
if (info.format == 1 && info.width == 1)
|
|
COM_CharBias(data + info.dataofs, info.samples*info.numchannels);
|
|
else if (info.format == 1 && info.width == 2)
|
|
COM_SwapLittleShortBlock((short *)(data + info.dataofs), info.samples*info.numchannels);
|
|
else if (info.format == 3 && info.width == 4)
|
|
{
|
|
S_ShortedLittleFloats(data + info.dataofs, info.samples*info.numchannels);
|
|
info.width = 2;
|
|
}
|
|
else
|
|
{
|
|
s->loadstate = SLS_FAILED;
|
|
switch(info.format)
|
|
{
|
|
case 1:
|
|
case 3: Con_Printf ("%s has an unsupported width (%i bits).\n", s->name, info.width*8); break;
|
|
case 6: Con_Printf ("%s uses unsupported a-law format.\n", s->name); break;
|
|
case 7: Con_Printf ("%s uses unsupported mu-law format.\n", s->name); break;
|
|
default: Con_Printf ("%s has an unsupported format.\n", s->name); break;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
return ResampleSfx (s, info.rate, info.numchannels, info.width, info.samples, info.loopstart, data + info.dataofs);
|
|
}
|
|
|
|
qboolean QDECL S_LoadOVSound (sfx_t *s, qbyte *data, size_t datalen, int sndspeed);
|
|
|
|
#ifdef FTE_TARGET_WEB
|
|
//web browsers contain their own decoding libraries that our openal stuff can use.
|
|
static qboolean QDECL S_LoadBrowserFile (sfx_t *s, qbyte *data, size_t datalen, int sndspeed)
|
|
{
|
|
sfxcache_t *sc;
|
|
s->decoder.buf = sc = BZ_Malloc(sizeof(sfxcache_t) + datalen);
|
|
s->loopstart = -1;
|
|
sc->data = (qbyte*)(sc+1);
|
|
sc->length = datalen;
|
|
sc->width = 0; //ie: not pcm
|
|
sc->speed = sndspeed;
|
|
sc->numchannels = 2;
|
|
sc->soundoffset = 0;
|
|
memcpy(sc->data, data, datalen);
|
|
|
|
return true;
|
|
}
|
|
#endif
|
|
|
|
//highest priority is last.
|
|
static S_LoadSound_t AudioInputPlugins[10] =
|
|
{
|
|
#ifdef FTE_TARGET_WEB
|
|
S_LoadBrowserFile,
|
|
#endif
|
|
#ifdef AVAIL_OGGVORBIS
|
|
S_LoadOVSound,
|
|
#endif
|
|
S_LoadWavSound,
|
|
#ifdef PACKAGE_DOOMWAD
|
|
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;
|
|
}
|
|
|
|
static void S_LoadedOrFailed (void *ctx, void *ctxdata, size_t a, size_t b)
|
|
{
|
|
sfx_t *s = ctx;
|
|
s->loadstate = a;
|
|
}
|
|
/*
|
|
==============
|
|
S_LoadSound
|
|
==============
|
|
*/
|
|
|
|
static void S_LoadSoundWorker (void *ctx, void *ctxdata, size_t a, size_t b)
|
|
{
|
|
sfx_t *s = ctx;
|
|
char namebuffer[256];
|
|
qbyte *data;
|
|
int i;
|
|
size_t result;
|
|
char *name = s->name;
|
|
size_t filesize;
|
|
|
|
s->loopstart = -1;
|
|
|
|
if (s->syspath)
|
|
{
|
|
vfsfile_t *f;
|
|
|
|
if ((f = VFSOS_Open(name, "rb")))
|
|
{
|
|
filesize = VFS_GETLEN(f);
|
|
data = BZ_Malloc (filesize);
|
|
result = VFS_READ(f, data, filesize);
|
|
|
|
if (result != filesize)
|
|
Con_SafePrintf("S_LoadSound() fread: Filename: %s, expected %"PRIuSIZE", result was %"PRIuSIZE"\n", name, filesize, result);
|
|
|
|
VFS_CLOSE(f);
|
|
}
|
|
else
|
|
{
|
|
Con_SafePrintf ("Couldn't load %s\n", namebuffer);
|
|
COM_AddWork(WG_MAIN, S_LoadedOrFailed, s, NULL, SLS_FAILED, 0);
|
|
return;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
//Con_Printf ("S_LoadSound: %x\n", (int)stackbuf);
|
|
// load it in
|
|
const char *prefixes[] = {"sound/", ""};
|
|
const char *extensions[] = {
|
|
".wav",
|
|
#ifdef AVAIL_OGGOPUS
|
|
".opus",
|
|
#endif
|
|
#ifdef AVAIL_OGGVORBIS
|
|
".ogg",
|
|
#endif
|
|
};
|
|
char altname[sizeof(namebuffer)];
|
|
char orig[16];
|
|
size_t pre, ex;
|
|
|
|
data = NULL;
|
|
filesize = 0;
|
|
if (*name == '*') //q2 sexed sounds
|
|
{
|
|
//clq2_parsestartsound detects this also, and should not try playing these sounds.
|
|
s->loadstate = SLS_FAILED;
|
|
return;
|
|
}
|
|
|
|
for (pre = 0; !data && pre < countof(prefixes); pre++)
|
|
{
|
|
if (name[0] == '.' && name[1] == '.' && name[2] == '/')
|
|
{ //someone's being specific. disable prefixes entirely.
|
|
if (pre)
|
|
break;
|
|
//not relative to sound/
|
|
Q_snprintfz(namebuffer, sizeof(namebuffer), "%s", name+3);
|
|
}
|
|
else
|
|
Q_snprintfz(namebuffer, sizeof(namebuffer), "%s%s", prefixes[pre], name);
|
|
|
|
data = FS_LoadMallocFile(namebuffer, &filesize);
|
|
if (data)
|
|
break;
|
|
COM_FileExtension(namebuffer, orig, sizeof(orig));
|
|
COM_StripExtension(namebuffer, altname, sizeof(altname));
|
|
for (ex = 0; ex < countof(extensions); ex++)
|
|
{
|
|
if (!strcmp(orig, extensions[ex]+1))
|
|
continue;
|
|
Q_snprintfz(namebuffer, sizeof(namebuffer), "%s%s", altname, extensions[ex]);
|
|
data = FS_LoadMallocFile(namebuffer, &filesize);
|
|
if (data)
|
|
{
|
|
Con_DPrintf("found a mangled name: %s\n", namebuffer);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (!data)
|
|
{
|
|
//FIXME: check to see if queued for download.
|
|
if (name[0] == '.' && name[1] == '.' && name[2] == '/')
|
|
Con_DPrintf ("Couldn't load %s\n", name+3);
|
|
else
|
|
Con_DPrintf ("Couldn't load sound/%s\n", name);
|
|
COM_AddWork(WG_MAIN, S_LoadedOrFailed, s, NULL, SLS_FAILED, 0);
|
|
return;
|
|
}
|
|
|
|
for (i = sizeof(AudioInputPlugins)/sizeof(AudioInputPlugins[0])-1; i >= 0; i--)
|
|
{
|
|
if (AudioInputPlugins[i])
|
|
{
|
|
if (AudioInputPlugins[i](s, data, filesize, snd_speed))
|
|
{
|
|
//wake up the main thread in case it decided to wait for us.
|
|
COM_AddWork(WG_MAIN, S_LoadedOrFailed, s, NULL, SLS_LOADED, 0);
|
|
BZ_Free(data);
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (s->loadstate != SLS_FAILED)
|
|
Con_Printf ("Format not recognised: %s\n", namebuffer);
|
|
|
|
COM_AddWork(WG_MAIN, S_LoadedOrFailed, s, NULL, SLS_FAILED, 0);
|
|
BZ_Free(data);
|
|
return;
|
|
}
|
|
|
|
qboolean S_LoadSound (sfx_t *s)
|
|
{
|
|
if (s->loadstate == SLS_NOTLOADED && sndcardinfo)
|
|
{
|
|
s->loadstate = SLS_LOADING;
|
|
COM_AddWork(WG_LOADER, S_LoadSoundWorker, s, NULL, 0, 0);
|
|
}
|
|
if (s->loadstate == SLS_FAILED)
|
|
return false; //it failed to load once before, don't bother trying again.
|
|
|
|
return true; //loaded okay, or still loading
|
|
}
|
|
|
|
/*
|
|
===============================================================================
|
|
|
|
WAV loading
|
|
|
|
===============================================================================
|
|
*/
|
|
|
|
typedef struct
|
|
{
|
|
char *wavname;
|
|
qbyte *data_p;
|
|
qbyte *iff_end;
|
|
qbyte *last_chunk;
|
|
qbyte *iff_data;
|
|
int iff_chunk_len;
|
|
} wavctx_t;
|
|
|
|
static short GetLittleShort(wavctx_t *ctx)
|
|
{
|
|
short val = 0;
|
|
val = *ctx->data_p;
|
|
val = val + (*(ctx->data_p+1)<<8);
|
|
ctx->data_p += 2;
|
|
return val;
|
|
}
|
|
|
|
static int GetLittleLong(wavctx_t *ctx)
|
|
{
|
|
int val = 0;
|
|
val = *ctx->data_p;
|
|
val = val + (*(ctx->data_p+1)<<8);
|
|
val = val + (*(ctx->data_p+2)<<16);
|
|
val = val + (*(ctx->data_p+3)<<24);
|
|
ctx->data_p += 4;
|
|
return val;
|
|
}
|
|
|
|
static unsigned int FindNextChunk(wavctx_t *ctx, char *name)
|
|
{
|
|
unsigned int dataleft;
|
|
|
|
while (1)
|
|
{
|
|
dataleft = ctx->iff_end - ctx->last_chunk;
|
|
if (dataleft < 8)
|
|
{ // didn't find the chunk
|
|
ctx->data_p = NULL;
|
|
return 0;
|
|
}
|
|
|
|
ctx->data_p=ctx->last_chunk;
|
|
ctx->data_p += 4;
|
|
dataleft-= 8;
|
|
ctx->iff_chunk_len = GetLittleLong(ctx);
|
|
if (ctx->iff_chunk_len < 0)
|
|
{
|
|
ctx->data_p = NULL;
|
|
return 0;
|
|
}
|
|
if (ctx->iff_chunk_len > dataleft)
|
|
{
|
|
Con_DPrintf ("\"%s\" seems truncated by %i bytes\n", ctx->wavname, ctx->iff_chunk_len-dataleft);
|
|
#if 1
|
|
ctx->iff_chunk_len = dataleft;
|
|
#else
|
|
ctx->data_p = NULL;
|
|
return 0;
|
|
#endif
|
|
}
|
|
|
|
dataleft-= ctx->iff_chunk_len;
|
|
// if (iff_chunk_len > 1024*1024)
|
|
// Sys_Error ("FindNextChunk: %i length is past the 1 meg sanity limit", iff_chunk_len);
|
|
ctx->data_p -= 8;
|
|
ctx->last_chunk = ctx->data_p + 8 + ctx->iff_chunk_len;
|
|
if ((ctx->iff_chunk_len&1) && dataleft)
|
|
ctx->last_chunk++;
|
|
if (!Q_strncmp(ctx->data_p, name, 4))
|
|
return ctx->iff_chunk_len;
|
|
}
|
|
}
|
|
|
|
static unsigned int FindChunk(wavctx_t *ctx, char *name)
|
|
{
|
|
ctx->last_chunk = ctx->iff_data;
|
|
return FindNextChunk (ctx, name);
|
|
}
|
|
|
|
|
|
#if 0
|
|
static 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
|
|
============
|
|
*/
|
|
static wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength)
|
|
{
|
|
wavinfo_t info;
|
|
int i;
|
|
int samples;
|
|
int chunklen;
|
|
wavctx_t ctx;
|
|
|
|
memset (&info, 0, sizeof(info));
|
|
|
|
if (!wav)
|
|
return info;
|
|
|
|
ctx.data_p = NULL;
|
|
ctx.last_chunk = NULL;
|
|
ctx.iff_chunk_len = 0;
|
|
|
|
ctx.iff_data = wav;
|
|
ctx.iff_end = wav + wavlength;
|
|
ctx.wavname = name;
|
|
|
|
// find "RIFF" chunk
|
|
chunklen = FindChunk(&ctx, "RIFF");
|
|
if (chunklen < 4 || Q_strncmp(ctx.data_p+8, "WAVE", 4))
|
|
{
|
|
Con_Printf("Missing RIFF/WAVE chunks in %s\n", name);
|
|
return info;
|
|
}
|
|
|
|
// get "fmt " chunk
|
|
ctx.iff_data = ctx.data_p + 12;
|
|
// DumpChunks ();
|
|
|
|
chunklen = FindChunk(&ctx, "fmt ");
|
|
if (chunklen < 24-8)
|
|
{
|
|
Con_Printf("Missing/truncated fmt chunk\n");
|
|
return info;
|
|
}
|
|
ctx.data_p += 8;
|
|
info.format = GetLittleShort(&ctx);
|
|
|
|
info.numchannels = GetLittleShort(&ctx);
|
|
info.rate = GetLittleLong(&ctx);
|
|
ctx.data_p += 4+2;
|
|
info.width = GetLittleShort(&ctx) / 8;
|
|
|
|
// get cue chunk
|
|
chunklen = FindChunk(&ctx, "cue ");
|
|
if (chunklen >= 36-8)
|
|
{
|
|
ctx.data_p += 32;
|
|
info.loopstart = GetLittleLong(&ctx);
|
|
// Con_Printf("loopstart=%d\n", sfx->loopstart);
|
|
|
|
// if the next chunk is a LIST chunk, look for a cue length marker
|
|
chunklen = FindNextChunk (&ctx, "LIST");
|
|
if (chunklen >= 32-8)
|
|
{
|
|
if (!strncmp (ctx.data_p + 28, "mark", 4))
|
|
{ // this is not a proper parse, but it works with cooledit...
|
|
ctx.data_p += 24;
|
|
i = GetLittleLong (&ctx); // samples in loop
|
|
info.samples = info.loopstart + i;
|
|
// Con_Printf("looped length: %i\n", i);
|
|
}
|
|
}
|
|
}
|
|
else
|
|
info.loopstart = -1;
|
|
|
|
// find data chunk
|
|
chunklen = FindChunk(&ctx, "data");
|
|
if (!ctx.data_p)
|
|
{
|
|
Con_Printf("Missing data chunk in %s\n", name);
|
|
return info;
|
|
}
|
|
|
|
ctx.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 = ctx.data_p - wav;
|
|
|
|
return info;
|
|
}
|