4c2066601a
Fixed up the -netquake / -spasm / -fitz args slightly, should actually be usable now. sv_mintic 0 is now treated as 0.013 when using nqplayerphysics, to try to make it smoother for nq clients. Preparing for astc's volume formats. Mostly for completeness, I was bored. Disabled for now because nothing supports them anyway. Fix broken mousewheel in SDL2 builds. Fix configs not getting loaded following initial downloads in the web port/etc. Make the near-cloud layer of q1 scrolling sky fully opaque by default (like vanilla). Sky fog now ignores depth, treating it as an infinite distance. Fix turbs not responding to fog. r_fullbright no longer needs vid_reload to take effect (and more efficient now). Tweaked the audio code to use an format enum instead of byte width, just with the same values still, primarily to clean up loaders that deal with S32 vs F32, or U8 vs S8. Added a cvar to control whether to use threads for the qcgc. Still disabled by default but no longer requires engine recompiles to enable! git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@5683 fc73d0e0-1445-4013-8a0c-d673dee63da5
1263 lines
30 KiB
C
1263 lines
30 KiB
C
/*
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Copyright (C) 1996-1997 Id Software, Inc.
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This program is free software; you can redistribute it and/or
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modify it under the terms of the GNU General Public License
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as published by the Free Software Foundation; either version 2
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of the License, or (at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
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See the GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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// snd_mem.c: sound caching
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#include "quakedef.h"
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#include "winquake.h"
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#include "fs.h"
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typedef struct
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{
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int format;
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int rate;
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int bitwidth;
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int numchannels;
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int loopstart;
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int samples;
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int dataofs; // chunk starts this many bytes from file start
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} wavinfo_t;
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static wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength);
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int cache_full_cycle;
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qbyte *S_Alloc (int size);
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#define LINEARUPSCALE(in, inrate, insamps, out, outrate, outlshift, outrshift) \
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{ \
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scale = inrate / (double)outrate; \
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infrac = floor(scale * 65536); \
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outsamps = insamps / scale; \
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inaccum = 0; \
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outnlsamps = floor(1.0 / scale); \
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outsamps -= outnlsamps; \
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\
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while (outsamps) \
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{ \
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*out = ((0xFFFF - inaccum)*in[0] + inaccum*in[1]) >> (16 - outlshift + outrshift); \
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inaccum += infrac; \
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in += (inaccum >> 16); \
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inaccum &= 0xFFFF; \
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out++; \
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outsamps--; \
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} \
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while (outnlsamps) \
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{ \
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*out = (*in >> outrshift) << outlshift; \
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out++; \
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outnlsamps--; \
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} \
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}
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#define LINEARUPSCALESTEREO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
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{ \
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scale = inrate / (double)outrate; \
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infrac = floor(scale * 65536); \
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outsamps = insamps / scale; \
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inaccum = 0; \
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outnlsamps = floor(1.0 / scale); \
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outsamps -= outnlsamps; \
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\
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while (outsamps) \
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{ \
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out[0] = ((0xFFFF - inaccum)*in[0] + inaccum*in[2]) >> (16 - outlshift + outrshift); \
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out[1] = ((0xFFFF - inaccum)*in[1] + inaccum*in[3]) >> (16 - outlshift + outrshift); \
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inaccum += infrac; \
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in += (inaccum >> 16) * 2; \
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inaccum &= 0xFFFF; \
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out += 2; \
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outsamps--; \
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} \
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while (outnlsamps) \
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{ \
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out[0] = (in[0] >> outrshift) << outlshift; \
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out[1] = (in[1] >> outrshift) << outlshift; \
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out += 2; \
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outnlsamps--; \
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} \
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}
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#define LINEARUPSCALESTEREOTOMONO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
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{ \
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scale = inrate / (double)outrate; \
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infrac = floor(scale * 65536); \
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outsamps = insamps / scale; \
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inaccum = 0; \
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outnlsamps = floor(1.0 / scale); \
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outsamps -= outnlsamps; \
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\
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while (outsamps) \
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{ \
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*out = ((((0xFFFF - inaccum)*in[0] + inaccum*in[2]) >> (16 - outlshift + outrshift)) + \
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(((0xFFFF - inaccum)*in[1] + inaccum*in[3]) >> (16 - outlshift + outrshift))) >> 1; \
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inaccum += infrac; \
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in += (inaccum >> 16) * 2; \
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inaccum &= 0xFFFF; \
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out++; \
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outsamps--; \
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} \
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while (outnlsamps) \
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{ \
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out[0] = (((in[0] >> outrshift) << outlshift) + ((in[1] >> outrshift) << outlshift)) >> 1; \
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out++; \
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outnlsamps--; \
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} \
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}
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#define LINEARDOWNSCALE(in, inrate, insamps, out, outrate, outlshift, outrshift) \
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{ \
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scale = outrate / (double)inrate; \
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infrac = floor(scale * 65536); \
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inaccum = 0; \
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insamps--; \
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outsampleft = 0; \
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\
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while (insamps) \
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{ \
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inaccum += infrac; \
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if (inaccum >> 16) \
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{ \
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inaccum &= 0xFFFF; \
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outsampleft += (infrac - inaccum) * (*in); \
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*out = outsampleft >> (16 - outlshift + outrshift); \
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out++; \
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outsampleft = inaccum * (*in); \
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} \
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else \
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outsampleft += infrac * (*in); \
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in++; \
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insamps--; \
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} \
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outsampleft += (0xFFFF - inaccum) * (*in);\
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*out = outsampleft >> (16 - outlshift + outrshift); \
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}
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#define LINEARDOWNSCALESTEREO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
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{ \
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scale = outrate / (double)inrate; \
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infrac = floor(scale * 65536); \
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inaccum = 0; \
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insamps--; \
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outsampleft = 0; \
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outsampright = 0; \
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\
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while (insamps) \
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{ \
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inaccum += infrac; \
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if (inaccum >> 16) \
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{ \
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inaccum &= 0xFFFF; \
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outsampleft += (infrac - inaccum) * in[0]; \
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outsampright += (infrac - inaccum) * in[1]; \
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out[0] = outsampleft >> (16 - outlshift + outrshift); \
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out[1] = outsampright >> (16 - outlshift + outrshift); \
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out += 2; \
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outsampleft = inaccum * in[0]; \
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outsampright = inaccum * in[1]; \
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} \
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else \
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{ \
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outsampleft += infrac * in[0]; \
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outsampright += infrac * in[1]; \
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} \
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in += 2; \
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insamps--; \
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} \
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outsampleft += (0xFFFF - inaccum) * in[0];\
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outsampright += (0xFFFF - inaccum) * in[1];\
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out[0] = outsampleft >> (16 - outlshift + outrshift); \
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out[1] = outsampright >> (16 - outlshift + outrshift); \
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}
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#define LINEARDOWNSCALESTEREOTOMONO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
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{ \
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scale = outrate / (double)inrate; \
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infrac = floor(scale * 65536); \
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inaccum = 0; \
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insamps--; \
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outsampleft = 0; \
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\
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while (insamps) \
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{ \
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inaccum += infrac; \
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if (inaccum >> 16) \
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{ \
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inaccum &= 0xFFFF; \
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outsampleft += (infrac - inaccum) * ((in[0] + in[1]) >> 1); \
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*out = outsampleft >> (16 - outlshift + outrshift); \
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out++; \
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outsampleft = inaccum * ((in[0] + in[1]) >> 1); \
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} \
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else \
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outsampleft += infrac * ((in[0] + in[1]) >> 1); \
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in += 2; \
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insamps--; \
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} \
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outsampleft += (0xFFFF - inaccum) * ((in[0] + in[1]) >> 1);\
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*out = outsampleft >> (16 - outlshift + outrshift); \
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}
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#define STANDARDRESCALE(in, inrate, insamps, out, outrate, outlshift, outrshift) \
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{ \
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scale = inrate / (double)outrate; \
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infrac = floor(scale * 65536); \
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outsamps = insamps / scale; \
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inaccum = 0; \
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\
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while (outsamps) \
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{ \
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*out = (*in >> outrshift) << outlshift; \
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inaccum += infrac; \
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in += (inaccum >> 16); \
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inaccum &= 0xFFFF; \
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out++; \
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outsamps--; \
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} \
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}
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#define STANDARDRESCALESTEREO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
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{ \
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scale = inrate / (double)outrate; \
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infrac = floor(scale * 65536); \
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outsamps = insamps / scale; \
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inaccum = 0; \
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\
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while (outsamps) \
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{ \
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out[0] = (in[0] >> outrshift) << outlshift; \
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out[1] = (in[1] >> outrshift) << outlshift; \
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inaccum += infrac; \
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in += (inaccum >> 16) * 2; \
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inaccum &= 0xFFFF; \
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out += 2; \
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outsamps--; \
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} \
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}
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#define STANDARDRESCALESTEREOTOMONO(in, inrate, insamps, out, outrate, outlshift, outrshift) \
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{ \
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scale = inrate / (double)outrate; \
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infrac = floor(scale * 65536); \
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outsamps = insamps / scale; \
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inaccum = 0; \
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\
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while (outsamps) \
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{ \
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out[0] = (((in[0] >> outrshift) << outlshift) + ((in[1] >> outrshift) << outlshift)) >> 1; \
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inaccum += infrac; \
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in += (inaccum >> 16) * 2; \
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inaccum &= 0xFFFF; \
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out++; \
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outsamps--; \
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} \
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}
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#define QUICKCONVERT(in, insamps, out, outlshift, outrshift) \
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{ \
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while (insamps) \
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{ \
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*out = (*in >> outrshift) << outlshift; \
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out++; \
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in++; \
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insamps--; \
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} \
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}
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#define QUICKCONVERTSTEREOTOMONO(in, insamps, out, outlshift, outrshift) \
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{ \
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while (insamps) \
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{ \
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*out = (((in[0] >> outrshift) << outlshift) + ((in[1] >> outrshift) << outlshift)) >> 1; \
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out++; \
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in += 2; \
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insamps--; \
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} \
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}
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// SND_ResampleStream: takes a sound stream and converts with given parameters. Limited to
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// 8-16-bit signed conversions and mono-to-mono/stereo-to-stereo conversions.
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// Not an in-place algorithm.
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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)
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{
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double scale;
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signed char *in8 = (signed char *)in;
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short *in16 = (short *)in;
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signed char *out8 = (signed char *)out;
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short *out16 = (short *)out;
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int outsamps, outnlsamps, outsampleft, outsampright;
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int infrac, inaccum;
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if (insamps <= 0)
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return;
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if (inchannels == outchannels && informat == outformat && inrate == outrate)
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{
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memcpy(out, in, informat*insamps*inchannels);
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return;
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}
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if (inchannels == 1 && outchannels == 1)
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{
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if (informat == QAF_S8)
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{
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if (outformat == QAF_S8)
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{
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if (inrate < outrate) // upsample
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{
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if (resampstyle)
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LINEARUPSCALE(in8, inrate, insamps, out8, outrate, 0, 0)
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else
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STANDARDRESCALE(in8, inrate, insamps, out8, outrate, 0, 0)
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}
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else // downsample
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{
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if (resampstyle > 1)
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LINEARDOWNSCALE(in8, inrate, insamps, out8, outrate, 0, 0)
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else
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STANDARDRESCALE(in8, inrate, insamps, out8, outrate, 0, 0)
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}
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return;
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}
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else if (outformat == QAF_S16)
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{
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if (inrate == outrate) // quick convert
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QUICKCONVERT(in8, insamps, out16, 8, 0)
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else if (inrate < outrate) // upsample
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{
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if (resampstyle)
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LINEARUPSCALE(in8, inrate, insamps, out16, outrate, 8, 0)
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else
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STANDARDRESCALE(in8, inrate, insamps, out16, outrate, 8, 0)
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}
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else // downsample
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{
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if (resampstyle > 1)
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LINEARDOWNSCALE(in8, inrate, insamps, out16, outrate, 8, 0)
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else
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STANDARDRESCALE(in8, inrate, insamps, out16, outrate, 8, 0)
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}
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return;
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}
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}
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else if (informat == QAF_S16) // 16-bit
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{
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if (outformat == QAF_S16)
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{
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if (inrate < outrate) // upsample
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{
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if (resampstyle)
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LINEARUPSCALE(in16, inrate, insamps, out16, outrate, 0, 0)
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else
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STANDARDRESCALE(in16, inrate, insamps, out16, outrate, 0, 0)
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}
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else // downsample
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{
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if (resampstyle > 1)
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LINEARDOWNSCALE(in16, inrate, insamps, out16, outrate, 0, 0)
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else
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STANDARDRESCALE(in16, inrate, insamps, out16, outrate, 0, 0)
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}
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return;
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}
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else if (outformat == QAF_S8)
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{
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if (inrate == outrate) // quick convert
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QUICKCONVERT(in16, insamps, out8, 0, 8)
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else if (inrate < outrate) // upsample
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{
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if (resampstyle)
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LINEARUPSCALE(in16, inrate, insamps, out8, outrate, 0, 8)
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else
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STANDARDRESCALE(in16, inrate, insamps, out8, outrate, 0, 8)
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}
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else // downsample
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{
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if (resampstyle > 1)
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LINEARDOWNSCALE(in16, inrate, insamps, out8, outrate, 0, 8)
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else
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STANDARDRESCALE(in16, inrate, insamps, out8, outrate, 0, 8)
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}
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return;
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}
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}
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}
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else if (outchannels == 2 && inchannels == 2)
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{
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if (informat == QAF_S8)
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{
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if (outformat == QAF_S8)
|
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{
|
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if (inrate < outrate) // upsample
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{
|
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if (resampstyle)
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LINEARUPSCALESTEREO(in8, inrate, insamps, out8, outrate, 0, 0)
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else
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STANDARDRESCALESTEREO(in8, inrate, insamps, out8, outrate, 0, 0)
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}
|
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else // downsample
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{
|
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if (resampstyle > 1)
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LINEARDOWNSCALESTEREO(in8, inrate, insamps, out8, outrate, 0, 0)
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else
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STANDARDRESCALESTEREO(in8, inrate, insamps, out8, outrate, 0, 0)
|
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}
|
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return;
|
|
}
|
|
else
|
|
{
|
|
if (inrate == outrate) // quick convert
|
|
{
|
|
insamps *= 2;
|
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QUICKCONVERT(in8, insamps, out16, 8, 0)
|
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}
|
|
else if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
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LINEARUPSCALESTEREO(in8, inrate, insamps, out16, outrate, 8, 0)
|
|
else
|
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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 (loadas8bit.ival < 0)
|
|
outformat = QAF_S16;
|
|
else if (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 == 4) //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->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 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;
|
|
}
|