1218 lines
49 KiB
C
1218 lines
49 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 "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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#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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// 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 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 CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
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ALenum isMono, ALenum OutputFormat,
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ALfloat *drysend, ALfloat *wetsend,
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ALfloat *pitch, ALfloat *drygainhf,
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ALfloat *wetgainhf)
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{
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ALfloat InnerAngle,OuterAngle,Angle,Distance,DryMix,WetMix=0.0f;
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ALfloat Direction[3],Position[3],SourceToListener[3];
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ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff,OuterGainHF;
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ALfloat ConeVolume,SourceVolume,PanningFB,PanningLR,ListenerGain;
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ALfloat U[3],V[3],N[3];
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ALfloat DopplerFactor, DopplerVelocity, flSpeedOfSound, flMaxVelocity;
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ALfloat Matrix[3][3];
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ALfloat flAttenuation;
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ALfloat RoomAttenuation;
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ALfloat MetersPerUnit;
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ALfloat RoomRolloff;
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ALfloat DryGainHF = 1.0f;
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ALfloat WetGainHF = 1.0f;
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ALfloat cw, a, g;
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//Get context properties
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DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
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DopplerVelocity = ALContext->DopplerVelocity;
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flSpeedOfSound = ALContext->flSpeedOfSound;
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//Get listener properties
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ListenerGain = ALContext->Listener.Gain;
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MetersPerUnit = ALContext->Listener.MetersPerUnit;
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//Get source properties
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SourceVolume = ALSource->flGain;
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memcpy(Position, ALSource->vPosition, sizeof(ALSource->vPosition));
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memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation));
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MinVolume = ALSource->flMinGain;
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MaxVolume = ALSource->flMaxGain;
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MinDist = ALSource->flRefDistance;
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MaxDist = ALSource->flMaxDistance;
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Rolloff = ALSource->flRollOffFactor;
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InnerAngle = ALSource->flInnerAngle;
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OuterAngle = ALSource->flOuterAngle;
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OuterGainHF = ALSource->OuterGainHF;
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RoomRolloff = ALSource->RoomRolloffFactor;
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//Only apply 3D calculations for mono buffers
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if(isMono != AL_FALSE)
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{
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//1. Translate Listener to origin (convert to head relative)
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// Note that Direction and SourceToListener are *not* transformed.
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// SourceToListener is used with the source and listener velocities,
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// which are untransformed, and Direction is used with SourceToListener
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// for the sound cone
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if(ALSource->bHeadRelative==AL_FALSE)
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{
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// Build transform matrix
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aluCrossproduct(ALContext->Listener.Forward, ALContext->Listener.Up, U); // Right-vector
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aluNormalize(U); // Normalized Right-vector
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memcpy(V, ALContext->Listener.Up, sizeof(V)); // Up-vector
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aluNormalize(V); // Normalized Up-vector
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memcpy(N, ALContext->Listener.Forward, sizeof(N)); // At-vector
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aluNormalize(N); // Normalized At-vector
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Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0];
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Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1];
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Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2];
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// Translate source position into listener space
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Position[0] -= ALContext->Listener.Position[0];
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Position[1] -= ALContext->Listener.Position[1];
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Position[2] -= ALContext->Listener.Position[2];
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SourceToListener[0] = -Position[0];
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SourceToListener[1] = -Position[1];
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SourceToListener[2] = -Position[2];
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// Transform source position and direction into listener space
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aluMatrixVector(Position, Matrix);
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}
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else
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{
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SourceToListener[0] = -Position[0];
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SourceToListener[1] = -Position[1];
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SourceToListener[2] = -Position[2];
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}
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aluNormalize(SourceToListener);
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aluNormalize(Direction);
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//2. Calculate distance attenuation
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Distance = aluSqrt(aluDotproduct(Position, Position));
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if(ALSource->Send[0].Slot)
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{
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if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB)
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RoomRolloff += ALSource->Send[0].Slot->effect.Reverb.RoomRolloffFactor;
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}
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flAttenuation = 1.0f;
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RoomAttenuation = 1.0f;
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switch (ALContext->DistanceModel)
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{
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case AL_INVERSE_DISTANCE_CLAMPED:
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Distance=__max(Distance,MinDist);
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Distance=__min(Distance,MaxDist);
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if (MaxDist < MinDist)
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break;
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//fall-through
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case AL_INVERSE_DISTANCE:
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if (MinDist > 0.0f)
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{
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if ((MinDist + (Rolloff * (Distance - MinDist))) > 0.0f)
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flAttenuation = MinDist / (MinDist + (Rolloff * (Distance - MinDist)));
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if ((MinDist + (RoomRolloff * (Distance - MinDist))) > 0.0f)
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RoomAttenuation = MinDist / (MinDist + (RoomRolloff * (Distance - MinDist)));
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}
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break;
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case AL_LINEAR_DISTANCE_CLAMPED:
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Distance=__max(Distance,MinDist);
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Distance=__min(Distance,MaxDist);
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if (MaxDist < MinDist)
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break;
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//fall-through
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case AL_LINEAR_DISTANCE:
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Distance=__min(Distance,MaxDist);
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if (MaxDist != MinDist)
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{
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flAttenuation = 1.0f - (Rolloff*(Distance-MinDist)/(MaxDist - MinDist));
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RoomAttenuation = 1.0f - (RoomRolloff*(Distance-MinDist)/(MaxDist - MinDist));
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}
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break;
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case AL_EXPONENT_DISTANCE_CLAMPED:
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Distance=__max(Distance,MinDist);
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Distance=__min(Distance,MaxDist);
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if (MaxDist < MinDist)
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break;
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//fall-through
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case AL_EXPONENT_DISTANCE:
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if ((Distance > 0.0f) && (MinDist > 0.0f))
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{
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flAttenuation = (ALfloat)pow(Distance/MinDist, -Rolloff);
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RoomAttenuation = (ALfloat)pow(Distance/MinDist, -RoomRolloff);
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}
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break;
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case AL_NONE:
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flAttenuation = 1.0f;
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RoomAttenuation = 1.0f;
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break;
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}
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// Distance-based air absorption
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if(ALSource->AirAbsorptionFactor > 0.0f && ALContext->DistanceModel != AL_NONE)
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{
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ALfloat dist = Distance-MinDist;
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ALfloat absorb;
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if(dist < 0.0f) dist = 0.0f;
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// Absorption calculation is done in dB
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absorb = (ALSource->AirAbsorptionFactor*AIRABSORBGAINDBHF) *
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(Distance*MetersPerUnit);
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// Convert dB to linear gain before applying
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absorb = pow(0.5, absorb/-6.0);
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DryGainHF *= absorb;
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WetGainHF *= absorb;
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}
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// Source Gain + Attenuation and clamp to Min/Max Gain
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DryMix = SourceVolume * flAttenuation;
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DryMix = __min(DryMix,MaxVolume);
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DryMix = __max(DryMix,MinVolume);
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WetMix = SourceVolume * RoomAttenuation;
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WetMix = __min(WetMix,MaxVolume);
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WetMix = __max(WetMix,MinVolume);
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//3. Apply directional soundcones
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Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f /
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3.141592654f;
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if(Angle >= InnerAngle && Angle <= OuterAngle)
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{
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ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
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ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)*scale);
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DryMix *= ConeVolume;
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if(ALSource->WetGainAuto)
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WetMix *= ConeVolume;
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if(ALSource->DryGainHFAuto)
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DryGainHF *= (1.0f+(OuterGainHF-1.0f)*scale);
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if(ALSource->WetGainHFAuto)
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WetGainHF *= (1.0f+(OuterGainHF-1.0f)*scale);
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}
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else if(Angle > OuterAngle)
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{
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ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f));
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DryMix *= ConeVolume;
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if(ALSource->WetGainAuto)
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WetMix *= ConeVolume;
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if(ALSource->DryGainHFAuto)
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DryGainHF *= (1.0f+(OuterGainHF-1.0f));
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if(ALSource->WetGainHFAuto)
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WetGainHF *= (1.0f+(OuterGainHF-1.0f));
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}
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//4. Calculate Velocity
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if(DopplerFactor != 0.0f)
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{
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ALfloat flVSS, flVLS = 0.0f;
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if(ALSource->bHeadRelative==AL_FALSE)
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flVLS = aluDotproduct(ALContext->Listener.Velocity, SourceToListener);
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flVSS = aluDotproduct(ALSource->vVelocity, SourceToListener);
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flMaxVelocity = (DopplerVelocity * flSpeedOfSound) / DopplerFactor;
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if (flVSS >= flMaxVelocity)
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flVSS = (flMaxVelocity - 1.0f);
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else if (flVSS <= -flMaxVelocity)
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flVSS = -flMaxVelocity + 1.0f;
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if (flVLS >= flMaxVelocity)
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flVLS = (flMaxVelocity - 1.0f);
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else if (flVLS <= -flMaxVelocity)
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flVLS = -flMaxVelocity + 1.0f;
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pitch[0] = ALSource->flPitch *
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((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVLS)) /
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((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVSS));
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}
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else
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pitch[0] = ALSource->flPitch;
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if(ALSource->Send[0].Slot &&
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ALSource->Send[0].Slot->effect.type != AL_EFFECT_NULL)
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{
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// If the slot's auxilliary send auto is off, the data sent to the
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// effect slot is the same as the dry path, sans filter effects
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if(!ALSource->Send[0].Slot->AuxSendAuto)
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{
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WetMix = DryMix;
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WetGainHF = DryGainHF;
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}
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// Note that these are really applied by the effect slot. However,
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// it's easier to handle them here (particularly the lowpass
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// filter). Applying the gain to the individual sources going to
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// the effect slot should have the same effect as applying the gain
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// to the accumulated sources in the effect slot.
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// vol1*g + vol2*g + ... voln*g = (vol1+vol2+...voln)*g
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WetMix *= ALSource->Send[0].Slot->Gain;
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if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB)
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{
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WetMix *= ALSource->Send[0].Slot->effect.Reverb.Gain;
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WetGainHF *= ALSource->Send[0].Slot->effect.Reverb.GainHF;
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WetGainHF *= pow(ALSource->Send[0].Slot->effect.Reverb.AirAbsorptionGainHF,
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Distance * MetersPerUnit);
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}
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|
}
|
|
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;
|
|
|
|
//6. Convert normalized position into pannings, then into channel volumes
|
|
aluNormalize(Position);
|
|
switch(aluChannelsFromFormat(OutputFormat))
|
|
{
|
|
case 1:
|
|
case 2:
|
|
PanningLR = 0.5f + 0.5f*Position[0];
|
|
drysend[FRONT_LEFT] = DryMix * aluSqrt(1.0f-PanningLR); //L Direct
|
|
drysend[FRONT_RIGHT] = DryMix * aluSqrt( PanningLR); //R Direct
|
|
drysend[BACK_LEFT] = 0.0f;
|
|
drysend[BACK_RIGHT] = 0.0f;
|
|
drysend[SIDE_LEFT] = 0.0f;
|
|
drysend[SIDE_RIGHT] = 0.0f;
|
|
break;
|
|
case 4:
|
|
/* TODO: Add center/lfe channel in spatial calculations? */
|
|
case 6:
|
|
// Apply a scalar so each individual speaker has more weight
|
|
PanningLR = 0.5f + (0.5f*Position[0]*1.41421356f);
|
|
PanningLR = __min(1.0f, PanningLR);
|
|
PanningLR = __max(0.0f, PanningLR);
|
|
PanningFB = 0.5f + (0.5f*Position[2]*1.41421356f);
|
|
PanningFB = __min(1.0f, PanningFB);
|
|
PanningFB = __max(0.0f, PanningFB);
|
|
drysend[FRONT_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*(1.0f-PanningFB));
|
|
drysend[FRONT_RIGHT] = DryMix * aluSqrt(( PanningLR)*(1.0f-PanningFB));
|
|
drysend[BACK_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*( PanningFB));
|
|
drysend[BACK_RIGHT] = DryMix * aluSqrt(( PanningLR)*( PanningFB));
|
|
drysend[SIDE_LEFT] = 0.0f;
|
|
drysend[SIDE_RIGHT] = 0.0f;
|
|
break;
|
|
case 7:
|
|
case 8:
|
|
PanningFB = 1.0f - fabs(Position[2]*1.15470054f);
|
|
PanningFB = __min(1.0f, PanningFB);
|
|
PanningFB = __max(0.0f, PanningFB);
|
|
PanningLR = 0.5f + (0.5*Position[0]*((1.0f-PanningFB)*2.0f));
|
|
PanningLR = __min(1.0f, PanningLR);
|
|
PanningLR = __max(0.0f, PanningLR);
|
|
if(Position[2] > 0.0f)
|
|
{
|
|
drysend[BACK_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*(1.0f-PanningFB));
|
|
drysend[BACK_RIGHT] = DryMix * aluSqrt(( PanningLR)*(1.0f-PanningFB));
|
|
drysend[SIDE_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*( PanningFB));
|
|
drysend[SIDE_RIGHT] = DryMix * aluSqrt(( PanningLR)*( PanningFB));
|
|
drysend[FRONT_LEFT] = 0.0f;
|
|
drysend[FRONT_RIGHT] = 0.0f;
|
|
}
|
|
else
|
|
{
|
|
drysend[FRONT_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*(1.0f-PanningFB));
|
|
drysend[FRONT_RIGHT] = DryMix * aluSqrt(( PanningLR)*(1.0f-PanningFB));
|
|
drysend[SIDE_LEFT] = DryMix * aluSqrt((1.0f-PanningLR)*( PanningFB));
|
|
drysend[SIDE_RIGHT] = DryMix * aluSqrt(( PanningLR)*( PanningFB));
|
|
drysend[BACK_LEFT] = 0.0f;
|
|
drysend[BACK_RIGHT] = 0.0f;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
*wetsend = WetMix;
|
|
|
|
// Update filter coefficients. Calculations based on the I3DL2 spec.
|
|
cw = cos(2.0f*3.141592654f * 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;
|
|
|
|
drysend[FRONT_LEFT] = SourceVolume * ListenerGain;
|
|
drysend[FRONT_RIGHT] = SourceVolume * ListenerGain;
|
|
drysend[SIDE_LEFT] = SourceVolume * ListenerGain;
|
|
drysend[SIDE_RIGHT] = SourceVolume * ListenerGain;
|
|
drysend[BACK_LEFT] = SourceVolume * ListenerGain;
|
|
drysend[BACK_RIGHT] = SourceVolume * ListenerGain;
|
|
drysend[CENTER] = SourceVolume * ListenerGain;
|
|
drysend[LFE] = SourceVolume * ListenerGain;
|
|
*wetsend = 0.0f;
|
|
WetGainHF = 1.0f;
|
|
|
|
*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 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 value;
|
|
ALshort *Data;
|
|
ALuint i,j,k;
|
|
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,
|
|
format, newDrySend, &newWetSend, &Pitch,
|
|
&DryGainHF, &WetGainHF);
|
|
|
|
Pitch = (Pitch*Frequency) / ALContext->Frequency;
|
|
|
|
//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
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
dryGainStep[i] = (newDrySend[i]-DrySend[i]) / rampLength;
|
|
wetGainStep = (newWetSend-(*WetSend)) / rampLength;
|
|
|
|
//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;
|
|
while(BufferSize--)
|
|
{
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
DrySend[i] += dryGainStep[i];
|
|
*WetSend += wetGainStep;
|
|
|
|
if(Channels==1)
|
|
{
|
|
ALfloat sample, outsamp;
|
|
//First order interpolator
|
|
sample = lerp(Data[k], Data[k+1], DataPosFrac);
|
|
|
|
//Direct path final mix buffer and panning
|
|
outsamp = lpFilter(DryFilter, sample);
|
|
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];
|
|
//Room path final mix buffer and panning
|
|
outsamp = lpFilter(WetFilter, sample);
|
|
WetBuffer[j] += outsamp*(*WetSend);
|
|
}
|
|
else
|
|
{
|
|
ALfloat samp1, samp2;
|
|
//First order interpolator (front left)
|
|
samp1 = lerp(Data[k*Channels], Data[(k+1)*Channels], DataPosFrac);
|
|
DryBuffer[j][FRONT_LEFT] += samp1*DrySend[FRONT_LEFT];
|
|
//First order interpolator (front right)
|
|
samp2 = lerp(Data[k*Channels+1], Data[(k+1)*Channels+1], DataPosFrac);
|
|
DryBuffer[j][FRONT_RIGHT] += samp2*DrySend[FRONT_RIGHT];
|
|
if(Channels >= 4)
|
|
{
|
|
int i = 2;
|
|
if(Channels >= 6)
|
|
{
|
|
if(Channels != 7)
|
|
{
|
|
//First order interpolator (center)
|
|
value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac);
|
|
DryBuffer[j][CENTER] += value*DrySend[CENTER];
|
|
i++;
|
|
}
|
|
//First order interpolator (lfe)
|
|
value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac);
|
|
DryBuffer[j][LFE] += value*DrySend[LFE];
|
|
i++;
|
|
}
|
|
//First order interpolator (back left)
|
|
value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac);
|
|
DryBuffer[j][BACK_LEFT] += value*DrySend[BACK_LEFT];
|
|
i++;
|
|
//First order interpolator (back right)
|
|
value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac);
|
|
DryBuffer[j][BACK_RIGHT] += value*DrySend[BACK_RIGHT];
|
|
i++;
|
|
if(Channels >= 7)
|
|
{
|
|
//First order interpolator (side left)
|
|
value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac);
|
|
DryBuffer[j][SIDE_LEFT] += value*DrySend[SIDE_LEFT];
|
|
i++;
|
|
//First order interpolator (side right)
|
|
value = lerp(Data[k*Channels+i], Data[(k+1)*Channels+i], DataPosFrac);
|
|
DryBuffer[j][SIDE_RIGHT] += value*DrySend[SIDE_RIGHT];
|
|
i++;
|
|
}
|
|
}
|
|
else if(DuplicateStereo)
|
|
{
|
|
//Duplicate stereo channels on the back speakers
|
|
DryBuffer[j][BACK_LEFT] += samp1*DrySend[BACK_LEFT];
|
|
DryBuffer[j][BACK_RIGHT] += samp2*DrySend[BACK_RIGHT];
|
|
}
|
|
}
|
|
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;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/* alSourceRewind */
|
|
/* alSourcePlay */
|
|
ALSource->state = AL_PLAYING;
|
|
ALSource->inuse = AL_TRUE;
|
|
ALSource->play = AL_TRUE;
|
|
ALSource->BuffersPlayed = 0;
|
|
ALSource->lBytesPlayed = 0;
|
|
BufferListItem = ALSource->queue;
|
|
while(BufferListItem != NULL)
|
|
{
|
|
BufferListItem->bufferstate = PENDING;
|
|
BufferListItem = BufferListItem->next;
|
|
}
|
|
ALSource->ulBufferID = ALSource->queue->buffer;
|
|
|
|
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)
|
|
{
|
|
ALfloat *DelayBuffer = ALEffectSlot->ReverbBuffer;
|
|
ALuint Pos = ALEffectSlot->ReverbPos;
|
|
ALuint LatePos = ALEffectSlot->ReverbLatePos;
|
|
ALuint ReflectPos = ALEffectSlot->ReverbReflectPos;
|
|
ALuint Length = ALEffectSlot->ReverbLength;
|
|
ALfloat DecayGain = ALEffectSlot->ReverbDecayGain;
|
|
ALfloat DecayHFRatio = ALEffectSlot->effect.Reverb.DecayHFRatio;
|
|
ALfloat ReflectGain = ALEffectSlot->effect.Reverb.ReflectionsGain;
|
|
ALfloat LateReverbGain = ALEffectSlot->effect.Reverb.LateReverbGain;
|
|
ALfloat sample, lowsample;
|
|
|
|
WetFilter = &ALEffectSlot->iirFilter;
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
{
|
|
DelayBuffer[Pos] = WetBuffer[i];
|
|
|
|
sample = DelayBuffer[ReflectPos] * ReflectGain;
|
|
|
|
DelayBuffer[LatePos] *= LateReverbGain;
|
|
|
|
Pos = (Pos+1) % Length;
|
|
lowsample = lpFilter(WetFilter, DelayBuffer[Pos]);
|
|
lowsample += (DelayBuffer[Pos]-lowsample) * DecayHFRatio;
|
|
|
|
DelayBuffer[LatePos] += lowsample * DecayGain;
|
|
|
|
sample += DelayBuffer[LatePos];
|
|
|
|
DryBuffer[i][FRONT_LEFT] += sample;
|
|
DryBuffer[i][FRONT_RIGHT] += sample;
|
|
DryBuffer[i][SIDE_LEFT] += sample;
|
|
DryBuffer[i][SIDE_RIGHT] += sample;
|
|
DryBuffer[i][BACK_LEFT] += sample;
|
|
DryBuffer[i][BACK_RIGHT] += sample;
|
|
|
|
LatePos = (LatePos+1) % Length;
|
|
ReflectPos = (ReflectPos+1) % Length;
|
|
}
|
|
|
|
ALEffectSlot->ReverbPos = Pos;
|
|
ALEffectSlot->ReverbLatePos = LatePos;
|
|
ALEffectSlot->ReverbReflectPos = ReflectPos;
|
|
}
|
|
|
|
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][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][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);
|
|
#ifdef _WIN32
|
|
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][LFE])>>8)+128);
|
|
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT])>>8)+128);
|
|
((ALubyte*)buffer)[4] = (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][LFE])>>8)+128);
|
|
#endif
|
|
((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][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][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][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][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]);
|
|
#ifdef _WIN32
|
|
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][LFE]);
|
|
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_LEFT]);
|
|
((ALshort*)buffer)[4] = 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][LFE]);
|
|
#endif
|
|
((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][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][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);
|
|
}
|