mirror of
https://github.com/nzp-team/fteqw.git
synced 2024-11-11 15:12:29 +00:00
484e8bbfc2
reworked network addresses to separate address family and connection type. this should make banning people more reliable, as well as simplifying a whole load of logic (no need to check for ipv4 AND ipv6).
tcpconnect will keep trying to connect even if the connection wasn't instant, instead of giving up instantly.
rewrote tcp connections quite a bit. sv_port_tcp now handles qtv+qizmo+http+ws+rtcbroker+tls equivalents.
qtv_streamport is now a legacy cvar and now acts equivalently to sv_port_tcp (but still separate).
rewrote screenshot and video capture code to use strides. this solves image-is-upside down issues with vulkan.
ignore alt key in browser port. oh no! no more red text! oh no! no more alt-being-wrongly-down-and-being-unable-to-type-anything-without-forcing-alt-released!
reworked audio decoder interface. now has clearly defined success/unavailable/end-of-file results. this should solve a whole load of issues with audio streaming.
fixed various openal audio streaming issues too. openal also got some workarounds for emscripten's poor emulation.
fixed ogg decoder to retain sync properly if seeked.
updated menu_media a bit. now reads vorbis comments/id3v1 tags to get proper track names. also saves the playlist so you don't have to manually repopulate the list so it might actually be usable now (after how many years?)
r_stains now defaults to 0, and is no longer enabled by presets. use decals if you want that sort of thing.
added fs_noreexec cvar, so configs will not be reexeced on gamedir change. this also means defaults won't be reapplied, etc.
added 'nvvk' renderer on windows, using nvidia's vulkan-inside-opengl gl extension. mostly just to see how much slower it is.
fixed up the ftp server quite a lot. more complete, more compliant, and should do ipv6 properly to-boot. file transfers also threaded.
fixed potential crash inside runclientphys.
experimental sv_antilag=3 setting. totally untested. the aim is to avoid missing due to lagged knockbacks. may be expensive for the server.
browser port's websockets support fixed. experimental support for webrtc ('works for me', requires a broker server).
updated avplug(renamed to ffmpeg so people know what it is) to use ffmpeg 3.2.4 properly, with its new encoder api. should be much more robust... also added experimental audio decoder for game music etc (currently doesn't resample, so playback rates are screwed, disabled by cvar).
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@5097 fc73d0e0-1445-4013-8a0c-d673dee63da5
1157 lines
27 KiB
C
1157 lines
27 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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}
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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
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STANDARDRESCALESTEREO(in16, inrate, insamps, out16, outrate, 0, 0)
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}
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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
|
|
================
|
|
*/
|
|
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)
|
|
sfx->loopstart = inloopstart;
|
|
else
|
|
sfx->loopstart = inloopstart * scale;
|
|
|
|
SND_ResampleStream (data,
|
|
inrate,
|
|
inwidth,
|
|
inchannels,
|
|
insamps,
|
|
sc->data,
|
|
sc->speed,
|
|
sc->width,
|
|
sc->numchannels,
|
|
snd_linearresample.ival);
|
|
|
|
return true;
|
|
}
|
|
|
|
//=============================================================================
|
|
#ifdef PACKAGE_DOOMWAD
|
|
#define DSPK_RATE 140
|
|
#define DSPK_BASE 170.0
|
|
#define DSPK_EXP 0.0433
|
|
|
|
/*
|
|
qboolean QDECL S_LoadDoomSpeakerSound (sfx_t *s, qbyte *data, size_t datalen, int sndspeed)
|
|
{
|
|
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 QDECL S_LoadDoomSound (sfx_t *s, qbyte *data, size_t 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 QDECL S_LoadWavSound (sfx_t *s, qbyte *data, size_t 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 QDECL S_LoadOVSound (sfx_t *s, qbyte *data, size_t datalen, int sndspeed);
|
|
|
|
#ifdef FTE_TARGET_WEB
|
|
//web browsers contain their own decoding libraries that our openal stuff can use.
|
|
static qboolean QDECL S_LoadBrowserFile (sfx_t *s, qbyte *data, size_t datalen, int sndspeed)
|
|
{
|
|
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 PACKAGE_DOOMWAD
|
|
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;
|
|
|
|
s->loopstart = -1;
|
|
|
|
if (s->syspath)
|
|
{
|
|
vfsfile_t *f;
|
|
|
|
if ((f = VFSOS_Open(name, "rb")))
|
|
{
|
|
filesize = VFS_GETLEN(f);
|
|
data = BZ_Malloc (filesize);
|
|
result = VFS_READ(f, data, filesize);
|
|
|
|
if (result != filesize)
|
|
Con_SafePrintf("S_LoadSound() fread: Filename: %s, expected %"PRIuSIZE", result was %"PRIuSIZE"\n", name, filesize, result);
|
|
|
|
VFS_CLOSE(f);
|
|
}
|
|
else
|
|
{
|
|
Con_SafePrintf ("Couldn't load %s\n", namebuffer);
|
|
COM_AddWork(WG_MAIN, S_LoadedOrFailed, s, NULL, SLS_FAILED, 0);
|
|
return;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
//Con_Printf ("S_LoadSound: %x\n", (int)stackbuf);
|
|
// load it in
|
|
const char *prefixes[] = {"sound/", ""};
|
|
const char *extensions[] = {".wav", ".ogg"};
|
|
char altname[sizeof(namebuffer)];
|
|
char orig[16];
|
|
size_t pre, ex;
|
|
|
|
data = NULL;
|
|
filesize = 0;
|
|
if (*name == '*') //q2 sexed sounds
|
|
{
|
|
//clq2_parsestartsound detects this also, and should not try playing these sounds.
|
|
s->loadstate = SLS_FAILED;
|
|
return;
|
|
}
|
|
|
|
for (pre = 0; !data && pre < countof(prefixes); pre++)
|
|
{
|
|
if (name[0] == '.' && name[1] == '.' && name[2] == '/')
|
|
{ //someone's being specific. disable prefixes entirely.
|
|
if (pre)
|
|
break;
|
|
//not relative to sound/
|
|
Q_snprintfz(namebuffer, sizeof(namebuffer), "%s", name+3);
|
|
}
|
|
else
|
|
Q_snprintfz(namebuffer, sizeof(namebuffer), "%s%s", prefixes[pre], name);
|
|
|
|
data = COM_LoadFile(namebuffer, 5, &filesize);
|
|
if (data)
|
|
break;
|
|
COM_FileExtension(namebuffer, orig, sizeof(orig));
|
|
COM_StripExtension(namebuffer, altname, sizeof(altname));
|
|
for (ex = 0; ex < countof(extensions); ex++)
|
|
{
|
|
if (!strcmp(orig, extensions[ex]+1))
|
|
continue;
|
|
Q_snprintfz(namebuffer, sizeof(namebuffer), "%s%s", altname, extensions[ex]);
|
|
data = COM_LoadFile(namebuffer, 5, &filesize);
|
|
if (data)
|
|
{
|
|
Con_DPrintf("found a mangled name: %s\n", namebuffer);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (!data)
|
|
{
|
|
//FIXME: check to see if queued for download.
|
|
if (name[0] == '.' && name[1] == '.' && name[2] == '/')
|
|
Con_DPrintf ("Couldn't load %s\n", name+3);
|
|
else
|
|
Con_DPrintf ("Couldn't load sound/%s\n", name);
|
|
COM_AddWork(WG_MAIN, S_LoadedOrFailed, s, NULL, SLS_FAILED, 0);
|
|
return;
|
|
}
|
|
|
|
for (i = sizeof(AudioInputPlugins)/sizeof(AudioInputPlugins[0])-1; i >= 0; i--)
|
|
{
|
|
if (AudioInputPlugins[i])
|
|
{
|
|
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;
|
|
}
|