fb95c21dcd
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@3337 fc73d0e0-1445-4013-8a0c-d673dee63da5
1123 lines
25 KiB
C
1123 lines
25 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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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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{
|
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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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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
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{
|
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if (resampstyle > 1)
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LINEARDOWNSCALESTEREO(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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}
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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(in16, insamps, out8, 0, 8)
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}
|
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else if (inrate < outrate) // upsample
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{
|
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if (resampstyle)
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LINEARUPSCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
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else
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STANDARDRESCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
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}
|
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else // downsample
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{
|
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if (resampstyle > 1)
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LINEARDOWNSCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
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else
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STANDARDRESCALESTEREO(in16, inrate, insamps, out8, outrate, 0, 8)
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}
|
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}
|
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}
|
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}
|
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#if 0
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else if (outchannels == 1 && inchannels == 2)
|
|
{
|
|
if (inwidth == 1)
|
|
{
|
|
if (outwidth == 1)
|
|
{
|
|
if (inrate < outrate) // upsample
|
|
{
|
|
if (resampstyle)
|
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LINEARUPSCALESTEREOTOMONO(in8, inrate, insamps, out8, outrate, 0, 0)
|
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else
|
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STANDARDRESCALESTEREOTOMONO(in8, inrate, insamps, out8, outrate, 0, 0)
|
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}
|
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else // downsample
|
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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
|
|
================
|
|
*/
|
|
void ResampleSfx (sfx_t *sfx, int inrate, int inwidth, qbyte *data)
|
|
{
|
|
extern cvar_t snd_linearresample;
|
|
double scale;
|
|
sfxcache_t *sc;
|
|
int insamps, outsamps;
|
|
|
|
sc = Cache_Check (&sfx->cache);
|
|
if (!sc)
|
|
return;
|
|
|
|
insamps = sc->length;
|
|
scale = snd_speed / (double)inrate;
|
|
outsamps = insamps * scale;
|
|
sc->length = outsamps;
|
|
if (sc->loopstart != -1)
|
|
sc->loopstart = sc->loopstart * scale;
|
|
|
|
sc->speed = snd_speed;
|
|
if (loadas8bit.value)
|
|
sc->width = 1;
|
|
else
|
|
sc->width = inwidth;
|
|
|
|
SND_ResampleStream (data,
|
|
inrate,
|
|
inwidth,
|
|
sc->numchannels,
|
|
insamps,
|
|
sc->data,
|
|
sc->speed,
|
|
sc->width,
|
|
sc->numchannels,
|
|
(int)snd_linearresample.value);
|
|
}
|
|
|
|
//=============================================================================
|
|
#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));
|
|
|
|
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;
|
|
}
|
|
|
|
sfxcache_t *S_LoadDoomSound (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, rate, len;
|
|
|
|
if (datalen < 8)
|
|
return NULL;
|
|
|
|
dataus = (unsigned short*)data;
|
|
|
|
if (LittleShort(dataus[0]) != 3)
|
|
return NULL;
|
|
|
|
rate = LittleShort(dataus[1]);
|
|
samples = LittleShort(dataus[2]);
|
|
|
|
data += 8;
|
|
datalen -= 8;
|
|
|
|
if (datalen != samples)
|
|
return NULL;
|
|
|
|
len = (int)((double)samples * (double)snd_speed / (double)rate);
|
|
|
|
sc = Cache_Alloc (&s->cache, len + sizeof(sfxcache_t), s->name);
|
|
if (!sc)
|
|
{
|
|
return NULL;
|
|
}
|
|
|
|
sc->length = samples;
|
|
sc->loopstart = -1;
|
|
sc->numchannels = 1;
|
|
sc->width = 1;
|
|
sc->speed = rate;
|
|
|
|
if (sc->width == 1)
|
|
COM_CharBias(data, sc->length);
|
|
else if (sc->width == 2)
|
|
COM_SwapLittleShortBlock((short *)data, sc->length);
|
|
|
|
ResampleSfx (s, sc->speed, sc->width, data);
|
|
|
|
return sc;
|
|
}
|
|
#endif
|
|
|
|
sfxcache_t *S_LoadWavSound (sfx_t *s, qbyte *data, int datalen, int sndspeed)
|
|
{
|
|
wavinfo_t info;
|
|
int len;
|
|
sfxcache_t *sc;
|
|
|
|
if (datalen < 4 || strncmp(data, "RIFF", 4))
|
|
return NULL;
|
|
|
|
info = GetWavinfo (s->name, data, datalen);
|
|
if (info.numchannels < 1 || info.numchannels > 2)
|
|
{
|
|
s->failedload = true;
|
|
Con_Printf ("%s has an unsupported quantity of channels.\n",s->name);
|
|
return NULL;
|
|
}
|
|
|
|
len = (int) ((double) info.samples * (double) snd_speed / (double) info.rate);
|
|
len = len * info.width * info.numchannels;
|
|
|
|
sc = Cache_Alloc ( &s->cache, len + sizeof(sfxcache_t), s->name);
|
|
if (!sc)
|
|
{
|
|
return NULL;
|
|
}
|
|
|
|
sc->length = info.samples;
|
|
sc->loopstart = info.loopstart;
|
|
sc->speed = info.rate;
|
|
sc->width = info.width;
|
|
sc->numchannels = info.numchannels;
|
|
|
|
if (sc->width == 1)
|
|
COM_CharBias(data + info.dataofs, sc->length*sc->numchannels);
|
|
else if (sc->width == 2)
|
|
COM_SwapLittleShortBlock((short *)(data + info.dataofs), sc->length*sc->numchannels);
|
|
|
|
ResampleSfx (s, sc->speed, sc->width, data + info.dataofs);
|
|
|
|
return sc;
|
|
}
|
|
|
|
sfxcache_t *S_LoadOVSound (sfx_t *s, qbyte *data, int datalen, int sndspeed);
|
|
|
|
S_LoadSound_t AudioInputPlugins[10] =
|
|
{
|
|
#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;
|
|
}
|
|
|
|
/*
|
|
==============
|
|
S_LoadSound
|
|
==============
|
|
*/
|
|
|
|
sfxcache_t *S_LoadSound (sfx_t *s)
|
|
{
|
|
char stackbuf[65536];
|
|
char namebuffer[256];
|
|
qbyte *data;
|
|
sfxcache_t *sc;
|
|
int i;
|
|
|
|
char *name = s->name;
|
|
|
|
if (s->failedload)
|
|
return NULL; //it failed to load once before, don't bother trying again.
|
|
|
|
// see if still in memory
|
|
sc = Cache_Check (&s->cache);
|
|
if (sc)
|
|
return sc;
|
|
|
|
s->decoder = NULL;
|
|
|
|
|
|
|
|
|
|
if (name[1] == ':' && name[2] == '\\')
|
|
{
|
|
FILE *f;
|
|
#ifndef _WIN32 //convert from windows to a suitable alternative.
|
|
char unixname[128];
|
|
sprintf(unixname, "/mnt/%c/%s", name[0]-'A'+'a', name+3);
|
|
name = unixname;
|
|
while (*name)
|
|
{
|
|
if (*name == '\\')
|
|
*name = '/';
|
|
name++;
|
|
}
|
|
name = unixname;
|
|
#endif
|
|
|
|
if ((f = fopen(name, "rb")))
|
|
{
|
|
com_filesize = COM_filelength(f);
|
|
data = Hunk_TempAlloc (com_filesize);
|
|
fread(data, 1, com_filesize, f);
|
|
fclose(f);
|
|
}
|
|
else
|
|
{
|
|
Con_SafePrintf ("Couldn't load %s\n", namebuffer);
|
|
return NULL;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
//Con_Printf ("S_LoadSound: %x\n", (int)stackbuf);
|
|
// load it in
|
|
|
|
data = NULL;
|
|
if (*name == '*') //q2 sexed sounds
|
|
{
|
|
//clq2_parsestartsound detects this also
|
|
//here we just precache the male sound name, which provides us with our default
|
|
Q_strcpy(namebuffer, "players/male/"); //q2
|
|
Q_strcat(namebuffer, name+1); //q2
|
|
}
|
|
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_LoadStackFile(name, stackbuf, sizeof(stackbuf));
|
|
}
|
|
|
|
// Con_Printf ("loading %s\n",namebuffer);
|
|
|
|
if (!data)
|
|
data = COM_LoadStackFile(namebuffer, stackbuf, sizeof(stackbuf));
|
|
if (!data)
|
|
{
|
|
char altname[sizeof(namebuffer)];
|
|
COM_StripExtension(namebuffer, altname, sizeof(altname));
|
|
COM_DefaultExtension(altname, ".ogg", sizeof(altname));
|
|
data = COM_LoadStackFile(altname, stackbuf, sizeof(stackbuf));
|
|
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);
|
|
s->failedload = true;
|
|
return NULL;
|
|
}
|
|
|
|
s->failedload = false;
|
|
|
|
for (i = sizeof(AudioInputPlugins)/sizeof(AudioInputPlugins[0])-1; i >= 0; i--)
|
|
{
|
|
if (AudioInputPlugins[i])
|
|
{
|
|
sc = AudioInputPlugins[i](s, data, com_filesize, snd_speed);
|
|
if (sc)
|
|
return sc;
|
|
}
|
|
}
|
|
|
|
if (!s->failedload)
|
|
Con_Printf ("Format not recognised: %s\n", namebuffer);
|
|
|
|
s->failedload = true;
|
|
return NULL;
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
===============================================================================
|
|
|
|
WAV loading
|
|
|
|
===============================================================================
|
|
*/
|
|
|
|
|
|
qbyte *data_p;
|
|
qbyte *iff_end;
|
|
qbyte *last_chunk;
|
|
qbyte *iff_data;
|
|
int iff_chunk_len;
|
|
|
|
|
|
short GetLittleShort(void)
|
|
{
|
|
short val = 0;
|
|
val = *data_p;
|
|
val = val + (*(data_p+1)<<8);
|
|
data_p += 2;
|
|
return val;
|
|
}
|
|
|
|
int GetLittleLong(void)
|
|
{
|
|
int val = 0;
|
|
val = *data_p;
|
|
val = val + (*(data_p+1)<<8);
|
|
val = val + (*(data_p+2)<<16);
|
|
val = val + (*(data_p+3)<<24);
|
|
data_p += 4;
|
|
return val;
|
|
}
|
|
|
|
unsigned int FindNextChunk(char *name)
|
|
{
|
|
unsigned int dataleft;
|
|
|
|
while (1)
|
|
{
|
|
dataleft = iff_end - last_chunk;
|
|
if (dataleft < 8)
|
|
{ // didn't find the chunk
|
|
data_p = NULL;
|
|
return 0;
|
|
}
|
|
|
|
data_p=last_chunk;
|
|
data_p += 4;
|
|
dataleft-= 8;
|
|
iff_chunk_len = GetLittleLong();
|
|
if (iff_chunk_len < 0)
|
|
{
|
|
data_p = NULL;
|
|
return 0;
|
|
}
|
|
if (iff_chunk_len > dataleft)
|
|
{
|
|
Con_Printf ("Sound file seems truncated by %i bytes\n", iff_chunk_len-dataleft);
|
|
#if 1
|
|
iff_chunk_len = dataleft;
|
|
#else
|
|
data_p = NULL;
|
|
return 0;
|
|
#endif
|
|
}
|
|
|
|
dataleft-= iff_chunk_len;
|
|
// if (iff_chunk_len > 1024*1024)
|
|
// Sys_Error ("FindNextChunk: %i length is past the 1 meg sanity limit", iff_chunk_len);
|
|
data_p -= 8;
|
|
last_chunk = data_p + 8 + iff_chunk_len;
|
|
if ((iff_chunk_len&1) && dataleft)
|
|
last_chunk++;
|
|
if (!Q_strncmp(data_p, name, 4))
|
|
return iff_chunk_len;
|
|
}
|
|
}
|
|
|
|
unsigned int FindChunk(char *name)
|
|
{
|
|
last_chunk = iff_data;
|
|
return FindNextChunk (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;
|
|
|
|
memset (&info, 0, sizeof(info));
|
|
|
|
if (!wav)
|
|
return info;
|
|
|
|
iff_data = wav;
|
|
iff_end = wav + wavlength;
|
|
|
|
// find "RIFF" chunk
|
|
chunklen = FindChunk("RIFF");
|
|
if (chunklen < 4 || Q_strncmp(data_p+8, "WAVE", 4))
|
|
{
|
|
Con_Printf("Missing RIFF/WAVE chunks in %s\n", name);
|
|
return info;
|
|
}
|
|
|
|
// get "fmt " chunk
|
|
iff_data = data_p + 12;
|
|
// DumpChunks ();
|
|
|
|
chunklen = FindChunk("fmt ");
|
|
if (chunklen < 24-8)
|
|
{
|
|
Con_Printf("Missing/truncated fmt chunk\n");
|
|
return info;
|
|
}
|
|
data_p += 8;
|
|
format = GetLittleShort();
|
|
if (format != 1)
|
|
{
|
|
Con_Printf("Microsoft PCM format only\n");
|
|
return info;
|
|
}
|
|
|
|
info.numchannels = GetLittleShort();
|
|
info.rate = GetLittleLong();
|
|
data_p += 4+2;
|
|
info.width = GetLittleShort() / 8;
|
|
|
|
// get cue chunk
|
|
chunklen = FindChunk("cue ");
|
|
if (chunklen >= 36-8)
|
|
{
|
|
data_p += 32;
|
|
info.loopstart = GetLittleLong();
|
|
// Con_Printf("loopstart=%d\n", sfx->loopstart);
|
|
|
|
// if the next chunk is a LIST chunk, look for a cue length marker
|
|
chunklen = FindNextChunk ("LIST");
|
|
if (chunklen >= 32-8)
|
|
{
|
|
if (!strncmp (data_p + 28, "mark", 4))
|
|
{ // this is not a proper parse, but it works with cooledit...
|
|
data_p += 24;
|
|
i = GetLittleLong (); // samples in loop
|
|
info.samples = info.loopstart + i;
|
|
// Con_Printf("looped length: %i\n", i);
|
|
}
|
|
}
|
|
}
|
|
else
|
|
info.loopstart = -1;
|
|
|
|
// find data chunk
|
|
chunklen = FindChunk("data");
|
|
if (!chunklen)
|
|
{
|
|
Con_Printf("Missing data chunk in %s\n", name);
|
|
return info;
|
|
}
|
|
|
|
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 = data_p - wav;
|
|
|
|
return info;
|
|
}
|