6d36834f8e
added waterfog command. waterfog overrides regular fog only when the view is in water. fixed 64bit printf format specifiers. should work better on winxp64. fixed some spec angle weirdness. fixed viewsize 99.99 weirdness with ezhud. fixed extra offset on the console (exhibited in 64bit builds, but not limited to). fixed .avi playback, can now actually display frames again. reimplemented line sparks. fixed r_editlights_save flipping the light's pitch. fixed issue with oggs failing to load. fixed condump to cope with unicode properly. made sv_bigcoords default except in quake. hexen2 kinda needs it for bsp angle precision. fixed nq server to not stall weirdly on map changes. fixed qwprogs svc_cdtrack not bugging out with nq clients on the server. fixed restart command to load the last map run by the server, instead of start.bsp (when idle) optimised d3d9 renderer a little. now uses less draw calls, especially with complex scenes. seems to get higher framerates than opengl now. fixed d3d9 renderer to not bug out quite so much when run fullscreen (shader subsystem is now correctly initialised). fixed a couple of bugs from font change. also now supports utf-8 in a few more places. r_editlights_reload no longer generates rtlights inside the void. this resolves a few glitches (but should also help framerates a little). fixed so corona-only lights won't generate shadowmaps and waste lots of time. removed lots of #defines from qclib. I should never have made them in the first place, but I was lazy. obviously there's more left that I cba to remove yet. fixed nested calls with variant-vectors. this fixes csaddon's light editor. fixed qcc hc calling conventions using redundant stores. disabled keywords can still be used by using __keyword instead. fixed ftegccgui grep feature. fixed motionless-dog qcc bug. tweaked qcc warnings a little. -Wall is now a viable setting. you should be able to fix all those warnings. fixed qw svc_intermission + dpp5+ clients bug. fixed annoying spam about disconnecting in hexen2. rewrote status command a little to cope with ipv6 addresses more gracefully fixed significant stall when hibernating/debugging a server with a player sitting on it. fixed truelightning. fixed rocketlight overriding pflags. fixed torches vanishing on vid_restart. fixed issue with decal scaling. fixed findentityfield builtin. fixed fteqcc issue with ptr+1 fixed use of arrays inside class functions. fixed/implemented fteqcc emulation of pointer opcodes. added __inout keyword to fteqcc, so that it doesn't feel so horrendous. fixed sizeof(*foo) fixed *struct = struct; fixed recursive structs. fixed fteqcc warning report. fixed sdl2 controller support, hopefully. attempted to implement xinput, including per-player audio playback. slightly fixed relaxed attitude to mouse focus when running fullscreen. fixed weird warnings/errors with 'ent.arrayhead' terms. now generates sane errors. implemented bindmaps (for csqc). fixed crashing bug with eprint builtin. implemented subset of music_playlist_* functionality. significant changes to music playback. fixed some more dpcsqc compat. fixed binds menu. now displays and accepts modifiers. fixed issues with huge lightmaps. fixed protocol determinism with dp clients connecting to fte servers. the initial getchallenge request now inhibits vanilla nq connection requests. implemented support for 'dupe' userinfo key, allowing clients to request client->server packet duplication. should probably queue them tbh. implemented sv_saveentfile command. fixed resume after breaking inside a stepped-over function. fixed erroneous footer after debugging. (I wonder just how many things I broke with these fixes) git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@4946 fc73d0e0-1445-4013-8a0c-d673dee63da5
1144 lines
26 KiB
C
1144 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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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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}
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}
|
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else // 16-bit
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{
|
|
if (outwidth == 2)
|
|
{
|
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if (inrate < outrate) // upsample
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|
{
|
|
if (resampstyle)
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LINEARUPSCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
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else
|
|
STANDARDRESCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
|
|
}
|
|
else // downsample
|
|
{
|
|
if (resampstyle > 1)
|
|
LINEARDOWNSCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
|
|
else
|
|
STANDARDRESCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if (inrate == outrate) // quick convert
|
|
{
|
|
insamps *= 2;
|
|
QUICKCONVERT(in16, insamps, out8, 0, 8)
|
|
}
|
|
else if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
|
LINEARUPSCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
else
|
|
STANDARDRESCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
}
|
|
else // downsample
|
|
{
|
|
if (resampstyle > 1)
|
|
LINEARDOWNSCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
else
|
|
STANDARDRESCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
}
|
|
}
|
|
}
|
|
}
|
|
#if 0
|
|
else if (outchannels == 1 && inchannels == 2)
|
|
{
|
|
if (inwidth == 1)
|
|
{
|
|
if (outwidth == 1)
|
|
{
|
|
if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
|
LINEARUPSCALESTEREOTOMONO(in8, inrate, insamps, out8, outrate, 0, 0)
|
|
else
|
|
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out8, outrate, 0, 0)
|
|
}
|
|
else // downsample
|
|
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out8, outrate, 0, 0)
|
|
}
|
|
else
|
|
{
|
|
if (inrate == outrate) // quick convert
|
|
QUICKCONVERTSTEREOTOMONO(in8, insamps, out16, 8, 0)
|
|
else if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
|
LINEARUPSCALESTEREOTOMONO(in8, inrate, insamps, out16, outrate, 8, 0)
|
|
else
|
|
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out16, outrate, 8, 0)
|
|
}
|
|
else // downsample
|
|
STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out16, outrate, 8, 0)
|
|
}
|
|
}
|
|
else // 16-bit
|
|
{
|
|
if (outwidth == 2)
|
|
{
|
|
if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
|
LINEARUPSCALESTEREOTOMONO(in16, inrate, insamps, out16, outrate, 0, 0)
|
|
else
|
|
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out16, outrate, 0, 0)
|
|
}
|
|
else // downsample
|
|
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out16, outrate, 0, 0)
|
|
}
|
|
else
|
|
{
|
|
if (inrate == outrate) // quick convert
|
|
QUICKCONVERTSTEREOTOMONO(in16, insamps, out8, 0, 8)
|
|
else if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
|
LINEARUPSCALESTEREOTOMONO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
else
|
|
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
}
|
|
else // downsample
|
|
STANDARDRESCALESTEREOTOMONO(in16, inrate, insamps, out8, outrate, 0, 8)
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
}
|
|
|
|
/*
|
|
================
|
|
ResampleSfx
|
|
================
|
|
*/
|
|
qboolean ResampleSfx (sfx_t *sfx, int inrate, int inchannels, int inwidth, int insamps, int inloopstart, qbyte *data)
|
|
{
|
|
extern cvar_t snd_linearresample;
|
|
double scale;
|
|
sfxcache_t *sc;
|
|
int outsamps;
|
|
int len;
|
|
int outwidth;
|
|
|
|
scale = snd_speed / (double)inrate;
|
|
outsamps = insamps * scale;
|
|
if (loadas8bit.ival < 0)
|
|
outwidth = 2;
|
|
else if (loadas8bit.ival)
|
|
outwidth = 1;
|
|
else
|
|
outwidth = inwidth;
|
|
len = outsamps * outwidth * inchannels;
|
|
|
|
sfx->decoder.buf = sc = BZ_Malloc(len + sizeof(sfxcache_t));
|
|
if (!sc)
|
|
{
|
|
return false;
|
|
}
|
|
|
|
sc->numchannels = inchannels;
|
|
sc->width = outwidth;
|
|
sc->speed = snd_speed;
|
|
sc->length = outsamps;
|
|
sc->soundoffset = 0;
|
|
sc->data = (qbyte*)(sc+1);
|
|
if (inloopstart == -1)
|
|
sc->loopstart = inloopstart;
|
|
else
|
|
sc->loopstart = inloopstart * scale;
|
|
|
|
SND_ResampleStream (data,
|
|
inrate,
|
|
inwidth,
|
|
inchannels,
|
|
insamps,
|
|
sc->data,
|
|
sc->speed,
|
|
sc->width,
|
|
sc->numchannels,
|
|
snd_linearresample.ival);
|
|
|
|
return true;
|
|
}
|
|
|
|
//=============================================================================
|
|
#ifdef DOOMWADS
|
|
#define DSPK_RATE 140
|
|
#define DSPK_BASE 170.0
|
|
#define DSPK_EXP 0.0433
|
|
|
|
/*
|
|
sfxcache_t *S_LoadDoomSpeakerSound (sfx_t *s, qbyte *data, int datalen, int sndspeed)
|
|
{
|
|
sfxcache_t *sc;
|
|
|
|
// format data from Unofficial Doom Specs v1.6
|
|
unsigned short *dataus;
|
|
int samples, len, inrate, inaccum;
|
|
qbyte *outdata;
|
|
qbyte towrite;
|
|
double timeraccum, timerfreq;
|
|
|
|
if (datalen < 4)
|
|
return NULL;
|
|
|
|
dataus = (unsigned short*)data;
|
|
|
|
if (LittleShort(dataus[0]) != 0)
|
|
return NULL;
|
|
|
|
samples = LittleShort(dataus[1]);
|
|
|
|
data += 4;
|
|
datalen -= 4;
|
|
|
|
if (datalen != samples)
|
|
return NULL;
|
|
|
|
len = (int)((double)samples * (double)snd_speed / DSPK_RATE);
|
|
|
|
sc = Cache_Alloc (&s->cache, len + sizeof(sfxcache_t), s->name);
|
|
if (!sc)
|
|
{
|
|
return NULL;
|
|
}
|
|
|
|
sc->length = len;
|
|
sc->loopstart = -1;
|
|
sc->numchannels = 1;
|
|
sc->width = 1;
|
|
sc->speed = snd_speed;
|
|
|
|
timeraccum = 0;
|
|
outdata = sc->data;
|
|
towrite = 0x40;
|
|
inrate = (int)((double)snd_speed / DSPK_RATE);
|
|
inaccum = inrate;
|
|
if (*data)
|
|
timerfreq = DSPK_BASE * pow((double)2.0, DSPK_EXP * (*data));
|
|
else
|
|
timerfreq = 0;
|
|
|
|
while (len > 0)
|
|
{
|
|
timeraccum += timerfreq;
|
|
if (timeraccum > (float)snd_speed)
|
|
{
|
|
towrite ^= 0xFF; // swap speaker component
|
|
timeraccum -= (float)snd_speed;
|
|
}
|
|
|
|
inaccum--;
|
|
if (!inaccum)
|
|
{
|
|
data++;
|
|
if (*data)
|
|
timerfreq = DSPK_BASE * pow((double)2.0, DSPK_EXP * (*data));
|
|
inaccum = inrate;
|
|
}
|
|
*outdata = towrite;
|
|
outdata++;
|
|
len--;
|
|
}
|
|
|
|
return sc;
|
|
}
|
|
*/
|
|
qboolean S_LoadDoomSound (sfx_t *s, qbyte *data, int datalen, int sndspeed)
|
|
{
|
|
// format data from Unofficial Doom Specs v1.6
|
|
unsigned short *dataus;
|
|
int samples, rate;
|
|
|
|
if (datalen < 8)
|
|
return false;
|
|
|
|
dataus = (unsigned short*)data;
|
|
|
|
if (LittleShort(dataus[0]) != 3)
|
|
return false;
|
|
|
|
rate = LittleShort(dataus[1]);
|
|
samples = LittleShort(dataus[2]);
|
|
|
|
data += 8;
|
|
datalen -= 8;
|
|
|
|
if (datalen != samples)
|
|
return false;
|
|
|
|
COM_CharBias(data, datalen);
|
|
|
|
return ResampleSfx (s, rate, 1, 1, samples, -1, data);
|
|
}
|
|
#endif
|
|
|
|
qboolean S_LoadWavSound (sfx_t *s, qbyte *data, int datalen, int sndspeed)
|
|
{
|
|
wavinfo_t info;
|
|
|
|
if (datalen < 4 || strncmp(data, "RIFF", 4))
|
|
return false;
|
|
|
|
info = GetWavinfo (s->name, data, datalen);
|
|
if (info.numchannels < 1 || info.numchannels > 2)
|
|
{
|
|
s->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.
|
|
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.
|
|
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;
|
|
}
|
|
|
|
void S_LoadedOrFailed (void *ctx, void *ctxdata, size_t a, size_t b)
|
|
{
|
|
sfx_t *s = ctx;
|
|
s->loadstate = a;
|
|
}
|
|
/*
|
|
==============
|
|
S_LoadSound
|
|
==============
|
|
*/
|
|
|
|
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 "fPRIzu", result was "fPRIzu"\n", name, filesize, result);
|
|
|
|
VFS_CLOSE(f);
|
|
}
|
|
else
|
|
{
|
|
Con_SafePrintf ("Couldn't load %s\n", namebuffer);
|
|
COM_AddWork(0, 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(0, 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))
|
|
{
|
|
s->loadstate = SLS_LOADED;
|
|
//wake up the main thread in case it decided to wait for us.
|
|
COM_AddWork(0, 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(0, 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(1, 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;
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
}
|
|
|
|
unsigned int FindChunk(wavctx_t *ctx, char *name)
|
|
{
|
|
ctx->last_chunk = ctx->iff_data;
|
|
return FindNextChunk (ctx, name);
|
|
}
|
|
|
|
|
|
#if 0
|
|
void DumpChunks(void)
|
|
{
|
|
char str[5];
|
|
|
|
str[4] = 0;
|
|
data_p=iff_data;
|
|
do
|
|
{
|
|
memcpy (str, data_p, 4);
|
|
data_p += 4;
|
|
iff_chunk_len = GetLittleLong();
|
|
Con_Printf ("0x%x : %s (%d)\n", (int)(data_p - 4), str, iff_chunk_len);
|
|
data_p += (iff_chunk_len + 1) & ~1;
|
|
} while (data_p < iff_end);
|
|
}
|
|
#endif
|
|
|
|
/*
|
|
============
|
|
GetWavinfo
|
|
============
|
|
*/
|
|
wavinfo_t GetWavinfo (char *name, qbyte *wav, int wavlength)
|
|
{
|
|
wavinfo_t info;
|
|
int i;
|
|
int format;
|
|
int samples;
|
|
int chunklen;
|
|
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;
|
|
}
|