53a7b3d47c
The ragdoll API is potentially usable now, but still really limited. Enabled SQL requests by default using sqlite. Note that you'll need the sqlite dll to use this. MySQL should still be usable, but I didn't try. MySQL requires -DUSE_MYSQL to compile it, and a dll and -mysql argument to enable it. Fixed nacl. NPFTE plugin now invokes an exe to run the game rather than running the game within the browser. externvalue builtin now accepts & prefix to return a pointer instead. Fixed vector autocvars. uri_get, bufstr_add, bufstr_free, now functional. QC debugger can now show asm if line numbers are not available. Added support for QC watchpoints. Use the watchpoint command. gl_specular now give specular even without rtlights, thankfully not as blatently, but its there. android will not crash due to supported audio formats, and gles2 can be selected via a cvar (requires full FTEDroidActivity/program restart). git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@4152 fc73d0e0-1445-4013-8a0c-d673dee63da5
1101 lines
25 KiB
C
1101 lines
25 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"
|
|
|
|
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
|
|
================
|
|
*/
|
|
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)
|
|
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);
|
|
|
|
return true;
|
|
}
|
|
|
|
//=============================================================================
|
|
#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;
|
|
}
|
|
*/
|
|
qboolean 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;
|
|
|
|
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
|
|
|
|
qboolean S_LoadWavSound (sfx_t *s, qbyte *data, int 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->failedload = true;
|
|
Con_Printf ("%s has an unsupported quantity of channels.\n",s->name);
|
|
return false;
|
|
}
|
|
|
|
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);
|
|
|
|
return ResampleSfx (s, info.rate, info.numchannels, info.width, info.samples, info.loopstart, data + info.dataofs);
|
|
}
|
|
|
|
qboolean 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
|
|
==============
|
|
*/
|
|
|
|
qboolean S_LoadSound (sfx_t *s)
|
|
{
|
|
char stackbuf[65536];
|
|
char namebuffer[256];
|
|
qbyte *data;
|
|
int i;
|
|
size_t result;
|
|
char *name = s->name;
|
|
|
|
if (s->failedload)
|
|
return false; //it failed to load once before, don't bother trying again.
|
|
|
|
// see if still in memory
|
|
if (s->decoder.buf)
|
|
return true;
|
|
|
|
if (name[1] == ':' && name[2] == '\\')
|
|
{
|
|
vfsfile_t *f;
|
|
int fsize;
|
|
#ifndef _WIN32 //convert from windows to a suitable alternative.
|
|
char unixname[128];
|
|
Q_snprintfz(unixname, sizeof(unixname), "/mnt/%c/%s", name[0]-'A'+'a', name+3);
|
|
name = unixname;
|
|
while (*name)
|
|
{
|
|
if (*name == '\\')
|
|
*name = '/';
|
|
name++;
|
|
}
|
|
name = unixname;
|
|
#endif
|
|
|
|
|
|
if ((f = VFSOS_Open(name, "rb")))
|
|
{
|
|
fsize = VFS_GETLEN(f);
|
|
data = Hunk_TempAlloc (fsize);
|
|
result = VFS_READ(f, data, fsize);
|
|
|
|
if (result != fsize)
|
|
Con_SafePrintf("S_LoadSound() fread: Filename: %s, expected %i, result was %u\n", name, fsize, (unsigned int)result);
|
|
|
|
VFS_CLOSE(f);
|
|
}
|
|
else
|
|
{
|
|
Con_SafePrintf ("Couldn't load %s\n", namebuffer);
|
|
return false;
|
|
}
|
|
}
|
|
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 false;
|
|
}
|
|
|
|
s->failedload = false;
|
|
|
|
for (i = sizeof(AudioInputPlugins)/sizeof(AudioInputPlugins[0])-1; i >= 0; i--)
|
|
{
|
|
if (AudioInputPlugins[i])
|
|
{
|
|
if (AudioInputPlugins[i](s, data, com_filesize, snd_speed))
|
|
{
|
|
return true;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (!s->failedload)
|
|
Con_Printf ("Format not recognised: %s\n", namebuffer);
|
|
|
|
s->failedload = true;
|
|
return false;
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
===============================================================================
|
|
|
|
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;
|
|
}
|