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fteqw/engine/client/snd_mem.c
Spoike 4d25c073ec big commit. :( 
fix crash from qwplayers with invalid modelindexes
rework allow_skybox a little. now applies immediately.
try to fix the appears-outside-of-map bug, again.
dir *.wav now shows a little extra info on mouse-over.
try loading music file extensions that we expect to be able to play via ffmpeg, not just the ones that are directly supported.
rework the hidden fps_presets, show them with tab completion.
fix a possible crash with r_temporalscenecache
fix lightmap updates on submodels not happening properly with the scenecache.
fix the serious memory leak with scenecache.
add r_glsl_pbr cvar to force use of pbr pathways in our glsl.
fix bug in alsa output not supporting float output properly.
preliminary work to have the mixer use floating point mixing. disabled for now.
try to update sys_register_file_associations on linux, still needs work though.
try to work around nquake's config quirks, so config files don't get overwritten.
repackage quake's conchars in order to add padding. this should avoid extra junk on the outside of glyphs.
give fteqcc(commandline version) some extra package extration/creation support. our paks should be more end-user-friendly, and our spanned pk3s are awesome.
write rune state to legacy saved games, so we don't ever have the missing-runes bug ever again...



git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@5780 fc73d0e0-1445-4013-8a0c-d673dee63da5
2020-10-26 06:30:35 +00:00

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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 "fs.h"
typedef struct
{
int format;
int rate;
int bitwidth;
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, qaudiofmt_t informat, int inchannels, int insamps, void *out, int outrate, qaudiofmt_t outformat, 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 && informat == outformat && inrate == outrate)
{
memcpy(out, in, informat*insamps*inchannels);
return;
}
if (inchannels == 1 && outchannels == 1)
{
if (informat == QAF_S8)
{
if (outformat == QAF_S8)
{
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 (outformat == QAF_S16)
{
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 if (informat == QAF_S16) // 16-bit
{
if (outformat == QAF_S16)
{
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 (outformat == QAF_S8)
{
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 (informat == QAF_S8)
{
if (outformat == QAF_S8)
{
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)
}
return;
}
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 if (informat == QAF_S16) // 16-bit
{
if (outformat == QAF_S16)
{
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 (outformat == QAF_S8)
{
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 (informat == QAF_S8)
{
if (outformat == QAF_S8)
{
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 (outformat == QAF_S16)
{
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 if (informat == QAF_S16) // 16-bit
{
if (outformat == QAF_S16)
{
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 (outformat == QAF_S8)
{
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, qaudiofmt_t informat, int insamps, int inloopstart, qbyte *data)
{
extern cvar_t snd_linearresample;
extern cvar_t snd_loadasstereo;
double scale;
sfxcache_t *sc;
int outsamps;
int len;
qaudiofmt_t outformat;
scale = snd_speed / (double)inrate;
outsamps = insamps * scale;
if (snd_loadas8bit.ival < 0)
outformat = QAF_S16;
else if (snd_loadas8bit.ival)
outformat = QAF_S8;
else
outformat = informat;
len = outsamps * QAF_BYTES(outformat) * inchannels;
sfx->decoder.buf = sc = BZ_Malloc(sizeof(sfxcache_t) + len);
if (!sc)
{
return false;
}
sc->numchannels = inchannels;
sc->format = outformat;
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,
informat,
inchannels,
insamps,
sc->data,
sc->speed,
sc->format,
sc->numchannels,
snd_linearresample.ival);
if (inchannels == 1 && snd_loadasstereo.ival)
{ //I'm implementing this to work around what looks like a firefox bug, where mono buffers don't get played (but stereo works just fine despite all the spacialisation issues associated with that).
sfxcache_t *nc = sfx->decoder.buf = BZ_Malloc(sizeof(sfxcache_t) + len*2);
*nc = *sc;
nc->data = (qbyte*)(nc+1);
SND_ResampleStream (sc->data,
sc->speed,
sc->format,
sc->numchannels,
outsamps,
nc->data,
nc->speed*2,
nc->format,
nc->numchannels,
false);
nc->numchannels *= 2;
BZ_Free(sc);
}
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, qboolean forcedecode)
{
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, qboolean forcedecode)
{
// 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
static qboolean QDECL S_LoadWavSound (sfx_t *s, qbyte *data, size_t datalen, int sndspeed, qboolean forcedecode)
{
wavinfo_t info;
qaudiofmt_t format;
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.bitwidth == 8) //unsigned bytes
{
COM_CharBias(data + info.dataofs, info.samples*info.numchannels);
format = QAF_S8;
}
else if (info.format == 1 && info.bitwidth == 16) //signed shorts
{
COM_SwapLittleShortBlock((short *)(data + info.dataofs), info.samples*info.numchannels);
format = QAF_S16;
}
else if (info.format == 1 && info.bitwidth == 32) //24 or 32bit int audio
{
short *out = (short *)(data + info.dataofs);
int *in = (int *)(data + info.dataofs);
size_t samples = info.samples*info.numchannels;
while(samples --> 0)
{ //in place size conversion, so we need to do it forwards.
*out++ = LittleLong(*in++)>>16; //just drop the least significant bits.
}
format = QAF_S16;
}
#ifdef MIXER_F32
else if (info.format == 3 && info.bitwidth == 32) //signed floats
{
if (bigendian)
{
size_t i = info.samples*info.numchannels;
float *ptr = (float*)(data + info.dataofs);
while(i --> 0)
ptr[i] = LittleFloat(ptr[i]);
}
format = QAF_F32;
}
#else
else if (info.format == 3 && info.bitwidth == 32) //signed floats
{
short *out = (short *)(data + info.dataofs);
float *in = (float *)(data + info.dataofs);
size_t samples = info.samples*info.numchannels;
int t;
while(samples --> 0)
{ //in place size conversion, so we need to do it forwards.
t = LittleFloat(*in++) * 32767;
t = bound(-32768, t, 32767);
*out++ = t;
}
format = QAF_S16;
}
#endif
else
{
s->loadstate = SLS_FAILED;
switch(info.format)
{
case 1/*WAVE_FORMAT_PCM*/:
case 3/*WAVE_FORMAT_IEEE_FLOAT*/: Con_Printf ("%s has an unsupported width (%i bits).\n", s->name, info.bitwidth); break;
case 6/*WAVE_FORMAT_ALAW*/: Con_Printf ("%s uses unsupported a-law format.\n", s->name); break;
case 7/*WAVE_FORMAT_MULAW*/: Con_Printf ("%s uses unsupported mu-law format.\n", s->name); break;
case 0xfffe/*WAVE_FORMAT_EXTENSIBLE*/:
default: Con_Printf ("%s has an unsupported format (%#x).\n", s->name, info.format); break;
}
return false;
}
return ResampleSfx (s, info.rate, info.numchannels, format, info.samples, info.loopstart, data + info.dataofs);
}
qboolean QDECL S_LoadOVSound (sfx_t *s, qbyte *data, size_t datalen, int sndspeed, qboolean forcedecode);
#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, qboolean forcedecode)
{
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->format = QAF_BLOB; //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 struct
{
S_LoadSound_t loadfunc;
void *module;
} 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(void *module, S_LoadSound_t loadfnc)
{
int i;
for (i = 0; i < sizeof(AudioInputPlugins)/sizeof(AudioInputPlugins[0]); i++)
{
if (!AudioInputPlugins[i].loadfunc)
{
AudioInputPlugins[i].module = module;
AudioInputPlugins[i].loadfunc = loadfnc;
return true;
}
}
return false;
}
void S_UnregisterSoundInputModule(void *module)
{ //unregister all sound handlers for the given module.
int i;
for (i = 0; i < sizeof(AudioInputPlugins)/sizeof(AudioInputPlugins[0]); i++)
{
if (AudioInputPlugins[i].module == module)
{
AudioInputPlugins[i].module = NULL;
AudioInputPlugins[i].loadfunc = NULL;
}
}
}
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 forcedecode, 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)
{
static float throttletimer;
Con_ThrottlePrintf(&throttletimer, 1, "S_LoadSound: %s%s requested, but could only find %s\n", prefixes[pre], name, namebuffer);
break;
}
}
}
if (data)
Validation_FileLoaded(name, data, filesize);
}
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].loadfunc)
{
if (AudioInputPlugins[i].loadfunc(s, data, filesize, snd_speed, forcedecode))
{
//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, qboolean force)
{
if (s->loadstate == SLS_NOTLOADED && sndcardinfo)
{
s->loadstate = SLS_LOADING;
COM_AddWork(WG_LOADER, S_LoadSoundWorker, s, NULL, force, 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)
{
extern cvar_t snd_ignorecueloops;
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.bitwidth = GetLittleShort(&ctx);
// get cue chunk
chunklen = FindChunk(&ctx, "cue ");
if (chunklen >= 36-8 && !snd_ignorecueloops.ival)
{
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<<3) / info.bitwidth / 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;
}
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