mirror of
https://github.com/nzp-team/fteqw.git
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27a59a0cbc
vulkan, wasapi, quake injector features added. irc, avplug, cef plugins/drivers reworked/updated/added openal reverb, doppler effects added. 'dir' console command now attempts to view clicked files. lots of warning fixes, should now only be deprecation warnings for most targets (depending on compiler version anyway...). SendEntity finally reworked to use flags properly. effectinfo improved, other smc-targetted fixes. mapcluster stuff now has support for linux. .basebone+.baseframe now exist in ssqc. qcc: -Fqccx supports qccx syntax, including qccx hacks. don't expect these to work in fteqw nor dp though. qcc: rewrote function call handling to use refs rather than defs. this makes struct passing more efficient and makes the __out keyword usable with fields etc. qccgui: can cope a little better with non-unicode files. can now represent most quake chars. qcc: suppressed warnings from *extensions.qc git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@5000 fc73d0e0-1445-4013-8a0c-d673dee63da5
1155 lines
26 KiB
C
1155 lines
26 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 rate;
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int width;
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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, int inwidth, int inchannels, int insamps, void *out, int outrate, int outwidth, 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 && inwidth == outwidth && inrate == outrate)
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{
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memcpy(out, in, inwidth*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 (inwidth == 1)
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{
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if (outwidth == 1)
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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
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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 // 16-bit
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{
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if (outwidth == 2)
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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
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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 (inwidth == 1)
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{
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if (outwidth == 1)
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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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}
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else
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{
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if (inrate == outrate) // quick convert
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{
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insamps *= 2;
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QUICKCONVERT(in8, insamps, out16, 8, 0)
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}
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else if (inrate < outrate) // upsample
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{
|
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if (resampstyle)
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LINEARUPSCALESTEREO(in8, inrate, insamps, out16, outrate, 8, 0)
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else
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STANDARDRESCALESTEREO(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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LINEARDOWNSCALESTEREO(in8, inrate, insamps, out16, outrate, 8, 0)
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else
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STANDARDRESCALESTEREO(in8, inrate, insamps, out16, outrate, 8, 0)
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}
|
|
}
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|
}
|
|
else // 16-bit
|
|
{
|
|
if (outwidth == 2)
|
|
{
|
|
if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
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LINEARUPSCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
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|
else
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STANDARDRESCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
|
|
}
|
|
else // downsample
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|
{
|
|
if (resampstyle > 1)
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LINEARDOWNSCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
|
|
else
|
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STANDARDRESCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
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}
|
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}
|
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else
|
|
{
|
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if (inrate == outrate) // quick convert
|
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{
|
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insamps *= 2;
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QUICKCONVERT(in16, insamps, out8, 0, 8)
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}
|
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else if (inrate < outrate) // upsample
|
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{
|
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if (resampstyle)
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LINEARUPSCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
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else
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STANDARDRESCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
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}
|
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else // downsample
|
|
{
|
|
if (resampstyle > 1)
|
|
LINEARDOWNSCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
else
|
|
STANDARDRESCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
}
|
|
}
|
|
}
|
|
}
|
|
#if 0
|
|
else if (outchannels == 1 && inchannels == 2)
|
|
{
|
|
if (inwidth == 1)
|
|
{
|
|
if (outwidth == 1)
|
|
{
|
|
if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
|
LINEARUPSCALESTEREOTOMONO(in8, inrate, insamps, out8, outrate, 0, 0)
|
|
else
|
|
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out8, outrate, 0, 0)
|
|
}
|
|
else // downsample
|
|
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out8, outrate, 0, 0)
|
|
}
|
|
else
|
|
{
|
|
if (inrate == outrate) // quick convert
|
|
QUICKCONVERTSTEREOTOMONO(in8, insamps, out16, 8, 0)
|
|
else if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
|
LINEARUPSCALESTEREOTOMONO(in8, inrate, insamps, out16, outrate, 8, 0)
|
|
else
|
|
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out16, outrate, 8, 0)
|
|
}
|
|
else // downsample
|
|
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out16, outrate, 8, 0)
|
|
}
|
|
}
|
|
else // 16-bit
|
|
{
|
|
if (outwidth == 2)
|
|
{
|
|
if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
|
LINEARUPSCALESTEREOTOMONO(in16, inrate, insamps, out16, outrate, 0, 0)
|
|
else
|
|
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out16, outrate, 0, 0)
|
|
}
|
|
else // downsample
|
|
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out16, outrate, 0, 0)
|
|
}
|
|
else
|
|
{
|
|
if (inrate == outrate) // quick convert
|
|
QUICKCONVERTSTEREOTOMONO(in16, insamps, out8, 0, 8)
|
|
else if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
|
LINEARUPSCALESTEREOTOMONO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
else
|
|
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
}
|
|
else // downsample
|
|
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
}
|
|
|
|
/*
|
|
================
|
|
ResampleSfx
|
|
================
|
|
*/
|
|
static qboolean ResampleSfx (sfx_t *sfx, int inrate, int inchannels, int inwidth, int insamps, int inloopstart, qbyte *data)
|
|
{
|
|
extern cvar_t snd_linearresample;
|
|
double scale;
|
|
sfxcache_t *sc;
|
|
int outsamps;
|
|
int len;
|
|
int outwidth;
|
|
|
|
scale = snd_speed / (double)inrate;
|
|
outsamps = insamps * scale;
|
|
if (loadas8bit.ival < 0)
|
|
outwidth = 2;
|
|
else if (loadas8bit.ival)
|
|
outwidth = 1;
|
|
else
|
|
outwidth = inwidth;
|
|
len = outsamps * outwidth * inchannels;
|
|
|
|
sfx->decoder.buf = sc = BZ_Malloc(len + sizeof(sfxcache_t));
|
|
if (!sc)
|
|
{
|
|
return false;
|
|
}
|
|
|
|
sc->numchannels = inchannels;
|
|
sc->width = outwidth;
|
|
sc->speed = snd_speed;
|
|
sc->length = outsamps;
|
|
sc->soundoffset = 0;
|
|
sc->data = (qbyte*)(sc+1);
|
|
if (inloopstart == -1)
|
|
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;
|
|
}
|
|
*/
|
|
static 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
|
|
|
|
static 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->loadstate = SLS_FAILED;
|
|
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);
|
|
|
|
#ifdef FTE_TARGET_WEB
|
|
//web browsers contain their own decoding libraries that our openal stuff can use.
|
|
static qboolean S_LoadBrowserFile (sfx_t *s, qbyte *data, int datalen, int sndspeed)
|
|
{
|
|
sfxcache_t *sc;
|
|
s->decoder.buf = sc = BZ_Malloc(sizeof(sfxcache_t) + datalen);
|
|
sc->data = (qbyte*)(sc+1);
|
|
sc->length = datalen;
|
|
sc->width = 0; //ie: not pcm
|
|
sc->loopstart = -1;
|
|
sc->speed = sndspeed;
|
|
sc->numchannels = 2;
|
|
sc->soundoffset = 0;
|
|
memcpy(sc->data, data, datalen);
|
|
|
|
return true;
|
|
}
|
|
#endif
|
|
|
|
//highest priority is last.
|
|
static S_LoadSound_t AudioInputPlugins[10] =
|
|
{
|
|
#ifdef FTE_TARGET_WEB
|
|
S_LoadBrowserFile,
|
|
#endif
|
|
#ifdef AVAIL_OGGVORBIS
|
|
S_LoadOVSound,
|
|
#endif
|
|
S_LoadWavSound,
|
|
#ifdef 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;
|
|
}
|
|
|
|
static void S_LoadedOrFailed (void *ctx, void *ctxdata, size_t a, size_t b)
|
|
{
|
|
sfx_t *s = ctx;
|
|
s->loadstate = a;
|
|
}
|
|
/*
|
|
==============
|
|
S_LoadSound
|
|
==============
|
|
*/
|
|
|
|
static void S_LoadSoundWorker (void *ctx, void *ctxdata, size_t a, size_t b)
|
|
{
|
|
sfx_t *s = ctx;
|
|
char namebuffer[256];
|
|
qbyte *data;
|
|
int i;
|
|
size_t result;
|
|
char *name = s->name;
|
|
size_t filesize;
|
|
|
|
if (name[1] == ':' && name[2] == '\\')
|
|
{
|
|
vfsfile_t *f;
|
|
#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")))
|
|
{
|
|
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
|
|
|
|
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;
|
|
}
|
|
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_LoadFile(namebuffer, 5, &filesize);
|
|
}
|
|
|
|
// Con_Printf ("loading %s\n",namebuffer);
|
|
|
|
if (!data)
|
|
data = COM_LoadFile(name, 5, &filesize);
|
|
if (!data)
|
|
{
|
|
char altname[sizeof(namebuffer)];
|
|
COM_StripExtension(namebuffer, altname, sizeof(altname));
|
|
COM_DefaultExtension(altname, ".ogg", sizeof(altname));
|
|
data = COM_LoadFile(altname, 5, &filesize);
|
|
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);
|
|
COM_AddWork(WG_MAIN, S_LoadedOrFailed, s, NULL, SLS_FAILED, 0);
|
|
return;
|
|
}
|
|
|
|
for (i = sizeof(AudioInputPlugins)/sizeof(AudioInputPlugins[0])-1; i >= 0; i--)
|
|
{
|
|
if (AudioInputPlugins[i])
|
|
{
|
|
if (AudioInputPlugins[i](s, data, filesize, snd_speed))
|
|
{
|
|
//wake up the main thread in case it decided to wait for us.
|
|
COM_AddWork(WG_MAIN, S_LoadedOrFailed, s, NULL, SLS_LOADED, 0);
|
|
BZ_Free(data);
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (s->loadstate != SLS_FAILED)
|
|
Con_Printf ("Format not recognised: %s\n", namebuffer);
|
|
|
|
COM_AddWork(WG_MAIN, S_LoadedOrFailed, s, NULL, SLS_FAILED, 0);
|
|
BZ_Free(data);
|
|
return;
|
|
}
|
|
|
|
qboolean S_LoadSound (sfx_t *s)
|
|
{
|
|
if (s->loadstate == SLS_NOTLOADED && sndcardinfo)
|
|
{
|
|
s->loadstate = SLS_LOADING;
|
|
COM_AddWork(WG_LOADER, S_LoadSoundWorker, s, NULL, 0, 0);
|
|
}
|
|
if (s->loadstate == SLS_FAILED)
|
|
return false; //it failed to load once before, don't bother trying again.
|
|
|
|
return true; //loaded okay, or still loading
|
|
}
|
|
|
|
/*
|
|
===============================================================================
|
|
|
|
WAV loading
|
|
|
|
===============================================================================
|
|
*/
|
|
|
|
typedef struct
|
|
{
|
|
char *wavname;
|
|
qbyte *data_p;
|
|
qbyte *iff_end;
|
|
qbyte *last_chunk;
|
|
qbyte *iff_data;
|
|
int iff_chunk_len;
|
|
} wavctx_t;
|
|
|
|
static short GetLittleShort(wavctx_t *ctx)
|
|
{
|
|
short val = 0;
|
|
val = *ctx->data_p;
|
|
val = val + (*(ctx->data_p+1)<<8);
|
|
ctx->data_p += 2;
|
|
return val;
|
|
}
|
|
|
|
static int GetLittleLong(wavctx_t *ctx)
|
|
{
|
|
int val = 0;
|
|
val = *ctx->data_p;
|
|
val = val + (*(ctx->data_p+1)<<8);
|
|
val = val + (*(ctx->data_p+2)<<16);
|
|
val = val + (*(ctx->data_p+3)<<24);
|
|
ctx->data_p += 4;
|
|
return val;
|
|
}
|
|
|
|
static unsigned int FindNextChunk(wavctx_t *ctx, char *name)
|
|
{
|
|
unsigned int dataleft;
|
|
|
|
while (1)
|
|
{
|
|
dataleft = ctx->iff_end - ctx->last_chunk;
|
|
if (dataleft < 8)
|
|
{ // didn't find the chunk
|
|
ctx->data_p = NULL;
|
|
return 0;
|
|
}
|
|
|
|
ctx->data_p=ctx->last_chunk;
|
|
ctx->data_p += 4;
|
|
dataleft-= 8;
|
|
ctx->iff_chunk_len = GetLittleLong(ctx);
|
|
if (ctx->iff_chunk_len < 0)
|
|
{
|
|
ctx->data_p = NULL;
|
|
return 0;
|
|
}
|
|
if (ctx->iff_chunk_len > dataleft)
|
|
{
|
|
Con_DPrintf ("\"%s\" seems truncated by %i bytes\n", ctx->wavname, ctx->iff_chunk_len-dataleft);
|
|
#if 1
|
|
ctx->iff_chunk_len = dataleft;
|
|
#else
|
|
ctx->data_p = NULL;
|
|
return 0;
|
|
#endif
|
|
}
|
|
|
|
dataleft-= ctx->iff_chunk_len;
|
|
// if (iff_chunk_len > 1024*1024)
|
|
// Sys_Error ("FindNextChunk: %i length is past the 1 meg sanity limit", iff_chunk_len);
|
|
ctx->data_p -= 8;
|
|
ctx->last_chunk = ctx->data_p + 8 + ctx->iff_chunk_len;
|
|
if ((ctx->iff_chunk_len&1) && dataleft)
|
|
ctx->last_chunk++;
|
|
if (!Q_strncmp(ctx->data_p, name, 4))
|
|
return ctx->iff_chunk_len;
|
|
}
|
|
}
|
|
|
|
static unsigned int FindChunk(wavctx_t *ctx, char *name)
|
|
{
|
|
ctx->last_chunk = ctx->iff_data;
|
|
return FindNextChunk (ctx, name);
|
|
}
|
|
|
|
|
|
#if 0
|
|
static void DumpChunks(void)
|
|
{
|
|
char str[5];
|
|
|
|
str[4] = 0;
|
|
data_p=iff_data;
|
|
do
|
|
{
|
|
memcpy (str, data_p, 4);
|
|
data_p += 4;
|
|
iff_chunk_len = GetLittleLong();
|
|
Con_Printf ("0x%x : %s (%d)\n", (int)(data_p - 4), str, iff_chunk_len);
|
|
data_p += (iff_chunk_len + 1) & ~1;
|
|
} while (data_p < iff_end);
|
|
}
|
|
#endif
|
|
|
|
/*
|
|
============
|
|
GetWavinfo
|
|
============
|
|
*/
|
|
static wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength)
|
|
{
|
|
wavinfo_t info;
|
|
int i;
|
|
int format;
|
|
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;
|
|
format = GetLittleShort(&ctx);
|
|
if (format != 1)
|
|
{
|
|
Con_Printf("Microsoft PCM format only\n");
|
|
return info;
|
|
}
|
|
|
|
info.numchannels = GetLittleShort(&ctx);
|
|
info.rate = GetLittleLong(&ctx);
|
|
ctx.data_p += 4+2;
|
|
info.width = GetLittleShort(&ctx) / 8;
|
|
|
|
// get cue chunk
|
|
chunklen = FindChunk(&ctx, "cue ");
|
|
if (chunklen >= 36-8)
|
|
{
|
|
ctx.data_p += 32;
|
|
info.loopstart = GetLittleLong(&ctx);
|
|
// Con_Printf("loopstart=%d\n", sfx->loopstart);
|
|
|
|
// if the next chunk is a LIST chunk, look for a cue length marker
|
|
chunklen = FindNextChunk (&ctx, "LIST");
|
|
if (chunklen >= 32-8)
|
|
{
|
|
if (!strncmp (ctx.data_p + 28, "mark", 4))
|
|
{ // this is not a proper parse, but it works with cooledit...
|
|
ctx.data_p += 24;
|
|
i = GetLittleLong (&ctx); // samples in loop
|
|
info.samples = info.loopstart + i;
|
|
// Con_Printf("looped length: %i\n", i);
|
|
}
|
|
}
|
|
}
|
|
else
|
|
info.loopstart = -1;
|
|
|
|
// find data chunk
|
|
chunklen = FindChunk(&ctx, "data");
|
|
if (!ctx.data_p)
|
|
{
|
|
Con_Printf("Missing data chunk in %s\n", name);
|
|
return info;
|
|
}
|
|
|
|
ctx.data_p += 8;
|
|
samples = chunklen / info.width /info.numchannels;
|
|
|
|
if (info.samples)
|
|
{
|
|
if (samples < info.samples)
|
|
{
|
|
info.samples = samples;
|
|
Con_Printf ("Sound %s has a bad loop length\n", name);
|
|
}
|
|
}
|
|
else
|
|
info.samples = samples;
|
|
|
|
if (info.loopstart > info.samples)
|
|
{
|
|
Con_Printf ("Sound %s has a bad loop start\n", name);
|
|
info.loopstart = info.samples;
|
|
}
|
|
|
|
info.dataofs = ctx.data_p - wav;
|
|
|
|
return info;
|
|
}
|