1546 lines
59 KiB
C
1546 lines
59 KiB
C
/**
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* OpenAL cross platform audio library
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* Copyright (C) 1999-2007 by authors.
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library 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. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#define _CRT_SECURE_NO_DEPRECATE // get rid of sprintf security warnings on VS2005
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#include "config.h"
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#include <math.h>
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#include <stdlib.h>
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#include <string.h>
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#include <ctype.h>
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#include <assert.h>
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#include "alMain.h"
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#include "AL/al.h"
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#include "AL/alc.h"
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#include "alSource.h"
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#include "alBuffer.h"
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#include "alThunk.h"
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#include "alListener.h"
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#include "alAuxEffectSlot.h"
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#include "alu.h"
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#include "bs2b.h"
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#include "alReverb.h"
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#if defined (HAVE_FLOAT_H)
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#include <float.h>
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#endif
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#ifndef M_PI
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#define M_PI 3.14159265358979323846 /* pi */
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#define M_PI_2 1.57079632679489661923 /* pi/2 */
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#endif
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#if defined(HAVE_STDINT_H)
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#include <stdint.h>
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typedef int64_t ALint64;
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#elif defined(HAVE___INT64)
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typedef __int64 ALint64;
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#elif (SIZEOF_LONG == 8)
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typedef long ALint64;
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#elif (SIZEOF_LONG_LONG == 8)
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typedef long long ALint64;
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#endif
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#ifdef HAVE_SQRTF
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#define aluSqrt(x) ((ALfloat)sqrtf((float)(x)))
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#else
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#define aluSqrt(x) ((ALfloat)sqrt((double)(x)))
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#endif
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#ifdef HAVE_ACOSF
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#define aluAcos(x) ((ALfloat)acosf((float)(x)))
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#else
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#define aluAcos(x) ((ALfloat)acos((double)(x)))
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#endif
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#ifdef HAVE_ATANF
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#define aluAtan(x) ((ALfloat)atanf((float)(x)))
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#else
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#define aluAtan(x) ((ALfloat)atan((double)(x)))
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#endif
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#ifdef HAVE_FABSF
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#define aluFabs(x) ((ALfloat)fabsf((float)(x)))
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#else
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#define aluFabs(x) ((ALfloat)fabs((double)(x)))
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#endif
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// fixes for mingw32.
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#if defined(max) && !defined(__max)
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#define __max max
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#endif
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#if defined(min) && !defined(__min)
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#define __min min
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#endif
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#define BUFFERSIZE 24000
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#define FRACTIONBITS 14
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#define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
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#define MAX_PITCH 65536
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/* Minimum ramp length in milliseconds. The value below was chosen to
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* adequately reduce clicks and pops from harsh gain changes. */
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#define MIN_RAMP_LENGTH 16
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ALboolean DuplicateStereo = AL_FALSE;
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/* NOTE: The AL_FORMAT_REAR* enums aren't handled here be cause they're
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* converted to AL_FORMAT_QUAD* when loaded */
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__inline ALuint aluBytesFromFormat(ALenum format)
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{
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switch(format)
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{
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case AL_FORMAT_MONO8:
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case AL_FORMAT_STEREO8:
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case AL_FORMAT_QUAD8_LOKI:
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case AL_FORMAT_QUAD8:
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case AL_FORMAT_51CHN8:
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case AL_FORMAT_61CHN8:
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case AL_FORMAT_71CHN8:
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return 1;
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case AL_FORMAT_MONO16:
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case AL_FORMAT_STEREO16:
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case AL_FORMAT_QUAD16_LOKI:
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case AL_FORMAT_QUAD16:
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case AL_FORMAT_51CHN16:
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case AL_FORMAT_61CHN16:
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case AL_FORMAT_71CHN16:
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return 2;
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case AL_FORMAT_MONO_FLOAT32:
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case AL_FORMAT_STEREO_FLOAT32:
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case AL_FORMAT_QUAD32:
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case AL_FORMAT_51CHN32:
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case AL_FORMAT_61CHN32:
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case AL_FORMAT_71CHN32:
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return 4;
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default:
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return 0;
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}
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}
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__inline ALuint aluChannelsFromFormat(ALenum format)
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{
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switch(format)
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{
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case AL_FORMAT_MONO8:
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case AL_FORMAT_MONO16:
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case AL_FORMAT_MONO_FLOAT32:
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return 1;
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case AL_FORMAT_STEREO8:
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case AL_FORMAT_STEREO16:
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case AL_FORMAT_STEREO_FLOAT32:
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return 2;
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case AL_FORMAT_QUAD8_LOKI:
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case AL_FORMAT_QUAD16_LOKI:
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case AL_FORMAT_QUAD8:
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case AL_FORMAT_QUAD16:
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case AL_FORMAT_QUAD32:
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return 4;
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case AL_FORMAT_51CHN8:
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case AL_FORMAT_51CHN16:
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case AL_FORMAT_51CHN32:
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return 6;
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case AL_FORMAT_61CHN8:
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case AL_FORMAT_61CHN16:
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case AL_FORMAT_61CHN32:
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return 7;
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case AL_FORMAT_71CHN8:
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case AL_FORMAT_71CHN16:
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case AL_FORMAT_71CHN32:
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return 8;
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default:
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return 0;
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}
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}
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static __inline ALfloat lpFilter(FILTER *iir, ALfloat input)
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{
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ALfloat *history = iir->history;
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ALfloat a = iir->coeff;
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ALfloat output = input;
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output = output + (history[0]-output)*a;
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history[0] = output;
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output = output + (history[1]-output)*a;
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history[1] = output;
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output = output + (history[2]-output)*a;
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history[2] = output;
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output = output + (history[3]-output)*a;
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history[3] = output;
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return output;
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}
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static __inline ALfloat lpFilterMC(FILTER *iir, ALuint chan, ALfloat input)
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{
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ALfloat *history = &iir->history[chan*2];
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ALfloat a = iir->coeff;
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ALfloat output = input;
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output = output + (history[0]-output)*a;
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history[0] = output;
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output = output + (history[1]-output)*a;
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history[1] = output;
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return output;
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}
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static __inline ALshort aluF2S(ALfloat Value)
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{
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ALint i;
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i = (ALint)Value;
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i = __min( 32767, i);
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i = __max(-32768, i);
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return ((ALshort)i);
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}
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static __inline ALvoid aluCrossproduct(ALfloat *inVector1,ALfloat *inVector2,ALfloat *outVector)
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{
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outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
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outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
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outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
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}
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static __inline ALfloat aluDotproduct(ALfloat *inVector1,ALfloat *inVector2)
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{
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return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] +
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inVector1[2]*inVector2[2];
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}
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static __inline ALvoid aluNormalize(ALfloat *inVector)
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{
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ALfloat length, inverse_length;
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length = aluSqrt(aluDotproduct(inVector, inVector));
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if(length != 0.0f)
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{
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inverse_length = 1.0f/length;
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inVector[0] *= inverse_length;
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inVector[1] *= inverse_length;
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inVector[2] *= inverse_length;
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}
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}
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static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat matrix[3][3])
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{
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ALfloat result[3];
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result[0] = vector[0]*matrix[0][0] + vector[1]*matrix[1][0] + vector[2]*matrix[2][0];
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result[1] = vector[0]*matrix[0][1] + vector[1]*matrix[1][1] + vector[2]*matrix[2][1];
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result[2] = vector[0]*matrix[0][2] + vector[1]*matrix[1][2] + vector[2]*matrix[2][2];
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memcpy(vector, result, sizeof(result));
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}
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static ALvoid SetSpeakerArrangement(const char *name, ALfloat SpeakerAngle[OUTPUTCHANNELS],
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ALint Speaker2Chan[OUTPUTCHANNELS], ALint chans)
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{
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const char *confkey;
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const char *next;
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const char *sep;
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const char *end;
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int i, val;
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confkey = GetConfigValue(NULL, name, "");
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next = confkey;
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while(next && *next)
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{
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confkey = next;
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next = strchr(confkey, ',');
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if(next)
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{
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do {
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next++;
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} while(isspace(*next));
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}
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sep = strchr(confkey, '=');
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if(!sep || confkey == sep)
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continue;
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end = sep - 1;
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while(isspace(*end) && end != confkey)
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end--;
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if(strncmp(confkey, "fl", end-confkey) == 0)
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val = FRONT_LEFT;
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else if(strncmp(confkey, "fr", end-confkey) == 0)
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val = FRONT_RIGHT;
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else if(strncmp(confkey, "fc", end-confkey) == 0)
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val = FRONT_CENTER;
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else if(strncmp(confkey, "bl", end-confkey) == 0)
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val = BACK_LEFT;
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else if(strncmp(confkey, "br", end-confkey) == 0)
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val = BACK_RIGHT;
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else if(strncmp(confkey, "bc", end-confkey) == 0)
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val = BACK_CENTER;
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else if(strncmp(confkey, "sl", end-confkey) == 0)
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val = SIDE_LEFT;
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else if(strncmp(confkey, "sr", end-confkey) == 0)
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val = SIDE_RIGHT;
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else
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{
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AL_PRINT("Unknown speaker for %s: \"%c%c\"\n", name, confkey[0], confkey[1]);
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continue;
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}
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sep++;
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while(isspace(*sep))
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sep++;
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for(i = 0;i < chans;i++)
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{
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if(Speaker2Chan[i] == val)
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{
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val = strtol(sep, NULL, 10);
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if(val >= -180 && val <= 180)
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SpeakerAngle[i] = val * M_PI/180.0f;
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else
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AL_PRINT("Invalid angle for speaker \"%c%c\": %d\n", confkey[0], confkey[1], val);
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break;
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}
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}
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}
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for(i = 1;i < chans;i++)
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{
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if(SpeakerAngle[i] <= SpeakerAngle[i-1])
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{
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AL_PRINT("Speaker %d of %d does not follow previous: %f > %f\n", i, chans,
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SpeakerAngle[i-1] * 180.0f/M_PI, SpeakerAngle[i] * 180.0f/M_PI);
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SpeakerAngle[i] = SpeakerAngle[i-1] + 1 * 180.0f/M_PI;
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}
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}
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}
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static __inline ALfloat aluLUTpos2Angle(ALint pos)
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{
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if(pos < QUADRANT_NUM)
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return aluAtan((ALfloat)pos / (ALfloat)(QUADRANT_NUM - pos));
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if(pos < 2 * QUADRANT_NUM)
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return M_PI_2 + aluAtan((ALfloat)(pos - QUADRANT_NUM) / (ALfloat)(2 * QUADRANT_NUM - pos));
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if(pos < 3 * QUADRANT_NUM)
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return aluAtan((ALfloat)(pos - 2 * QUADRANT_NUM) / (ALfloat)(3 * QUADRANT_NUM - pos)) - M_PI;
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return aluAtan((ALfloat)(pos - 3 * QUADRANT_NUM) / (ALfloat)(4 * QUADRANT_NUM - pos)) - M_PI_2;
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}
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ALvoid aluInitPanning(ALCcontext *Context)
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{
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ALint pos, offset, s;
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ALfloat Alpha, Theta;
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ALfloat SpeakerAngle[OUTPUTCHANNELS];
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ALint Speaker2Chan[OUTPUTCHANNELS];
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for(s = 0;s < OUTPUTCHANNELS;s++)
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{
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int s2;
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for(s2 = 0;s2 < OUTPUTCHANNELS;s2++)
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Context->ChannelMatrix[s][s2] = ((s==s2) ? 1.0f : 0.0f);
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}
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switch(Context->Device->Format)
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{
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/* Mono is rendered as stereo, then downmixed during post-process */
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case AL_FORMAT_MONO8:
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case AL_FORMAT_MONO16:
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case AL_FORMAT_MONO_FLOAT32:
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Context->ChannelMatrix[FRONT_CENTER][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = 1.0f;
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Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = 1.0f;
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Context->ChannelMatrix[BACK_LEFT][FRONT_LEFT] = 1.0f;
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Context->ChannelMatrix[BACK_RIGHT][FRONT_RIGHT] = 1.0f;
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Context->ChannelMatrix[BACK_CENTER][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
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Context->NumChan = 2;
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Speaker2Chan[0] = FRONT_LEFT;
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Speaker2Chan[1] = FRONT_RIGHT;
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SpeakerAngle[0] = -90.0f * M_PI/180.0f;
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SpeakerAngle[1] = 90.0f * M_PI/180.0f;
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break;
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case AL_FORMAT_STEREO8:
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case AL_FORMAT_STEREO16:
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case AL_FORMAT_STEREO_FLOAT32:
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Context->ChannelMatrix[FRONT_CENTER][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = 1.0f;
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Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = 1.0f;
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Context->ChannelMatrix[BACK_LEFT][FRONT_LEFT] = 1.0f;
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Context->ChannelMatrix[BACK_RIGHT][FRONT_RIGHT] = 1.0f;
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Context->ChannelMatrix[BACK_CENTER][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
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Context->NumChan = 2;
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Speaker2Chan[0] = FRONT_LEFT;
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Speaker2Chan[1] = FRONT_RIGHT;
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SpeakerAngle[0] = -90.0f * M_PI/180.0f;
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SpeakerAngle[1] = 90.0f * M_PI/180.0f;
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SetSpeakerArrangement("layout_STEREO", SpeakerAngle, Speaker2Chan, Context->NumChan);
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break;
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case AL_FORMAT_QUAD8:
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case AL_FORMAT_QUAD16:
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case AL_FORMAT_QUAD32:
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Context->ChannelMatrix[FRONT_CENTER][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_LEFT][BACK_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_RIGHT][BACK_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
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Context->NumChan = 4;
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Speaker2Chan[0] = BACK_LEFT;
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Speaker2Chan[1] = FRONT_LEFT;
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Speaker2Chan[2] = FRONT_RIGHT;
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Speaker2Chan[3] = BACK_RIGHT;
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SpeakerAngle[0] = -135.0f * M_PI/180.0f;
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SpeakerAngle[1] = -45.0f * M_PI/180.0f;
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SpeakerAngle[2] = 45.0f * M_PI/180.0f;
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SpeakerAngle[3] = 135.0f * M_PI/180.0f;
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SetSpeakerArrangement("layout_QUAD", SpeakerAngle, Speaker2Chan, Context->NumChan);
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break;
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case AL_FORMAT_51CHN8:
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case AL_FORMAT_51CHN16:
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case AL_FORMAT_51CHN32:
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Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_LEFT][BACK_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_RIGHT][BACK_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
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Context->NumChan = 5;
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Speaker2Chan[0] = BACK_LEFT;
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Speaker2Chan[1] = FRONT_LEFT;
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Speaker2Chan[2] = FRONT_CENTER;
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Speaker2Chan[3] = FRONT_RIGHT;
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Speaker2Chan[4] = BACK_RIGHT;
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SpeakerAngle[0] = -110.0f * M_PI/180.0f;
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SpeakerAngle[1] = -30.0f * M_PI/180.0f;
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SpeakerAngle[2] = 0.0f * M_PI/180.0f;
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SpeakerAngle[3] = 30.0f * M_PI/180.0f;
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SpeakerAngle[4] = 110.0f * M_PI/180.0f;
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SetSpeakerArrangement("layout_51CHN", SpeakerAngle, Speaker2Chan, Context->NumChan);
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break;
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case AL_FORMAT_61CHN8:
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case AL_FORMAT_61CHN16:
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case AL_FORMAT_61CHN32:
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Context->ChannelMatrix[BACK_LEFT][BACK_CENTER] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_LEFT][SIDE_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_RIGHT][BACK_CENTER] = aluSqrt(0.5);
|
|
Context->ChannelMatrix[BACK_RIGHT][SIDE_RIGHT] = aluSqrt(0.5);
|
|
Context->NumChan = 6;
|
|
Speaker2Chan[0] = SIDE_LEFT;
|
|
Speaker2Chan[1] = FRONT_LEFT;
|
|
Speaker2Chan[2] = FRONT_CENTER;
|
|
Speaker2Chan[3] = FRONT_RIGHT;
|
|
Speaker2Chan[4] = SIDE_RIGHT;
|
|
Speaker2Chan[5] = BACK_CENTER;
|
|
SpeakerAngle[0] = -90.0f * M_PI/180.0f;
|
|
SpeakerAngle[1] = -30.0f * M_PI/180.0f;
|
|
SpeakerAngle[2] = 0.0f * M_PI/180.0f;
|
|
SpeakerAngle[3] = 30.0f * M_PI/180.0f;
|
|
SpeakerAngle[4] = 90.0f * M_PI/180.0f;
|
|
SpeakerAngle[5] = 180.0f * M_PI/180.0f;
|
|
SetSpeakerArrangement("layout_61CHN", SpeakerAngle, Speaker2Chan, Context->NumChan);
|
|
break;
|
|
|
|
case AL_FORMAT_71CHN8:
|
|
case AL_FORMAT_71CHN16:
|
|
case AL_FORMAT_71CHN32:
|
|
Context->ChannelMatrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
|
|
Context->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
|
|
Context->NumChan = 7;
|
|
Speaker2Chan[0] = BACK_LEFT;
|
|
Speaker2Chan[1] = SIDE_LEFT;
|
|
Speaker2Chan[2] = FRONT_LEFT;
|
|
Speaker2Chan[3] = FRONT_CENTER;
|
|
Speaker2Chan[4] = FRONT_RIGHT;
|
|
Speaker2Chan[5] = SIDE_RIGHT;
|
|
Speaker2Chan[6] = BACK_RIGHT;
|
|
SpeakerAngle[0] = -150.0f * M_PI/180.0f;
|
|
SpeakerAngle[1] = -90.0f * M_PI/180.0f;
|
|
SpeakerAngle[2] = -30.0f * M_PI/180.0f;
|
|
SpeakerAngle[3] = 0.0f * M_PI/180.0f;
|
|
SpeakerAngle[4] = 30.0f * M_PI/180.0f;
|
|
SpeakerAngle[5] = 90.0f * M_PI/180.0f;
|
|
SpeakerAngle[6] = 150.0f * M_PI/180.0f;
|
|
SetSpeakerArrangement("layout_71CHN", SpeakerAngle, Speaker2Chan, Context->NumChan);
|
|
break;
|
|
|
|
default:
|
|
assert(0);
|
|
}
|
|
|
|
for(pos = 0; pos < LUT_NUM; pos++)
|
|
{
|
|
/* source angle */
|
|
Theta = aluLUTpos2Angle(pos);
|
|
|
|
/* clear all values */
|
|
offset = OUTPUTCHANNELS * pos;
|
|
for(s = 0; s < OUTPUTCHANNELS; s++)
|
|
Context->PanningLUT[offset+s] = 0.0f;
|
|
|
|
/* set panning values */
|
|
for(s = 0; s < Context->NumChan - 1; s++)
|
|
{
|
|
if(Theta >= SpeakerAngle[s] && Theta < SpeakerAngle[s+1])
|
|
{
|
|
/* source between speaker s and speaker s+1 */
|
|
Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
|
|
(SpeakerAngle[s+1]-SpeakerAngle[s]);
|
|
Context->PanningLUT[offset + Speaker2Chan[s]] = cos(Alpha);
|
|
Context->PanningLUT[offset + Speaker2Chan[s+1]] = sin(Alpha);
|
|
break;
|
|
}
|
|
}
|
|
if(s == Context->NumChan - 1)
|
|
{
|
|
/* source between last and first speaker */
|
|
if(Theta < SpeakerAngle[0])
|
|
Theta += 2.0f * M_PI;
|
|
Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
|
|
(2.0f * M_PI + SpeakerAngle[0]-SpeakerAngle[s]);
|
|
Context->PanningLUT[offset + Speaker2Chan[s]] = cos(Alpha);
|
|
Context->PanningLUT[offset + Speaker2Chan[0]] = sin(Alpha);
|
|
}
|
|
}
|
|
}
|
|
|
|
static __inline ALint aluCart2LUTpos(ALfloat re, ALfloat im)
|
|
{
|
|
ALint pos = 0;
|
|
ALfloat denom = aluFabs(re) + aluFabs(im);
|
|
if(denom > 0.0f)
|
|
pos = (ALint)(QUADRANT_NUM*aluFabs(im) / denom + 0.5);
|
|
|
|
if(re < 0.0)
|
|
pos = 2 * QUADRANT_NUM - pos;
|
|
if(im < 0.0)
|
|
pos = LUT_NUM - pos;
|
|
return pos%LUT_NUM;
|
|
}
|
|
|
|
static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
|
|
ALenum isMono, ALfloat *drysend,
|
|
ALfloat *wetsend, ALfloat *pitch,
|
|
ALfloat *drygainhf, ALfloat *wetgainhf)
|
|
{
|
|
ALfloat InnerAngle,OuterAngle,Angle,Distance,DryMix,WetMix=0.0f;
|
|
ALfloat Direction[3],Position[3],SourceToListener[3];
|
|
ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff,OuterGainHF;
|
|
ALfloat ConeVolume,SourceVolume,ListenerGain;
|
|
ALfloat U[3],V[3],N[3];
|
|
ALfloat DopplerFactor, DopplerVelocity, flSpeedOfSound, flMaxVelocity;
|
|
ALfloat Matrix[3][3];
|
|
ALfloat flAttenuation;
|
|
ALfloat RoomAttenuation;
|
|
ALfloat MetersPerUnit;
|
|
ALfloat RoomRolloff;
|
|
ALfloat DryGainHF = 1.0f;
|
|
ALfloat WetGainHF = 1.0f;
|
|
ALfloat DirGain, AmbientGain;
|
|
const ALfloat *SpeakerGain;
|
|
ALint pos, s;
|
|
ALfloat cw, a, g;
|
|
|
|
//Get context properties
|
|
DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
|
|
DopplerVelocity = ALContext->DopplerVelocity;
|
|
flSpeedOfSound = ALContext->flSpeedOfSound;
|
|
|
|
//Get listener properties
|
|
ListenerGain = ALContext->Listener.Gain;
|
|
MetersPerUnit = ALContext->Listener.MetersPerUnit;
|
|
|
|
//Get source properties
|
|
SourceVolume = ALSource->flGain;
|
|
memcpy(Position, ALSource->vPosition, sizeof(ALSource->vPosition));
|
|
memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation));
|
|
MinVolume = ALSource->flMinGain;
|
|
MaxVolume = ALSource->flMaxGain;
|
|
MinDist = ALSource->flRefDistance;
|
|
MaxDist = ALSource->flMaxDistance;
|
|
Rolloff = ALSource->flRollOffFactor;
|
|
InnerAngle = ALSource->flInnerAngle;
|
|
OuterAngle = ALSource->flOuterAngle;
|
|
OuterGainHF = ALSource->OuterGainHF;
|
|
RoomRolloff = ALSource->RoomRolloffFactor;
|
|
|
|
//Only apply 3D calculations for mono buffers
|
|
if(isMono != AL_FALSE)
|
|
{
|
|
//1. Translate Listener to origin (convert to head relative)
|
|
// Note that Direction and SourceToListener are *not* transformed.
|
|
// SourceToListener is used with the source and listener velocities,
|
|
// which are untransformed, and Direction is used with SourceToListener
|
|
// for the sound cone
|
|
if(ALSource->bHeadRelative==AL_FALSE)
|
|
{
|
|
// Build transform matrix
|
|
aluCrossproduct(ALContext->Listener.Forward, ALContext->Listener.Up, U); // Right-vector
|
|
aluNormalize(U); // Normalized Right-vector
|
|
memcpy(V, ALContext->Listener.Up, sizeof(V)); // Up-vector
|
|
aluNormalize(V); // Normalized Up-vector
|
|
memcpy(N, ALContext->Listener.Forward, sizeof(N)); // At-vector
|
|
aluNormalize(N); // Normalized At-vector
|
|
Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0];
|
|
Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1];
|
|
Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2];
|
|
|
|
// Translate source position into listener space
|
|
Position[0] -= ALContext->Listener.Position[0];
|
|
Position[1] -= ALContext->Listener.Position[1];
|
|
Position[2] -= ALContext->Listener.Position[2];
|
|
|
|
SourceToListener[0] = -Position[0];
|
|
SourceToListener[1] = -Position[1];
|
|
SourceToListener[2] = -Position[2];
|
|
|
|
// Transform source position into listener space
|
|
aluMatrixVector(Position, Matrix);
|
|
}
|
|
else
|
|
{
|
|
SourceToListener[0] = -Position[0];
|
|
SourceToListener[1] = -Position[1];
|
|
SourceToListener[2] = -Position[2];
|
|
}
|
|
aluNormalize(SourceToListener);
|
|
aluNormalize(Direction);
|
|
|
|
//2. Calculate distance attenuation
|
|
Distance = aluSqrt(aluDotproduct(Position, Position));
|
|
|
|
if(ALSource->Send[0].Slot)
|
|
{
|
|
if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB)
|
|
RoomRolloff += ALSource->Send[0].Slot->effect.Reverb.RoomRolloffFactor;
|
|
}
|
|
|
|
flAttenuation = 1.0f;
|
|
RoomAttenuation = 1.0f;
|
|
switch (ALSource->DistanceModel)
|
|
{
|
|
case AL_INVERSE_DISTANCE_CLAMPED:
|
|
Distance=__max(Distance,MinDist);
|
|
Distance=__min(Distance,MaxDist);
|
|
if (MaxDist < MinDist)
|
|
break;
|
|
//fall-through
|
|
case AL_INVERSE_DISTANCE:
|
|
if (MinDist > 0.0f)
|
|
{
|
|
if ((MinDist + (Rolloff * (Distance - MinDist))) > 0.0f)
|
|
flAttenuation = MinDist / (MinDist + (Rolloff * (Distance - MinDist)));
|
|
if ((MinDist + (RoomRolloff * (Distance - MinDist))) > 0.0f)
|
|
RoomAttenuation = MinDist / (MinDist + (RoomRolloff * (Distance - MinDist)));
|
|
}
|
|
break;
|
|
|
|
case AL_LINEAR_DISTANCE_CLAMPED:
|
|
Distance=__max(Distance,MinDist);
|
|
Distance=__min(Distance,MaxDist);
|
|
if (MaxDist < MinDist)
|
|
break;
|
|
//fall-through
|
|
case AL_LINEAR_DISTANCE:
|
|
Distance=__min(Distance,MaxDist);
|
|
if (MaxDist != MinDist)
|
|
{
|
|
flAttenuation = 1.0f - (Rolloff*(Distance-MinDist)/(MaxDist - MinDist));
|
|
RoomAttenuation = 1.0f - (RoomRolloff*(Distance-MinDist)/(MaxDist - MinDist));
|
|
}
|
|
break;
|
|
|
|
case AL_EXPONENT_DISTANCE_CLAMPED:
|
|
Distance=__max(Distance,MinDist);
|
|
Distance=__min(Distance,MaxDist);
|
|
if (MaxDist < MinDist)
|
|
break;
|
|
//fall-through
|
|
case AL_EXPONENT_DISTANCE:
|
|
if ((Distance > 0.0f) && (MinDist > 0.0f))
|
|
{
|
|
flAttenuation = (ALfloat)pow(Distance/MinDist, -Rolloff);
|
|
RoomAttenuation = (ALfloat)pow(Distance/MinDist, -RoomRolloff);
|
|
}
|
|
break;
|
|
|
|
case AL_NONE:
|
|
flAttenuation = 1.0f;
|
|
RoomAttenuation = 1.0f;
|
|
break;
|
|
}
|
|
|
|
// Distance-based air absorption
|
|
if(ALSource->AirAbsorptionFactor > 0.0f && ALContext->DistanceModel != AL_NONE)
|
|
{
|
|
ALfloat dist = Distance-MinDist;
|
|
ALfloat absorb;
|
|
|
|
if(dist < 0.0f) dist = 0.0f;
|
|
// Absorption calculation is done in dB
|
|
absorb = (ALSource->AirAbsorptionFactor*AIRABSORBGAINDBHF) *
|
|
(dist*MetersPerUnit);
|
|
// Convert dB to linear gain before applying
|
|
absorb = pow(10.0, absorb/20.0);
|
|
DryGainHF *= absorb;
|
|
WetGainHF *= absorb;
|
|
}
|
|
|
|
// Source Gain + Attenuation and clamp to Min/Max Gain
|
|
DryMix = SourceVolume * flAttenuation;
|
|
DryMix = __min(DryMix,MaxVolume);
|
|
DryMix = __max(DryMix,MinVolume);
|
|
|
|
WetMix = SourceVolume * RoomAttenuation;
|
|
WetMix = __min(WetMix,MaxVolume);
|
|
WetMix = __max(WetMix,MinVolume);
|
|
|
|
//3. Apply directional soundcones
|
|
Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f/M_PI;
|
|
if(Angle >= InnerAngle && Angle <= OuterAngle)
|
|
{
|
|
ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
|
|
ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)*scale);
|
|
DryMix *= ConeVolume;
|
|
if(ALSource->WetGainAuto)
|
|
WetMix *= ConeVolume;
|
|
if(ALSource->DryGainHFAuto)
|
|
DryGainHF *= (1.0f+(OuterGainHF-1.0f)*scale);
|
|
if(ALSource->WetGainHFAuto)
|
|
WetGainHF *= (1.0f+(OuterGainHF-1.0f)*scale);
|
|
}
|
|
else if(Angle > OuterAngle)
|
|
{
|
|
ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f));
|
|
DryMix *= ConeVolume;
|
|
if(ALSource->WetGainAuto)
|
|
WetMix *= ConeVolume;
|
|
if(ALSource->DryGainHFAuto)
|
|
DryGainHF *= (1.0f+(OuterGainHF-1.0f));
|
|
if(ALSource->WetGainHFAuto)
|
|
WetGainHF *= (1.0f+(OuterGainHF-1.0f));
|
|
}
|
|
|
|
//4. Calculate Velocity
|
|
if(DopplerFactor != 0.0f)
|
|
{
|
|
ALfloat flVSS, flVLS = 0.0f;
|
|
|
|
if(ALSource->bHeadRelative==AL_FALSE)
|
|
flVLS = aluDotproduct(ALContext->Listener.Velocity, SourceToListener);
|
|
flVSS = aluDotproduct(ALSource->vVelocity, SourceToListener);
|
|
|
|
flMaxVelocity = (DopplerVelocity * flSpeedOfSound) / DopplerFactor;
|
|
|
|
if (flVSS >= flMaxVelocity)
|
|
flVSS = (flMaxVelocity - 1.0f);
|
|
else if (flVSS <= -flMaxVelocity)
|
|
flVSS = -flMaxVelocity + 1.0f;
|
|
|
|
if (flVLS >= flMaxVelocity)
|
|
flVLS = (flMaxVelocity - 1.0f);
|
|
else if (flVLS <= -flMaxVelocity)
|
|
flVLS = -flMaxVelocity + 1.0f;
|
|
|
|
pitch[0] = ALSource->flPitch *
|
|
((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVLS)) /
|
|
((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVSS));
|
|
}
|
|
else
|
|
pitch[0] = ALSource->flPitch;
|
|
|
|
if(ALSource->Send[0].Slot &&
|
|
ALSource->Send[0].Slot->effect.type != AL_EFFECT_NULL)
|
|
{
|
|
if(ALSource->Send[0].Slot->AuxSendAuto)
|
|
{
|
|
// Apply minimal attenuation in place of missing statistical
|
|
// reverb model.
|
|
WetMix *= pow(DryMix, 1.0f / 2.0f);
|
|
}
|
|
else
|
|
{
|
|
// If the slot's auxilliary send auto is off, the data sent to the
|
|
// effect slot is the same as the dry path, sans filter effects
|
|
WetMix = DryMix;
|
|
WetGainHF = DryGainHF;
|
|
}
|
|
|
|
// Note that this is really applied by the effect slot. However,
|
|
// it's easier (more optimal) to handle it here.
|
|
if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB)
|
|
WetGainHF *= ALSource->Send[0].Slot->effect.Reverb.GainHF;
|
|
}
|
|
else
|
|
{
|
|
WetMix = 0.0f;
|
|
WetGainHF = 1.0f;
|
|
}
|
|
|
|
//5. Apply filter gains and filters
|
|
switch(ALSource->DirectFilter.type)
|
|
{
|
|
case AL_FILTER_LOWPASS:
|
|
DryMix *= ALSource->DirectFilter.Gain;
|
|
DryGainHF *= ALSource->DirectFilter.GainHF;
|
|
break;
|
|
}
|
|
|
|
switch(ALSource->Send[0].WetFilter.type)
|
|
{
|
|
case AL_FILTER_LOWPASS:
|
|
WetMix *= ALSource->Send[0].WetFilter.Gain;
|
|
WetGainHF *= ALSource->Send[0].WetFilter.GainHF;
|
|
break;
|
|
}
|
|
|
|
DryMix *= ListenerGain;
|
|
WetMix *= ListenerGain;
|
|
|
|
// Use energy-preserving panning algorithm for multi-speaker playback
|
|
aluNormalize(Position);
|
|
|
|
pos = aluCart2LUTpos(-Position[2], Position[0]);
|
|
SpeakerGain = &ALContext->PanningLUT[OUTPUTCHANNELS * pos];
|
|
|
|
DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
|
|
// elevation adjustment for directional gain. this sucks, but
|
|
// has low complexity
|
|
AmbientGain = 1.0/aluSqrt(ALContext->NumChan) * (1.0-DirGain);
|
|
for(s = 0; s < OUTPUTCHANNELS; s++)
|
|
{
|
|
ALfloat gain = SpeakerGain[s]*DirGain + AmbientGain;
|
|
drysend[s] = DryMix * gain;
|
|
}
|
|
*wetsend = WetMix;
|
|
|
|
// Update filter coefficients. Calculations based on the I3DL2 spec.
|
|
cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / ALContext->Frequency);
|
|
// We use four chained one-pole filters, so we need to take the fourth
|
|
// root of the squared gain, which is the same as the square root of
|
|
// the base gain.
|
|
// Be careful with gains < 0.0001, as that causes the coefficient to
|
|
// head towards 1, which will flatten the signal
|
|
g = aluSqrt(__max(DryGainHF, 0.0001f));
|
|
a = 0.0f;
|
|
if(g < 0.9999f) // 1-epsilon
|
|
a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g);
|
|
ALSource->iirFilter.coeff = a;
|
|
|
|
g = aluSqrt(__max(WetGainHF, 0.0001f));
|
|
a = 0.0f;
|
|
if(g < 0.9999f) // 1-epsilon
|
|
a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g);
|
|
ALSource->Send[0].iirFilter.coeff = a;
|
|
|
|
*drygainhf = DryGainHF;
|
|
*wetgainhf = WetGainHF;
|
|
}
|
|
else
|
|
{
|
|
//1. Multi-channel buffers always play "normal"
|
|
pitch[0] = ALSource->flPitch;
|
|
|
|
DryMix = SourceVolume;
|
|
|
|
switch(ALSource->DirectFilter.type)
|
|
{
|
|
case AL_FILTER_LOWPASS:
|
|
DryMix *= ALSource->DirectFilter.Gain;
|
|
DryGainHF *= ALSource->DirectFilter.GainHF;
|
|
break;
|
|
}
|
|
|
|
drysend[FRONT_LEFT] = DryMix * ListenerGain;
|
|
drysend[FRONT_RIGHT] = DryMix * ListenerGain;
|
|
drysend[SIDE_LEFT] = DryMix * ListenerGain;
|
|
drysend[SIDE_RIGHT] = DryMix * ListenerGain;
|
|
drysend[BACK_LEFT] = DryMix * ListenerGain;
|
|
drysend[BACK_RIGHT] = DryMix * ListenerGain;
|
|
drysend[FRONT_CENTER] = DryMix * ListenerGain;
|
|
drysend[BACK_CENTER] = DryMix * ListenerGain;
|
|
drysend[LFE] = DryMix * ListenerGain;
|
|
*wetsend = 0.0f;
|
|
|
|
cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / ALContext->Frequency);
|
|
g = __max(DryGainHF, 0.01f);
|
|
a = 0.0f;
|
|
if(g < 0.9999f) // 1-epsilon
|
|
a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g);
|
|
ALSource->iirFilter.coeff = a;
|
|
ALSource->Send[0].iirFilter.coeff = 0.0f;
|
|
|
|
*drygainhf = DryGainHF;
|
|
*wetgainhf = WetGainHF;
|
|
}
|
|
}
|
|
|
|
static __inline ALshort lerp(ALshort val1, ALshort val2, ALint frac)
|
|
{
|
|
return val1 + (((val2-val1)*frac)>>FRACTIONBITS);
|
|
}
|
|
|
|
ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum format)
|
|
{
|
|
static float DryBuffer[BUFFERSIZE][OUTPUTCHANNELS];
|
|
static float WetBuffer[BUFFERSIZE];
|
|
ALfloat (*Matrix)[OUTPUTCHANNELS] = ALContext->ChannelMatrix;
|
|
ALfloat newDrySend[OUTPUTCHANNELS] = { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f };
|
|
ALfloat newWetSend = 0.0f;
|
|
ALfloat DryGainHF = 0.0f;
|
|
ALfloat WetGainHF = 0.0f;
|
|
ALfloat *DrySend;
|
|
ALfloat *WetSend;
|
|
ALuint rampLength;
|
|
ALfloat dryGainStep[OUTPUTCHANNELS];
|
|
ALfloat wetGainStep;
|
|
ALuint BlockAlign,BufferSize;
|
|
ALuint DataSize=0,DataPosInt=0,DataPosFrac=0;
|
|
ALuint Channels,Frequency,ulExtraSamples;
|
|
ALfloat Pitch;
|
|
ALint Looping,State;
|
|
ALint increment;
|
|
ALuint Buffer;
|
|
ALuint SamplesToDo;
|
|
ALsource *ALSource;
|
|
ALbuffer *ALBuffer;
|
|
ALeffectslot *ALEffectSlot;
|
|
ALfloat values[OUTPUTCHANNELS];
|
|
ALfloat value;
|
|
ALshort *Data;
|
|
ALuint i,j,k,out;
|
|
ALbufferlistitem *BufferListItem;
|
|
ALuint loop;
|
|
ALint64 DataSize64,DataPos64;
|
|
FILTER *DryFilter, *WetFilter;
|
|
int fpuState;
|
|
|
|
SuspendContext(ALContext);
|
|
|
|
#if defined(HAVE_FESETROUND)
|
|
fpuState = fegetround();
|
|
fesetround(FE_TOWARDZERO);
|
|
#elif defined(HAVE__CONTROLFP)
|
|
fpuState = _controlfp(0, 0);
|
|
_controlfp(_RC_CHOP, _MCW_RC);
|
|
#else
|
|
(void)fpuState;
|
|
#endif
|
|
|
|
//Figure output format variables
|
|
BlockAlign = aluChannelsFromFormat(format);
|
|
BlockAlign *= aluBytesFromFormat(format);
|
|
|
|
size /= BlockAlign;
|
|
while(size > 0)
|
|
{
|
|
//Setup variables
|
|
SamplesToDo = min(size, BUFFERSIZE);
|
|
if(ALContext)
|
|
{
|
|
ALEffectSlot = ALContext->AuxiliaryEffectSlot;
|
|
ALSource = ALContext->Source;
|
|
rampLength = ALContext->Frequency * MIN_RAMP_LENGTH / 1000;
|
|
}
|
|
else
|
|
{
|
|
ALEffectSlot = NULL;
|
|
ALSource = NULL;
|
|
rampLength = 0;
|
|
}
|
|
rampLength = max(rampLength, SamplesToDo);
|
|
|
|
//Clear mixing buffer
|
|
memset(WetBuffer, 0, SamplesToDo*sizeof(ALfloat));
|
|
memset(DryBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat));
|
|
|
|
//Actual mixing loop
|
|
while(ALSource)
|
|
{
|
|
j = 0;
|
|
State = ALSource->state;
|
|
|
|
while(State == AL_PLAYING && j < SamplesToDo)
|
|
{
|
|
DataSize = 0;
|
|
DataPosInt = 0;
|
|
DataPosFrac = 0;
|
|
|
|
//Get buffer info
|
|
if((Buffer = ALSource->ulBufferID))
|
|
{
|
|
ALBuffer = (ALbuffer*)ALTHUNK_LOOKUPENTRY(Buffer);
|
|
|
|
Data = ALBuffer->data;
|
|
Channels = aluChannelsFromFormat(ALBuffer->format);
|
|
DataSize = ALBuffer->size;
|
|
DataSize /= Channels * aluBytesFromFormat(ALBuffer->format);
|
|
Frequency = ALBuffer->frequency;
|
|
DataPosInt = ALSource->position;
|
|
DataPosFrac = ALSource->position_fraction;
|
|
|
|
if(DataPosInt >= DataSize)
|
|
goto skipmix;
|
|
|
|
CalcSourceParams(ALContext, ALSource,
|
|
(Channels==1) ? AL_TRUE : AL_FALSE,
|
|
newDrySend, &newWetSend, &Pitch,
|
|
&DryGainHF, &WetGainHF);
|
|
|
|
Pitch = (Pitch*Frequency) / ALContext->Frequency;
|
|
|
|
if(DuplicateStereo && Channels > 1)
|
|
{
|
|
if(Channels == 2)
|
|
{
|
|
Matrix[FRONT_LEFT][SIDE_LEFT] = 1.0f;
|
|
Matrix[FRONT_RIGHT][SIDE_RIGHT] = 1.0f;
|
|
Matrix[FRONT_LEFT][BACK_LEFT] = 1.0f;
|
|
Matrix[FRONT_RIGHT][BACK_RIGHT] = 1.0f;
|
|
}
|
|
else
|
|
{
|
|
Matrix[FRONT_LEFT][SIDE_LEFT] = 0.0f;
|
|
Matrix[FRONT_RIGHT][SIDE_RIGHT] = 0.0f;
|
|
Matrix[FRONT_LEFT][BACK_LEFT] = 0.0f;
|
|
Matrix[FRONT_RIGHT][BACK_RIGHT] = 0.0f;
|
|
}
|
|
}
|
|
|
|
//Get source info
|
|
DryFilter = &ALSource->iirFilter;
|
|
WetFilter = &ALSource->Send[0].iirFilter;
|
|
DrySend = ALSource->DryGains;
|
|
WetSend = &ALSource->WetGain;
|
|
|
|
//Compute the gain steps for each output channel
|
|
if(ALSource->FirstStart && DataPosInt == 0 && DataPosFrac == 0)
|
|
{
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
{
|
|
DrySend[i] = newDrySend[i];
|
|
dryGainStep[i] = 0;
|
|
}
|
|
*WetSend = newWetSend;
|
|
wetGainStep = 0;
|
|
}
|
|
else
|
|
{
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
dryGainStep[i] = (newDrySend[i]-DrySend[i]) / rampLength;
|
|
wetGainStep = (newWetSend-(*WetSend)) / rampLength;
|
|
}
|
|
ALSource->FirstStart = AL_FALSE;
|
|
|
|
//Compute 18.14 fixed point step
|
|
if(Pitch > (float)MAX_PITCH)
|
|
Pitch = (float)MAX_PITCH;
|
|
increment = (ALint)(Pitch*(ALfloat)(1L<<FRACTIONBITS));
|
|
if(increment <= 0)
|
|
increment = (1<<FRACTIONBITS);
|
|
|
|
//Figure out how many samples we can mix.
|
|
DataSize64 = DataSize;
|
|
DataSize64 <<= FRACTIONBITS;
|
|
DataPos64 = DataPosInt;
|
|
DataPos64 <<= FRACTIONBITS;
|
|
DataPos64 += DataPosFrac;
|
|
BufferSize = (ALuint)((DataSize64-DataPos64+(increment-1)) / increment);
|
|
|
|
BufferListItem = ALSource->queue;
|
|
for(loop = 0; loop < ALSource->BuffersPlayed; loop++)
|
|
{
|
|
if(BufferListItem)
|
|
BufferListItem = BufferListItem->next;
|
|
}
|
|
if (BufferListItem)
|
|
{
|
|
if (BufferListItem->next)
|
|
{
|
|
ALbuffer *NextBuf = (ALbuffer*)ALTHUNK_LOOKUPENTRY(BufferListItem->next->buffer);
|
|
if(NextBuf && NextBuf->data)
|
|
{
|
|
ulExtraSamples = min(NextBuf->size, (ALint)(ALBuffer->padding*Channels*2));
|
|
memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
|
|
}
|
|
}
|
|
else if (ALSource->bLooping)
|
|
{
|
|
ALbuffer *NextBuf = (ALbuffer*)ALTHUNK_LOOKUPENTRY(ALSource->queue->buffer);
|
|
if (NextBuf && NextBuf->data)
|
|
{
|
|
ulExtraSamples = min(NextBuf->size, (ALint)(ALBuffer->padding*Channels*2));
|
|
memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
|
|
}
|
|
}
|
|
else
|
|
memset(&Data[DataSize*Channels], 0, (ALBuffer->padding*Channels*2));
|
|
}
|
|
BufferSize = min(BufferSize, (SamplesToDo-j));
|
|
|
|
//Actual sample mixing loop
|
|
k = 0;
|
|
Data += DataPosInt*Channels;
|
|
|
|
if(Channels == 1) /* Mono */
|
|
{
|
|
ALfloat outsamp;
|
|
|
|
while(BufferSize--)
|
|
{
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
DrySend[i] += dryGainStep[i];
|
|
*WetSend += wetGainStep;
|
|
|
|
//First order interpolator
|
|
value = lerp(Data[k], Data[k+1], DataPosFrac);
|
|
|
|
//Direct path final mix buffer and panning
|
|
outsamp = lpFilter(DryFilter, value);
|
|
DryBuffer[j][FRONT_LEFT] += outsamp*DrySend[FRONT_LEFT];
|
|
DryBuffer[j][FRONT_RIGHT] += outsamp*DrySend[FRONT_RIGHT];
|
|
DryBuffer[j][SIDE_LEFT] += outsamp*DrySend[SIDE_LEFT];
|
|
DryBuffer[j][SIDE_RIGHT] += outsamp*DrySend[SIDE_RIGHT];
|
|
DryBuffer[j][BACK_LEFT] += outsamp*DrySend[BACK_LEFT];
|
|
DryBuffer[j][BACK_RIGHT] += outsamp*DrySend[BACK_RIGHT];
|
|
DryBuffer[j][FRONT_CENTER] += outsamp*DrySend[FRONT_CENTER];
|
|
DryBuffer[j][BACK_CENTER] += outsamp*DrySend[BACK_CENTER];
|
|
|
|
//Room path final mix buffer and panning
|
|
outsamp = lpFilter(WetFilter, value);
|
|
WetBuffer[j] += outsamp*(*WetSend);
|
|
|
|
DataPosFrac += increment;
|
|
k += DataPosFrac>>FRACTIONBITS;
|
|
DataPosFrac &= FRACTIONMASK;
|
|
j++;
|
|
}
|
|
}
|
|
else if(Channels == 2) /* Stereo */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT
|
|
};
|
|
|
|
#define DO_MIX() do { \
|
|
*WetSend += wetGainStep*BufferSize; \
|
|
while(BufferSize--) \
|
|
{ \
|
|
for(i = 0;i < OUTPUTCHANNELS;i++) \
|
|
DrySend[i] += dryGainStep[i]; \
|
|
\
|
|
for(i = 0;i < Channels;i++) \
|
|
{ \
|
|
value = lerp(Data[k*Channels + i], Data[(k+1)*Channels + i], DataPosFrac); \
|
|
values[i] = lpFilterMC(DryFilter, chans[i], value)*DrySend[chans[i]]; \
|
|
} \
|
|
for(out = 0;out < OUTPUTCHANNELS;out++) \
|
|
{ \
|
|
ALfloat sum = 0.0f; \
|
|
for(i = 0;i < Channels;i++) \
|
|
sum += values[i]*Matrix[chans[i]][out]; \
|
|
DryBuffer[j][out] += sum; \
|
|
} \
|
|
\
|
|
DataPosFrac += increment; \
|
|
k += DataPosFrac>>FRACTIONBITS; \
|
|
DataPosFrac &= FRACTIONMASK; \
|
|
j++; \
|
|
} \
|
|
} while(0)
|
|
|
|
DO_MIX();
|
|
}
|
|
else if(Channels == 4) /* Quad */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
BACK_LEFT, BACK_RIGHT
|
|
};
|
|
|
|
DO_MIX();
|
|
}
|
|
else if(Channels == 6) /* 5.1 */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
FRONT_CENTER, LFE,
|
|
BACK_LEFT, BACK_RIGHT
|
|
};
|
|
|
|
DO_MIX();
|
|
}
|
|
else if(Channels == 7) /* 6.1 */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
FRONT_CENTER, LFE,
|
|
BACK_CENTER,
|
|
SIDE_LEFT, SIDE_RIGHT
|
|
};
|
|
|
|
DO_MIX();
|
|
}
|
|
else if(Channels == 8) /* 7.1 */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
FRONT_CENTER, LFE,
|
|
BACK_LEFT, BACK_RIGHT,
|
|
SIDE_LEFT, SIDE_RIGHT
|
|
};
|
|
|
|
DO_MIX();
|
|
#undef DO_MIX
|
|
}
|
|
else /* Unknown? */
|
|
{
|
|
*WetSend += wetGainStep*BufferSize;
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
DrySend[i] += dryGainStep[i]*BufferSize;
|
|
while(BufferSize--)
|
|
{
|
|
DataPosFrac += increment;
|
|
k += DataPosFrac>>FRACTIONBITS;
|
|
DataPosFrac &= FRACTIONMASK;
|
|
j++;
|
|
}
|
|
}
|
|
DataPosInt += k;
|
|
|
|
//Update source info
|
|
ALSource->position = DataPosInt;
|
|
ALSource->position_fraction = DataPosFrac;
|
|
|
|
skipmix: ;
|
|
}
|
|
|
|
//Handle looping sources
|
|
if(!Buffer || DataPosInt >= DataSize)
|
|
{
|
|
//queueing
|
|
if(ALSource->queue)
|
|
{
|
|
Looping = ALSource->bLooping;
|
|
if(ALSource->BuffersPlayed < (ALSource->BuffersInQueue-1))
|
|
{
|
|
BufferListItem = ALSource->queue;
|
|
for(loop = 0; loop <= ALSource->BuffersPlayed; loop++)
|
|
{
|
|
if(BufferListItem)
|
|
{
|
|
if(!Looping)
|
|
BufferListItem->bufferstate = PROCESSED;
|
|
BufferListItem = BufferListItem->next;
|
|
}
|
|
}
|
|
if(BufferListItem)
|
|
ALSource->ulBufferID = BufferListItem->buffer;
|
|
ALSource->position = DataPosInt-DataSize;
|
|
ALSource->position_fraction = DataPosFrac;
|
|
ALSource->BuffersPlayed++;
|
|
}
|
|
else
|
|
{
|
|
if(!Looping)
|
|
{
|
|
/* alSourceStop */
|
|
ALSource->state = AL_STOPPED;
|
|
ALSource->inuse = AL_FALSE;
|
|
ALSource->BuffersPlayed = ALSource->BuffersInQueue;
|
|
BufferListItem = ALSource->queue;
|
|
while(BufferListItem != NULL)
|
|
{
|
|
BufferListItem->bufferstate = PROCESSED;
|
|
BufferListItem = BufferListItem->next;
|
|
}
|
|
ALSource->position = DataSize;
|
|
ALSource->position_fraction = 0;
|
|
}
|
|
else
|
|
{
|
|
/* alSourceRewind */
|
|
/* alSourcePlay */
|
|
ALSource->state = AL_PLAYING;
|
|
ALSource->inuse = AL_TRUE;
|
|
ALSource->play = AL_TRUE;
|
|
ALSource->BuffersPlayed = 0;
|
|
BufferListItem = ALSource->queue;
|
|
while(BufferListItem != NULL)
|
|
{
|
|
BufferListItem->bufferstate = PENDING;
|
|
BufferListItem = BufferListItem->next;
|
|
}
|
|
ALSource->ulBufferID = ALSource->queue->buffer;
|
|
|
|
if(ALSource->BuffersInQueue == 1)
|
|
ALSource->position = DataPosInt%DataSize;
|
|
else
|
|
ALSource->position = DataPosInt-DataSize;
|
|
ALSource->position_fraction = DataPosFrac;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
//Get source state
|
|
State = ALSource->state;
|
|
}
|
|
|
|
ALSource = ALSource->next;
|
|
}
|
|
|
|
// effect slot processing
|
|
while(ALEffectSlot)
|
|
{
|
|
if(ALEffectSlot->effect.type == AL_EFFECT_REVERB)
|
|
VerbProcess(ALEffectSlot->ReverbState, SamplesToDo, WetBuffer, DryBuffer);
|
|
|
|
ALEffectSlot = ALEffectSlot->next;
|
|
}
|
|
|
|
//Post processing loop
|
|
switch(format)
|
|
{
|
|
case AL_FORMAT_MONO8:
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT]+DryBuffer[i][FRONT_RIGHT])>>8)+128);
|
|
buffer = ((ALubyte*)buffer) + 1;
|
|
}
|
|
break;
|
|
case AL_FORMAT_STEREO8:
|
|
if(ALContext && ALContext->bs2b)
|
|
{
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
float samples[2];
|
|
samples[0] = DryBuffer[i][FRONT_LEFT];
|
|
samples[1] = DryBuffer[i][FRONT_RIGHT];
|
|
bs2b_cross_feed(ALContext->bs2b, samples);
|
|
((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(samples[0])>>8)+128);
|
|
((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(samples[1])>>8)+128);
|
|
buffer = ((ALubyte*)buffer) + 2;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT])>>8)+128);
|
|
buffer = ((ALubyte*)buffer) + 2;
|
|
}
|
|
}
|
|
break;
|
|
case AL_FORMAT_QUAD8:
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT])>>8)+128);
|
|
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128);
|
|
buffer = ((ALubyte*)buffer) + 4;
|
|
}
|
|
break;
|
|
case AL_FORMAT_51CHN8:
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT])>>8)+128);
|
|
#ifdef _WIN32 /* Of course, Windows can't use the same ordering... */
|
|
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_CENTER])>>8)+128);
|
|
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128);
|
|
((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128);
|
|
#else
|
|
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128);
|
|
((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_CENTER])>>8)+128);
|
|
((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128);
|
|
#endif
|
|
buffer = ((ALubyte*)buffer) + 6;
|
|
}
|
|
break;
|
|
case AL_FORMAT_61CHN8:
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT])>>8)+128);
|
|
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_CENTER])>>8)+128);
|
|
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128);
|
|
((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][BACK_CENTER])>>8)+128);
|
|
((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[6] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_RIGHT])>>8)+128);
|
|
buffer = ((ALubyte*)buffer) + 7;
|
|
}
|
|
break;
|
|
case AL_FORMAT_71CHN8:
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT])>>8)+128);
|
|
#ifdef _WIN32
|
|
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_CENTER])>>8)+128);
|
|
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128);
|
|
((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128);
|
|
#else
|
|
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT])>>8)+128);
|
|
((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_CENTER])>>8)+128);
|
|
((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128);
|
|
#endif
|
|
((ALubyte*)buffer)[6] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[7] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_RIGHT])>>8)+128);
|
|
buffer = ((ALubyte*)buffer) + 8;
|
|
}
|
|
break;
|
|
|
|
case AL_FORMAT_MONO16:
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]+DryBuffer[i][FRONT_RIGHT]);
|
|
buffer = ((ALshort*)buffer) + 1;
|
|
}
|
|
break;
|
|
case AL_FORMAT_STEREO16:
|
|
if(ALContext && ALContext->bs2b)
|
|
{
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
float samples[2];
|
|
samples[0] = DryBuffer[i][FRONT_LEFT];
|
|
samples[1] = DryBuffer[i][FRONT_RIGHT];
|
|
bs2b_cross_feed(ALContext->bs2b, samples);
|
|
((ALshort*)buffer)[0] = aluF2S(samples[0]);
|
|
((ALshort*)buffer)[1] = aluF2S(samples[1]);
|
|
buffer = ((ALshort*)buffer) + 2;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]);
|
|
((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]);
|
|
buffer = ((ALshort*)buffer) + 2;
|
|
}
|
|
}
|
|
break;
|
|
case AL_FORMAT_QUAD16:
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]);
|
|
((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]);
|
|
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][BACK_LEFT]);
|
|
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_RIGHT]);
|
|
buffer = ((ALshort*)buffer) + 4;
|
|
}
|
|
break;
|
|
case AL_FORMAT_51CHN16:
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]);
|
|
((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]);
|
|
#ifdef _WIN32
|
|
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][FRONT_CENTER]);
|
|
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][LFE]);
|
|
((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][BACK_LEFT]);
|
|
((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][BACK_RIGHT]);
|
|
#else
|
|
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][BACK_LEFT]);
|
|
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_RIGHT]);
|
|
((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][FRONT_CENTER]);
|
|
((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][LFE]);
|
|
#endif
|
|
buffer = ((ALshort*)buffer) + 6;
|
|
}
|
|
break;
|
|
case AL_FORMAT_61CHN16:
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]);
|
|
((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]);
|
|
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][FRONT_CENTER]);
|
|
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][LFE]);
|
|
((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][BACK_CENTER]);
|
|
((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][SIDE_LEFT]);
|
|
((ALshort*)buffer)[6] = aluF2S(DryBuffer[i][SIDE_RIGHT]);
|
|
buffer = ((ALshort*)buffer) + 7;
|
|
}
|
|
break;
|
|
case AL_FORMAT_71CHN16:
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
((ALshort*)buffer)[0] = aluF2S(DryBuffer[i][FRONT_LEFT]);
|
|
((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]);
|
|
#ifdef _WIN32
|
|
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][FRONT_CENTER]);
|
|
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][LFE]);
|
|
((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][BACK_LEFT]);
|
|
((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][BACK_RIGHT]);
|
|
#else
|
|
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][BACK_LEFT]);
|
|
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_RIGHT]);
|
|
((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][FRONT_CENTER]);
|
|
((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][LFE]);
|
|
#endif
|
|
((ALshort*)buffer)[6] = aluF2S(DryBuffer[i][SIDE_LEFT]);
|
|
((ALshort*)buffer)[7] = aluF2S(DryBuffer[i][SIDE_RIGHT]);
|
|
buffer = ((ALshort*)buffer) + 8;
|
|
}
|
|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
|
|
size -= SamplesToDo;
|
|
}
|
|
|
|
#if defined(HAVE_FESETROUND)
|
|
fesetround(fpuState);
|
|
#elif defined(HAVE__CONTROLFP)
|
|
_controlfp(fpuState, 0xfffff);
|
|
#endif
|
|
|
|
ProcessContext(ALContext);
|
|
}
|