1584 lines
56 KiB
C
1584 lines
56 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.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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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 "alSource.h"
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#include "alBuffer.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 "hrtf.h"
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#include "uhjfilter.h"
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#include "static_assert.h"
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#include "mixer_defs.h"
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#include "backends/base.h"
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struct ChanMap {
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enum Channel channel;
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ALfloat angle;
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ALfloat elevation;
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};
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/* Cone scalar */
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ALfloat ConeScale = 1.0f;
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/* Localized Z scalar for mono sources */
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ALfloat ZScale = 1.0f;
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extern inline ALfloat minf(ALfloat a, ALfloat b);
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extern inline ALfloat maxf(ALfloat a, ALfloat b);
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extern inline ALfloat clampf(ALfloat val, ALfloat min, ALfloat max);
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extern inline ALdouble mind(ALdouble a, ALdouble b);
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extern inline ALdouble maxd(ALdouble a, ALdouble b);
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extern inline ALdouble clampd(ALdouble val, ALdouble min, ALdouble max);
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extern inline ALuint minu(ALuint a, ALuint b);
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extern inline ALuint maxu(ALuint a, ALuint b);
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extern inline ALuint clampu(ALuint val, ALuint min, ALuint max);
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extern inline ALint mini(ALint a, ALint b);
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extern inline ALint maxi(ALint a, ALint b);
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extern inline ALint clampi(ALint val, ALint min, ALint max);
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extern inline ALint64 mini64(ALint64 a, ALint64 b);
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extern inline ALint64 maxi64(ALint64 a, ALint64 b);
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extern inline ALint64 clampi64(ALint64 val, ALint64 min, ALint64 max);
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extern inline ALuint64 minu64(ALuint64 a, ALuint64 b);
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extern inline ALuint64 maxu64(ALuint64 a, ALuint64 b);
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extern inline ALuint64 clampu64(ALuint64 val, ALuint64 min, ALuint64 max);
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extern inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu);
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extern inline ALfloat resample_fir4(ALfloat val0, ALfloat val1, ALfloat val2, ALfloat val3, ALuint frac);
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extern inline ALfloat resample_fir8(ALfloat val0, ALfloat val1, ALfloat val2, ALfloat val3, ALfloat val4, ALfloat val5, ALfloat val6, ALfloat val7, ALuint frac);
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extern inline void aluVectorSet(aluVector *restrict vector, ALfloat x, ALfloat y, ALfloat z, ALfloat w);
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extern inline void aluMatrixfSetRow(aluMatrixf *matrix, ALuint row,
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ALfloat m0, ALfloat m1, ALfloat m2, ALfloat m3);
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extern inline void aluMatrixfSet(aluMatrixf *matrix,
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ALfloat m00, ALfloat m01, ALfloat m02, ALfloat m03,
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ALfloat m10, ALfloat m11, ALfloat m12, ALfloat m13,
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ALfloat m20, ALfloat m21, ALfloat m22, ALfloat m23,
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ALfloat m30, ALfloat m31, ALfloat m32, ALfloat m33);
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extern inline void aluMatrixdSetRow(aluMatrixd *matrix, ALuint row,
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ALdouble m0, ALdouble m1, ALdouble m2, ALdouble m3);
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extern inline void aluMatrixdSet(aluMatrixd *matrix,
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ALdouble m00, ALdouble m01, ALdouble m02, ALdouble m03,
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ALdouble m10, ALdouble m11, ALdouble m12, ALdouble m13,
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ALdouble m20, ALdouble m21, ALdouble m22, ALdouble m23,
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ALdouble m30, ALdouble m31, ALdouble m32, ALdouble m33);
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static inline HrtfMixerFunc SelectHrtfMixer(void)
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{
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return MixHrtf_SSE;
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#endif
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return MixHrtf_Neon;
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#endif
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return MixHrtf_C;
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}
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static inline void aluCrossproduct(const ALfloat *inVector1, const 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(const aluVector *vec1, const aluVector *vec2)
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{
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return vec1->v[0]*vec2->v[0] + vec1->v[1]*vec2->v[1] + vec1->v[2]*vec2->v[2];
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}
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static inline ALfloat aluNormalize(ALfloat *vec)
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{
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ALfloat length = sqrtf(vec[0]*vec[0] + vec[1]*vec[1] + vec[2]*vec[2]);
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if(length > 0.0f)
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{
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ALfloat inv_length = 1.0f/length;
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vec[0] *= inv_length;
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vec[1] *= inv_length;
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vec[2] *= inv_length;
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}
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return length;
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}
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static inline void aluCrossproductd(const ALdouble *inVector1, const ALdouble *inVector2, ALdouble *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 ALdouble aluNormalized(ALdouble *vec)
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{
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ALdouble length = sqrt(vec[0]*vec[0] + vec[1]*vec[1] + vec[2]*vec[2]);
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if(length > 0.0)
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{
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ALdouble inv_length = 1.0/length;
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vec[0] *= inv_length;
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vec[1] *= inv_length;
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vec[2] *= inv_length;
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}
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return length;
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}
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static inline ALvoid aluMatrixdFloat3(ALfloat *vec, ALfloat w, const aluMatrixd *mtx)
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{
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ALdouble v[4] = { vec[0], vec[1], vec[2], w };
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vec[0] = (ALfloat)(v[0]*mtx->m[0][0] + v[1]*mtx->m[1][0] + v[2]*mtx->m[2][0] + v[3]*mtx->m[3][0]);
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vec[1] = (ALfloat)(v[0]*mtx->m[0][1] + v[1]*mtx->m[1][1] + v[2]*mtx->m[2][1] + v[3]*mtx->m[3][1]);
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vec[2] = (ALfloat)(v[0]*mtx->m[0][2] + v[1]*mtx->m[1][2] + v[2]*mtx->m[2][2] + v[3]*mtx->m[3][2]);
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}
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static inline ALvoid aluMatrixdDouble3(ALdouble *vec, ALdouble w, const aluMatrixd *mtx)
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{
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ALdouble v[4] = { vec[0], vec[1], vec[2], w };
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vec[0] = v[0]*mtx->m[0][0] + v[1]*mtx->m[1][0] + v[2]*mtx->m[2][0] + v[3]*mtx->m[3][0];
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vec[1] = v[0]*mtx->m[0][1] + v[1]*mtx->m[1][1] + v[2]*mtx->m[2][1] + v[3]*mtx->m[3][1];
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vec[2] = v[0]*mtx->m[0][2] + v[1]*mtx->m[1][2] + v[2]*mtx->m[2][2] + v[3]*mtx->m[3][2];
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}
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static inline aluVector aluMatrixdVector(const aluMatrixd *mtx, const aluVector *vec)
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{
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aluVector v;
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v.v[0] = (ALfloat)(vec->v[0]*mtx->m[0][0] + vec->v[1]*mtx->m[1][0] + vec->v[2]*mtx->m[2][0] + vec->v[3]*mtx->m[3][0]);
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v.v[1] = (ALfloat)(vec->v[0]*mtx->m[0][1] + vec->v[1]*mtx->m[1][1] + vec->v[2]*mtx->m[2][1] + vec->v[3]*mtx->m[3][1]);
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v.v[2] = (ALfloat)(vec->v[0]*mtx->m[0][2] + vec->v[1]*mtx->m[1][2] + vec->v[2]*mtx->m[2][2] + vec->v[3]*mtx->m[3][2]);
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v.v[3] = (ALfloat)(vec->v[0]*mtx->m[0][3] + vec->v[1]*mtx->m[1][3] + vec->v[2]*mtx->m[2][3] + vec->v[3]*mtx->m[3][3]);
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return v;
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}
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/* Prepares the interpolator for a given rate (determined by increment). A
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* result of AL_FALSE indicates that the filter output will completely cut
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* the input signal.
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*
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* With a bit of work, and a trade of memory for CPU cost, this could be
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* modified for use with an interpolated increment for buttery-smooth pitch
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* changes.
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*/
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static ALboolean BsincPrepare(const ALuint increment, BsincState *state)
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{
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static const ALfloat scaleBase = 1.510578918e-01f, scaleRange = 1.177936623e+00f;
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static const ALuint m[BSINC_SCALE_COUNT] = { 24, 24, 24, 24, 24, 24, 24, 20, 20, 20, 16, 16, 16, 12, 12, 12 };
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static const ALuint to[4][BSINC_SCALE_COUNT] =
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{
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{ 0, 24, 408, 792, 1176, 1560, 1944, 2328, 2648, 2968, 3288, 3544, 3800, 4056, 4248, 4440 },
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{ 4632, 5016, 5400, 5784, 6168, 6552, 6936, 7320, 7640, 7960, 8280, 8536, 8792, 9048, 9240, 0 },
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{ 0, 9432, 9816, 10200, 10584, 10968, 11352, 11736, 12056, 12376, 12696, 12952, 13208, 13464, 13656, 13848 },
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{ 14040, 14424, 14808, 15192, 15576, 15960, 16344, 16728, 17048, 17368, 17688, 17944, 18200, 18456, 18648, 0 }
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};
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static const ALuint tm[2][BSINC_SCALE_COUNT] =
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{
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{ 0, 24, 24, 24, 24, 24, 24, 20, 20, 20, 16, 16, 16, 12, 12, 12 },
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{ 24, 24, 24, 24, 24, 24, 24, 20, 20, 20, 16, 16, 16, 12, 12, 0 }
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};
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ALfloat sf;
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ALuint si, pi;
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ALboolean uncut = AL_TRUE;
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if(increment > FRACTIONONE)
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{
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sf = (ALfloat)FRACTIONONE / increment;
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if(sf < scaleBase)
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{
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/* Signal has been completely cut. The return result can be used
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* to skip the filter (and output zeros) as an optimization.
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*/
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sf = 0.0f;
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si = 0;
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uncut = AL_FALSE;
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}
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else
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{
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sf = (BSINC_SCALE_COUNT - 1) * (sf - scaleBase) * scaleRange;
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si = fastf2u(sf);
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/* The interpolation factor is fit to this diagonally-symmetric
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* curve to reduce the transition ripple caused by interpolating
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* different scales of the sinc function.
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*/
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sf = 1.0f - cosf(asinf(sf - si));
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}
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}
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else
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{
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sf = 0.0f;
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si = BSINC_SCALE_COUNT - 1;
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}
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state->sf = sf;
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state->m = m[si];
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state->l = -(ALint)((m[si] / 2) - 1);
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/* The CPU cost of this table re-mapping could be traded for the memory
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* cost of a complete table map (1024 elements large).
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*/
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for(pi = 0;pi < BSINC_PHASE_COUNT;pi++)
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{
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state->coeffs[pi].filter = &bsincTab[to[0][si] + tm[0][si]*pi];
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state->coeffs[pi].scDelta = &bsincTab[to[1][si] + tm[1][si]*pi];
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state->coeffs[pi].phDelta = &bsincTab[to[2][si] + tm[0][si]*pi];
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state->coeffs[pi].spDelta = &bsincTab[to[3][si] + tm[1][si]*pi];
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}
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return uncut;
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}
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static ALvoid CalcListenerParams(ALlistener *Listener)
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{
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ALdouble N[3], V[3], U[3], P[3];
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/* AT then UP */
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N[0] = Listener->Forward[0];
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N[1] = Listener->Forward[1];
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N[2] = Listener->Forward[2];
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aluNormalized(N);
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V[0] = Listener->Up[0];
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V[1] = Listener->Up[1];
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V[2] = Listener->Up[2];
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aluNormalized(V);
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/* Build and normalize right-vector */
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aluCrossproductd(N, V, U);
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aluNormalized(U);
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aluMatrixdSet(&Listener->Params.Matrix,
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U[0], V[0], -N[0], 0.0,
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U[1], V[1], -N[1], 0.0,
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U[2], V[2], -N[2], 0.0,
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0.0, 0.0, 0.0, 1.0
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);
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P[0] = Listener->Position.v[0];
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P[1] = Listener->Position.v[1];
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P[2] = Listener->Position.v[2];
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aluMatrixdDouble3(P, 1.0, &Listener->Params.Matrix);
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aluMatrixdSetRow(&Listener->Params.Matrix, 3, -P[0], -P[1], -P[2], 1.0f);
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Listener->Params.Velocity = aluMatrixdVector(&Listener->Params.Matrix, &Listener->Velocity);
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}
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ALvoid CalcNonAttnSourceParams(ALvoice *voice, const ALsource *ALSource, const ALCcontext *ALContext)
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{
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static const struct ChanMap MonoMap[1] = {
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{ FrontCenter, 0.0f, 0.0f }
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}, StereoMap[2] = {
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{ FrontLeft, DEG2RAD(-30.0f), DEG2RAD(0.0f) },
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{ FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) }
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}, RearMap[2] = {
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{ BackLeft, DEG2RAD(-150.0f), DEG2RAD(0.0f) },
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{ BackRight, DEG2RAD( 150.0f), DEG2RAD(0.0f) }
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}, QuadMap[4] = {
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{ FrontLeft, DEG2RAD( -45.0f), DEG2RAD(0.0f) },
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{ FrontRight, DEG2RAD( 45.0f), DEG2RAD(0.0f) },
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{ BackLeft, DEG2RAD(-135.0f), DEG2RAD(0.0f) },
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{ BackRight, DEG2RAD( 135.0f), DEG2RAD(0.0f) }
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}, X51Map[6] = {
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{ FrontLeft, DEG2RAD( -30.0f), DEG2RAD(0.0f) },
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{ FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) },
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{ FrontCenter, DEG2RAD( 0.0f), DEG2RAD(0.0f) },
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{ LFE, 0.0f, 0.0f },
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{ SideLeft, DEG2RAD(-110.0f), DEG2RAD(0.0f) },
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{ SideRight, DEG2RAD( 110.0f), DEG2RAD(0.0f) }
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}, X61Map[7] = {
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{ FrontLeft, DEG2RAD(-30.0f), DEG2RAD(0.0f) },
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{ FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) },
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{ FrontCenter, DEG2RAD( 0.0f), DEG2RAD(0.0f) },
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{ LFE, 0.0f, 0.0f },
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{ BackCenter, DEG2RAD(180.0f), DEG2RAD(0.0f) },
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{ SideLeft, DEG2RAD(-90.0f), DEG2RAD(0.0f) },
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{ SideRight, DEG2RAD( 90.0f), DEG2RAD(0.0f) }
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}, X71Map[8] = {
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{ FrontLeft, DEG2RAD( -30.0f), DEG2RAD(0.0f) },
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{ FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) },
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{ FrontCenter, DEG2RAD( 0.0f), DEG2RAD(0.0f) },
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{ LFE, 0.0f, 0.0f },
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{ BackLeft, DEG2RAD(-150.0f), DEG2RAD(0.0f) },
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{ BackRight, DEG2RAD( 150.0f), DEG2RAD(0.0f) },
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{ SideLeft, DEG2RAD( -90.0f), DEG2RAD(0.0f) },
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{ SideRight, DEG2RAD( 90.0f), DEG2RAD(0.0f) }
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};
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const ALCdevice *Device = ALContext->Device;
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ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
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ALbufferlistitem *BufferListItem;
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enum FmtChannels Channels;
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ALfloat DryGain, DryGainHF, DryGainLF;
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ALfloat WetGain[MAX_SENDS];
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ALfloat WetGainHF[MAX_SENDS];
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ALfloat WetGainLF[MAX_SENDS];
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ALeffectslot *SendSlots[MAX_SENDS];
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ALuint NumSends, Frequency;
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ALboolean Relative;
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const struct ChanMap *chans = NULL;
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ALuint num_channels = 0;
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ALboolean DirectChannels;
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ALboolean isbformat = AL_FALSE;
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ALfloat Pitch;
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ALuint i, j, c;
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/* Get device properties */
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NumSends = Device->NumAuxSends;
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Frequency = Device->Frequency;
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/* Get listener properties */
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ListenerGain = ALContext->Listener->Gain;
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/* Get source properties */
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SourceVolume = ALSource->Gain;
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MinVolume = ALSource->MinGain;
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MaxVolume = ALSource->MaxGain;
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Pitch = ALSource->Pitch;
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Relative = ALSource->HeadRelative;
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DirectChannels = ALSource->DirectChannels;
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voice->Direct.OutBuffer = Device->Dry.Buffer;
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voice->Direct.OutChannels = Device->Dry.NumChannels;
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for(i = 0;i < NumSends;i++)
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{
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SendSlots[i] = ALSource->Send[i].Slot;
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if(!SendSlots[i] && i == 0)
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SendSlots[i] = Device->DefaultSlot;
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if(!SendSlots[i] || SendSlots[i]->EffectType == AL_EFFECT_NULL)
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{
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SendSlots[i] = NULL;
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voice->Send[i].OutBuffer = NULL;
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voice->Send[i].OutChannels = 0;
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}
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else
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{
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voice->Send[i].OutBuffer = SendSlots[i]->WetBuffer;
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voice->Send[i].OutChannels = SendSlots[i]->NumChannels;
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}
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}
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/* Calculate the stepping value */
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Channels = FmtMono;
|
|
BufferListItem = ATOMIC_LOAD(&ALSource->queue);
|
|
while(BufferListItem != NULL)
|
|
{
|
|
ALbuffer *ALBuffer;
|
|
if((ALBuffer=BufferListItem->buffer) != NULL)
|
|
{
|
|
Pitch = Pitch * ALBuffer->Frequency / Frequency;
|
|
if(Pitch > (ALfloat)MAX_PITCH)
|
|
voice->Step = MAX_PITCH<<FRACTIONBITS;
|
|
else
|
|
voice->Step = maxi(fastf2i(Pitch*FRACTIONONE + 0.5f), 1);
|
|
BsincPrepare(voice->Step, &voice->SincState);
|
|
|
|
Channels = ALBuffer->FmtChannels;
|
|
break;
|
|
}
|
|
BufferListItem = BufferListItem->next;
|
|
}
|
|
|
|
/* Calculate gains */
|
|
DryGain = clampf(SourceVolume, MinVolume, MaxVolume);
|
|
DryGain *= ALSource->Direct.Gain * ListenerGain;
|
|
DryGainHF = ALSource->Direct.GainHF;
|
|
DryGainLF = ALSource->Direct.GainLF;
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
WetGain[i] = clampf(SourceVolume, MinVolume, MaxVolume);
|
|
WetGain[i] *= ALSource->Send[i].Gain * ListenerGain;
|
|
WetGainHF[i] = ALSource->Send[i].GainHF;
|
|
WetGainLF[i] = ALSource->Send[i].GainLF;
|
|
}
|
|
|
|
switch(Channels)
|
|
{
|
|
case FmtMono:
|
|
chans = MonoMap;
|
|
num_channels = 1;
|
|
break;
|
|
|
|
case FmtStereo:
|
|
chans = StereoMap;
|
|
num_channels = 2;
|
|
break;
|
|
|
|
case FmtRear:
|
|
chans = RearMap;
|
|
num_channels = 2;
|
|
break;
|
|
|
|
case FmtQuad:
|
|
chans = QuadMap;
|
|
num_channels = 4;
|
|
break;
|
|
|
|
case FmtX51:
|
|
chans = X51Map;
|
|
num_channels = 6;
|
|
break;
|
|
|
|
case FmtX61:
|
|
chans = X61Map;
|
|
num_channels = 7;
|
|
break;
|
|
|
|
case FmtX71:
|
|
chans = X71Map;
|
|
num_channels = 8;
|
|
break;
|
|
|
|
case FmtBFormat2D:
|
|
num_channels = 3;
|
|
isbformat = AL_TRUE;
|
|
DirectChannels = AL_FALSE;
|
|
break;
|
|
|
|
case FmtBFormat3D:
|
|
num_channels = 4;
|
|
isbformat = AL_TRUE;
|
|
DirectChannels = AL_FALSE;
|
|
break;
|
|
}
|
|
|
|
if(isbformat)
|
|
{
|
|
ALfloat N[3], V[3], U[3];
|
|
aluMatrixf matrix;
|
|
ALfloat scale;
|
|
|
|
/* AT then UP */
|
|
N[0] = ALSource->Orientation[0][0];
|
|
N[1] = ALSource->Orientation[0][1];
|
|
N[2] = ALSource->Orientation[0][2];
|
|
aluNormalize(N);
|
|
V[0] = ALSource->Orientation[1][0];
|
|
V[1] = ALSource->Orientation[1][1];
|
|
V[2] = ALSource->Orientation[1][2];
|
|
aluNormalize(V);
|
|
if(!Relative)
|
|
{
|
|
const aluMatrixd *lmatrix = &ALContext->Listener->Params.Matrix;
|
|
aluMatrixdFloat3(N, 0.0f, lmatrix);
|
|
aluMatrixdFloat3(V, 0.0f, lmatrix);
|
|
}
|
|
/* Build and normalize right-vector */
|
|
aluCrossproduct(N, V, U);
|
|
aluNormalize(U);
|
|
|
|
/* Build a rotate + conversion matrix (B-Format -> N3D), and include
|
|
* scaling for first-order content on second- or third-order output.
|
|
*/
|
|
scale = Device->Dry.AmbiScale * 1.732050808f;
|
|
aluMatrixfSet(&matrix,
|
|
1.414213562f, 0.0f, 0.0f, 0.0f,
|
|
0.0f, -N[0]*scale, N[1]*scale, -N[2]*scale,
|
|
0.0f, U[0]*scale, -U[1]*scale, U[2]*scale,
|
|
0.0f, -V[0]*scale, V[1]*scale, -V[2]*scale
|
|
);
|
|
|
|
for(c = 0;c < num_channels;c++)
|
|
ComputeFirstOrderGains(Device->Dry.AmbiCoeffs, Device->Dry.NumChannels, matrix.m[c],
|
|
DryGain, voice->Direct.Gains[c].Target);
|
|
|
|
/* Rebuild the matrix, without the second- or third-order output
|
|
* scaling (effects take first-order content, and will do the scaling
|
|
* themselves when mixing to the output).
|
|
*/
|
|
scale = 1.732050808f;
|
|
aluMatrixfSetRow(&matrix, 1, 0.0f, -N[0]*scale, N[1]*scale, -N[2]*scale);
|
|
aluMatrixfSetRow(&matrix, 2, 0.0f, U[0]*scale, -U[1]*scale, U[2]*scale);
|
|
aluMatrixfSetRow(&matrix, 3, 0.0f, -V[0]*scale, V[1]*scale, -V[2]*scale);
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
if(!SendSlots[i])
|
|
{
|
|
for(c = 0;c < num_channels;c++)
|
|
{
|
|
for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
|
|
voice->Send[i].Gains[c].Target[j] = 0.0f;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
for(c = 0;c < num_channels;c++)
|
|
{
|
|
const ALeffectslot *Slot = SendSlots[i];
|
|
ComputeFirstOrderGains(Slot->AmbiCoeffs, Slot->NumChannels, matrix.m[c],
|
|
WetGain[i], voice->Send[i].Gains[c].Target);
|
|
}
|
|
}
|
|
}
|
|
|
|
voice->IsHrtf = AL_FALSE;
|
|
}
|
|
else
|
|
{
|
|
ALfloat coeffs[MAX_AMBI_COEFFS];
|
|
|
|
if(DirectChannels)
|
|
{
|
|
if(Device->Hrtf || Device->Uhj_Encoder)
|
|
{
|
|
/* DirectChannels with HRTF or UHJ enabled. Skip the virtual
|
|
* channels and write FrontLeft and FrontRight inputs to the
|
|
* first and second outputs.
|
|
*/
|
|
voice->Direct.OutBuffer = Device->RealOut.Buffer;
|
|
voice->Direct.OutChannels = Device->RealOut.NumChannels;
|
|
for(c = 0;c < num_channels;c++)
|
|
{
|
|
int idx;
|
|
for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
|
|
voice->Direct.Gains[c].Target[j] = 0.0f;
|
|
if((idx=GetChannelIdxByName(Device->RealOut, chans[c].channel)) != -1)
|
|
voice->Direct.Gains[c].Target[idx] = DryGain;
|
|
}
|
|
}
|
|
else for(c = 0;c < num_channels;c++)
|
|
{
|
|
int idx;
|
|
for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
|
|
voice->Direct.Gains[c].Target[j] = 0.0f;
|
|
if((idx=GetChannelIdxByName(Device->Dry, chans[c].channel)) != -1)
|
|
voice->Direct.Gains[c].Target[idx] = DryGain;
|
|
}
|
|
|
|
/* Auxiliary sends still use normal panning since they mix to B-Format, which can't
|
|
* channel-match. */
|
|
for(c = 0;c < num_channels;c++)
|
|
{
|
|
CalcAngleCoeffs(chans[c].angle, chans[c].elevation, coeffs);
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
if(!SendSlots[i])
|
|
{
|
|
for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
|
|
voice->Send[i].Gains[c].Target[j] = 0.0f;
|
|
}
|
|
else
|
|
{
|
|
const ALeffectslot *Slot = SendSlots[i];
|
|
ComputePanningGains(Slot->AmbiCoeffs, Slot->NumChannels, coeffs,
|
|
WetGain[i], voice->Send[i].Gains[c].Target);
|
|
}
|
|
}
|
|
}
|
|
|
|
voice->IsHrtf = AL_FALSE;
|
|
}
|
|
else if(Device->Render_Mode == HrtfRender)
|
|
{
|
|
/* Full HRTF rendering. Skip the virtual channels and render each
|
|
* input channel to the real outputs.
|
|
*/
|
|
voice->Direct.OutBuffer = Device->RealOut.Buffer;
|
|
voice->Direct.OutChannels = Device->RealOut.NumChannels;
|
|
for(c = 0;c < num_channels;c++)
|
|
{
|
|
if(chans[c].channel == LFE)
|
|
{
|
|
/* Skip LFE */
|
|
voice->Direct.Hrtf[c].Target.Delay[0] = 0;
|
|
voice->Direct.Hrtf[c].Target.Delay[1] = 0;
|
|
for(i = 0;i < HRIR_LENGTH;i++)
|
|
{
|
|
voice->Direct.Hrtf[c].Target.Coeffs[i][0] = 0.0f;
|
|
voice->Direct.Hrtf[c].Target.Coeffs[i][1] = 0.0f;
|
|
}
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
|
|
voice->Send[i].Gains[c].Target[j] = 0.0f;
|
|
}
|
|
|
|
continue;
|
|
}
|
|
|
|
/* Get the static HRIR coefficients and delays for this channel. */
|
|
GetLerpedHrtfCoeffs(Device->Hrtf,
|
|
chans[c].elevation, chans[c].angle, 1.0f, DryGain,
|
|
voice->Direct.Hrtf[c].Target.Coeffs,
|
|
voice->Direct.Hrtf[c].Target.Delay
|
|
);
|
|
|
|
/* Normal panning for auxiliary sends. */
|
|
CalcAngleCoeffs(chans[c].angle, chans[c].elevation, coeffs);
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
if(!SendSlots[i])
|
|
{
|
|
for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
|
|
voice->Send[i].Gains[c].Target[j] = 0.0f;
|
|
}
|
|
else
|
|
{
|
|
const ALeffectslot *Slot = SendSlots[i];
|
|
ComputePanningGains(Slot->AmbiCoeffs, Slot->NumChannels, coeffs,
|
|
WetGain[i], voice->Send[i].Gains[c].Target);
|
|
}
|
|
}
|
|
}
|
|
|
|
voice->IsHrtf = AL_TRUE;
|
|
}
|
|
else
|
|
{
|
|
/* Non-HRTF rendering. Use normal panning to the output. */
|
|
for(c = 0;c < num_channels;c++)
|
|
{
|
|
/* Special-case LFE */
|
|
if(chans[c].channel == LFE)
|
|
{
|
|
int idx;
|
|
for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
|
|
voice->Direct.Gains[c].Target[j] = 0.0f;
|
|
if((idx=GetChannelIdxByName(Device->Dry, chans[c].channel)) != -1)
|
|
voice->Direct.Gains[c].Target[idx] = DryGain;
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
ALuint j;
|
|
for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
|
|
voice->Send[i].Gains[c].Target[j] = 0.0f;
|
|
}
|
|
continue;
|
|
}
|
|
|
|
if(Device->Render_Mode == StereoPair)
|
|
{
|
|
/* Clamp X so it remains within 30 degrees of 0 or 180 degree azimuth. */
|
|
ALfloat x = sinf(chans[c].angle) * cosf(chans[c].elevation);
|
|
coeffs[0] = clampf(-x, -0.5f, 0.5f) + 0.5;
|
|
voice->Direct.Gains[c].Target[0] = coeffs[0] * DryGain;
|
|
voice->Direct.Gains[c].Target[1] = (1.0f-coeffs[0]) * DryGain;
|
|
for(j = 2;j < MAX_OUTPUT_CHANNELS;j++)
|
|
voice->Direct.Gains[c].Target[j] = 0.0f;
|
|
|
|
CalcAngleCoeffs(chans[c].angle, chans[c].elevation, coeffs);
|
|
}
|
|
else
|
|
{
|
|
CalcAngleCoeffs(chans[c].angle, chans[c].elevation, coeffs);
|
|
ComputePanningGains(Device->Dry.AmbiCoeffs, Device->Dry.NumChannels, coeffs,
|
|
DryGain, voice->Direct.Gains[c].Target);
|
|
}
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
if(!SendSlots[i])
|
|
{
|
|
ALuint j;
|
|
for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
|
|
voice->Send[i].Gains[c].Target[j] = 0.0f;
|
|
}
|
|
else
|
|
{
|
|
const ALeffectslot *Slot = SendSlots[i];
|
|
ComputePanningGains(Slot->AmbiCoeffs, Slot->NumChannels, coeffs,
|
|
WetGain[i], voice->Send[i].Gains[c].Target);
|
|
}
|
|
}
|
|
}
|
|
|
|
voice->IsHrtf = AL_FALSE;
|
|
}
|
|
}
|
|
|
|
{
|
|
ALfloat hfscale = ALSource->Direct.HFReference / Frequency;
|
|
ALfloat lfscale = ALSource->Direct.LFReference / Frequency;
|
|
DryGainHF = maxf(DryGainHF, 0.0001f);
|
|
DryGainLF = maxf(DryGainLF, 0.0001f);
|
|
for(c = 0;c < num_channels;c++)
|
|
{
|
|
voice->Direct.Filters[c].ActiveType = AF_None;
|
|
if(DryGainHF != 1.0f) voice->Direct.Filters[c].ActiveType |= AF_LowPass;
|
|
if(DryGainLF != 1.0f) voice->Direct.Filters[c].ActiveType |= AF_HighPass;
|
|
ALfilterState_setParams(
|
|
&voice->Direct.Filters[c].LowPass, ALfilterType_HighShelf,
|
|
DryGainHF, hfscale, calc_rcpQ_from_slope(DryGainHF, 0.75f)
|
|
);
|
|
ALfilterState_setParams(
|
|
&voice->Direct.Filters[c].HighPass, ALfilterType_LowShelf,
|
|
DryGainLF, lfscale, calc_rcpQ_from_slope(DryGainLF, 0.75f)
|
|
);
|
|
}
|
|
}
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
ALfloat hfscale = ALSource->Send[i].HFReference / Frequency;
|
|
ALfloat lfscale = ALSource->Send[i].LFReference / Frequency;
|
|
WetGainHF[i] = maxf(WetGainHF[i], 0.0001f);
|
|
WetGainLF[i] = maxf(WetGainLF[i], 0.0001f);
|
|
for(c = 0;c < num_channels;c++)
|
|
{
|
|
voice->Send[i].Filters[c].ActiveType = AF_None;
|
|
if(WetGainHF[i] != 1.0f) voice->Send[i].Filters[c].ActiveType |= AF_LowPass;
|
|
if(WetGainLF[i] != 1.0f) voice->Send[i].Filters[c].ActiveType |= AF_HighPass;
|
|
ALfilterState_setParams(
|
|
&voice->Send[i].Filters[c].LowPass, ALfilterType_HighShelf,
|
|
WetGainHF[i], hfscale, calc_rcpQ_from_slope(WetGainHF[i], 0.75f)
|
|
);
|
|
ALfilterState_setParams(
|
|
&voice->Send[i].Filters[c].HighPass, ALfilterType_LowShelf,
|
|
WetGainLF[i], lfscale, calc_rcpQ_from_slope(WetGainLF[i], 0.75f)
|
|
);
|
|
}
|
|
}
|
|
}
|
|
|
|
ALvoid CalcSourceParams(ALvoice *voice, const ALsource *ALSource, const ALCcontext *ALContext)
|
|
{
|
|
const ALCdevice *Device = ALContext->Device;
|
|
aluVector Position, Velocity, Direction, SourceToListener;
|
|
ALfloat InnerAngle,OuterAngle,Angle,Distance,ClampedDist;
|
|
ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff;
|
|
ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
|
|
ALfloat DopplerFactor, SpeedOfSound;
|
|
ALfloat AirAbsorptionFactor;
|
|
ALfloat RoomAirAbsorption[MAX_SENDS];
|
|
ALbufferlistitem *BufferListItem;
|
|
ALeffectslot *SendSlots[MAX_SENDS];
|
|
ALfloat Attenuation;
|
|
ALfloat RoomAttenuation[MAX_SENDS];
|
|
ALfloat MetersPerUnit;
|
|
ALfloat RoomRolloffBase;
|
|
ALfloat RoomRolloff[MAX_SENDS];
|
|
ALfloat DecayDistance[MAX_SENDS];
|
|
ALfloat DryGain;
|
|
ALfloat DryGainHF;
|
|
ALfloat DryGainLF;
|
|
ALboolean DryGainHFAuto;
|
|
ALfloat WetGain[MAX_SENDS];
|
|
ALfloat WetGainHF[MAX_SENDS];
|
|
ALfloat WetGainLF[MAX_SENDS];
|
|
ALboolean WetGainAuto;
|
|
ALboolean WetGainHFAuto;
|
|
ALfloat Pitch;
|
|
ALuint Frequency;
|
|
ALint NumSends;
|
|
ALint i;
|
|
|
|
DryGainHF = 1.0f;
|
|
DryGainLF = 1.0f;
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
{
|
|
WetGainHF[i] = 1.0f;
|
|
WetGainLF[i] = 1.0f;
|
|
}
|
|
|
|
/* Get context/device properties */
|
|
DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
|
|
SpeedOfSound = ALContext->SpeedOfSound * ALContext->DopplerVelocity;
|
|
NumSends = Device->NumAuxSends;
|
|
Frequency = Device->Frequency;
|
|
|
|
/* Get listener properties */
|
|
ListenerGain = ALContext->Listener->Gain;
|
|
MetersPerUnit = ALContext->Listener->MetersPerUnit;
|
|
|
|
/* Get source properties */
|
|
SourceVolume = ALSource->Gain;
|
|
MinVolume = ALSource->MinGain;
|
|
MaxVolume = ALSource->MaxGain;
|
|
Pitch = ALSource->Pitch;
|
|
Position = ALSource->Position;
|
|
Direction = ALSource->Direction;
|
|
Velocity = ALSource->Velocity;
|
|
MinDist = ALSource->RefDistance;
|
|
MaxDist = ALSource->MaxDistance;
|
|
Rolloff = ALSource->RollOffFactor;
|
|
InnerAngle = ALSource->InnerAngle;
|
|
OuterAngle = ALSource->OuterAngle;
|
|
AirAbsorptionFactor = ALSource->AirAbsorptionFactor;
|
|
DryGainHFAuto = ALSource->DryGainHFAuto;
|
|
WetGainAuto = ALSource->WetGainAuto;
|
|
WetGainHFAuto = ALSource->WetGainHFAuto;
|
|
RoomRolloffBase = ALSource->RoomRolloffFactor;
|
|
|
|
voice->Direct.OutBuffer = Device->Dry.Buffer;
|
|
voice->Direct.OutChannels = Device->Dry.NumChannels;
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
SendSlots[i] = ALSource->Send[i].Slot;
|
|
|
|
if(!SendSlots[i] && i == 0)
|
|
SendSlots[i] = Device->DefaultSlot;
|
|
if(!SendSlots[i] || SendSlots[i]->EffectType == AL_EFFECT_NULL)
|
|
{
|
|
SendSlots[i] = NULL;
|
|
RoomRolloff[i] = 0.0f;
|
|
DecayDistance[i] = 0.0f;
|
|
RoomAirAbsorption[i] = 1.0f;
|
|
}
|
|
else if(SendSlots[i]->AuxSendAuto)
|
|
{
|
|
RoomRolloff[i] = RoomRolloffBase;
|
|
if(IsReverbEffect(SendSlots[i]->EffectType))
|
|
{
|
|
RoomRolloff[i] += SendSlots[i]->EffectProps.Reverb.RoomRolloffFactor;
|
|
DecayDistance[i] = SendSlots[i]->EffectProps.Reverb.DecayTime *
|
|
SPEEDOFSOUNDMETRESPERSEC;
|
|
RoomAirAbsorption[i] = SendSlots[i]->EffectProps.Reverb.AirAbsorptionGainHF;
|
|
}
|
|
else
|
|
{
|
|
DecayDistance[i] = 0.0f;
|
|
RoomAirAbsorption[i] = 1.0f;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/* If the slot's auxiliary send auto is off, the data sent to the
|
|
* effect slot is the same as the dry path, sans filter effects */
|
|
RoomRolloff[i] = Rolloff;
|
|
DecayDistance[i] = 0.0f;
|
|
RoomAirAbsorption[i] = AIRABSORBGAINHF;
|
|
}
|
|
|
|
if(!SendSlots[i])
|
|
{
|
|
voice->Send[i].OutBuffer = NULL;
|
|
voice->Send[i].OutChannels = 0;
|
|
}
|
|
else
|
|
{
|
|
voice->Send[i].OutBuffer = SendSlots[i]->WetBuffer;
|
|
voice->Send[i].OutChannels = SendSlots[i]->NumChannels;
|
|
}
|
|
}
|
|
|
|
/* Transform source to listener space (convert to head relative) */
|
|
if(ALSource->HeadRelative == AL_FALSE)
|
|
{
|
|
const aluMatrixd *Matrix = &ALContext->Listener->Params.Matrix;
|
|
/* Transform source vectors */
|
|
Position = aluMatrixdVector(Matrix, &Position);
|
|
Velocity = aluMatrixdVector(Matrix, &Velocity);
|
|
Direction = aluMatrixdVector(Matrix, &Direction);
|
|
}
|
|
else
|
|
{
|
|
const aluVector *lvelocity = &ALContext->Listener->Params.Velocity;
|
|
/* Offset the source velocity to be relative of the listener velocity */
|
|
Velocity.v[0] += lvelocity->v[0];
|
|
Velocity.v[1] += lvelocity->v[1];
|
|
Velocity.v[2] += lvelocity->v[2];
|
|
}
|
|
|
|
aluNormalize(Direction.v);
|
|
SourceToListener.v[0] = -Position.v[0];
|
|
SourceToListener.v[1] = -Position.v[1];
|
|
SourceToListener.v[2] = -Position.v[2];
|
|
SourceToListener.v[3] = 0.0f;
|
|
Distance = aluNormalize(SourceToListener.v);
|
|
|
|
/* Calculate distance attenuation */
|
|
ClampedDist = Distance;
|
|
|
|
Attenuation = 1.0f;
|
|
for(i = 0;i < NumSends;i++)
|
|
RoomAttenuation[i] = 1.0f;
|
|
switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel :
|
|
ALContext->DistanceModel)
|
|
{
|
|
case InverseDistanceClamped:
|
|
ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
|
|
if(MaxDist < MinDist)
|
|
break;
|
|
/*fall-through*/
|
|
case InverseDistance:
|
|
if(MinDist > 0.0f)
|
|
{
|
|
ALfloat dist = lerp(MinDist, ClampedDist, Rolloff);
|
|
if(dist > 0.0f) Attenuation = MinDist / dist;
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
dist = lerp(MinDist, ClampedDist, RoomRolloff[i]);
|
|
if(dist > 0.0f) RoomAttenuation[i] = MinDist / dist;
|
|
}
|
|
}
|
|
break;
|
|
|
|
case LinearDistanceClamped:
|
|
ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
|
|
if(MaxDist < MinDist)
|
|
break;
|
|
/*fall-through*/
|
|
case LinearDistance:
|
|
if(MaxDist != MinDist)
|
|
{
|
|
Attenuation = 1.0f - (Rolloff*(ClampedDist-MinDist)/(MaxDist - MinDist));
|
|
Attenuation = maxf(Attenuation, 0.0f);
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(ClampedDist-MinDist)/(MaxDist - MinDist));
|
|
RoomAttenuation[i] = maxf(RoomAttenuation[i], 0.0f);
|
|
}
|
|
}
|
|
break;
|
|
|
|
case ExponentDistanceClamped:
|
|
ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
|
|
if(MaxDist < MinDist)
|
|
break;
|
|
/*fall-through*/
|
|
case ExponentDistance:
|
|
if(ClampedDist > 0.0f && MinDist > 0.0f)
|
|
{
|
|
Attenuation = powf(ClampedDist/MinDist, -Rolloff);
|
|
for(i = 0;i < NumSends;i++)
|
|
RoomAttenuation[i] = powf(ClampedDist/MinDist, -RoomRolloff[i]);
|
|
}
|
|
break;
|
|
|
|
case DisableDistance:
|
|
ClampedDist = MinDist;
|
|
break;
|
|
}
|
|
|
|
/* Source Gain + Attenuation */
|
|
DryGain = SourceVolume * Attenuation;
|
|
for(i = 0;i < NumSends;i++)
|
|
WetGain[i] = SourceVolume * RoomAttenuation[i];
|
|
|
|
/* Distance-based air absorption */
|
|
if(AirAbsorptionFactor > 0.0f && ClampedDist > MinDist)
|
|
{
|
|
ALfloat meters = (ClampedDist-MinDist) * MetersPerUnit;
|
|
DryGainHF *= powf(AIRABSORBGAINHF, AirAbsorptionFactor*meters);
|
|
for(i = 0;i < NumSends;i++)
|
|
WetGainHF[i] *= powf(RoomAirAbsorption[i], AirAbsorptionFactor*meters);
|
|
}
|
|
|
|
if(WetGainAuto)
|
|
{
|
|
ALfloat ApparentDist = 1.0f/maxf(Attenuation, 0.00001f) - 1.0f;
|
|
|
|
/* Apply a decay-time transformation to the wet path, based on the
|
|
* attenuation of the dry path.
|
|
*
|
|
* Using the apparent distance, based on the distance attenuation, the
|
|
* initial decay of the reverb effect is calculated and applied to the
|
|
* wet path.
|
|
*/
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
if(DecayDistance[i] > 0.0f)
|
|
WetGain[i] *= powf(0.001f/*-60dB*/, ApparentDist/DecayDistance[i]);
|
|
}
|
|
}
|
|
|
|
/* Calculate directional soundcones */
|
|
Angle = RAD2DEG(acosf(aluDotproduct(&Direction, &SourceToListener)) * ConeScale) * 2.0f;
|
|
if(Angle > InnerAngle && Angle <= OuterAngle)
|
|
{
|
|
ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
|
|
ConeVolume = lerp(1.0f, ALSource->OuterGain, scale);
|
|
ConeHF = lerp(1.0f, ALSource->OuterGainHF, scale);
|
|
}
|
|
else if(Angle > OuterAngle)
|
|
{
|
|
ConeVolume = ALSource->OuterGain;
|
|
ConeHF = ALSource->OuterGainHF;
|
|
}
|
|
else
|
|
{
|
|
ConeVolume = 1.0f;
|
|
ConeHF = 1.0f;
|
|
}
|
|
|
|
DryGain *= ConeVolume;
|
|
if(WetGainAuto)
|
|
{
|
|
for(i = 0;i < NumSends;i++)
|
|
WetGain[i] *= ConeVolume;
|
|
}
|
|
if(DryGainHFAuto)
|
|
DryGainHF *= ConeHF;
|
|
if(WetGainHFAuto)
|
|
{
|
|
for(i = 0;i < NumSends;i++)
|
|
WetGainHF[i] *= ConeHF;
|
|
}
|
|
|
|
/* Clamp to Min/Max Gain */
|
|
DryGain = clampf(DryGain, MinVolume, MaxVolume);
|
|
for(i = 0;i < NumSends;i++)
|
|
WetGain[i] = clampf(WetGain[i], MinVolume, MaxVolume);
|
|
|
|
/* Apply gain and frequency filters */
|
|
DryGain *= ALSource->Direct.Gain * ListenerGain;
|
|
DryGainHF *= ALSource->Direct.GainHF;
|
|
DryGainLF *= ALSource->Direct.GainLF;
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
WetGain[i] *= ALSource->Send[i].Gain * ListenerGain;
|
|
WetGainHF[i] *= ALSource->Send[i].GainHF;
|
|
WetGainLF[i] *= ALSource->Send[i].GainLF;
|
|
}
|
|
|
|
/* Calculate velocity-based doppler effect */
|
|
if(DopplerFactor > 0.0f)
|
|
{
|
|
const aluVector *lvelocity = &ALContext->Listener->Params.Velocity;
|
|
ALfloat VSS, VLS;
|
|
|
|
if(SpeedOfSound < 1.0f)
|
|
{
|
|
DopplerFactor *= 1.0f/SpeedOfSound;
|
|
SpeedOfSound = 1.0f;
|
|
}
|
|
|
|
VSS = aluDotproduct(&Velocity, &SourceToListener) * DopplerFactor;
|
|
VLS = aluDotproduct(lvelocity, &SourceToListener) * DopplerFactor;
|
|
|
|
Pitch *= clampf(SpeedOfSound-VLS, 1.0f, SpeedOfSound*2.0f - 1.0f) /
|
|
clampf(SpeedOfSound-VSS, 1.0f, SpeedOfSound*2.0f - 1.0f);
|
|
}
|
|
|
|
BufferListItem = ATOMIC_LOAD(&ALSource->queue);
|
|
while(BufferListItem != NULL)
|
|
{
|
|
ALbuffer *ALBuffer;
|
|
if((ALBuffer=BufferListItem->buffer) != NULL)
|
|
{
|
|
/* Calculate fixed-point stepping value, based on the pitch, buffer
|
|
* frequency, and output frequency. */
|
|
Pitch = Pitch * ALBuffer->Frequency / Frequency;
|
|
if(Pitch > (ALfloat)MAX_PITCH)
|
|
voice->Step = MAX_PITCH<<FRACTIONBITS;
|
|
else
|
|
voice->Step = maxi(fastf2i(Pitch*FRACTIONONE + 0.5f), 1);
|
|
BsincPrepare(voice->Step, &voice->SincState);
|
|
|
|
break;
|
|
}
|
|
BufferListItem = BufferListItem->next;
|
|
}
|
|
|
|
if(Device->Render_Mode == HrtfRender)
|
|
{
|
|
/* Full HRTF rendering. Skip the virtual channels and render to the
|
|
* real outputs.
|
|
*/
|
|
aluVector dir = {{ 0.0f, 0.0f, -1.0f, 0.0f }};
|
|
ALfloat ev = 0.0f, az = 0.0f;
|
|
ALfloat radius = ALSource->Radius;
|
|
ALfloat dirfact = 1.0f;
|
|
ALfloat coeffs[MAX_AMBI_COEFFS];
|
|
|
|
voice->Direct.OutBuffer = Device->RealOut.Buffer;
|
|
voice->Direct.OutChannels = Device->RealOut.NumChannels;
|
|
|
|
if(Distance > FLT_EPSILON)
|
|
{
|
|
dir.v[0] = -SourceToListener.v[0];
|
|
dir.v[1] = -SourceToListener.v[1];
|
|
dir.v[2] = -SourceToListener.v[2] * ZScale;
|
|
|
|
/* Calculate elevation and azimuth only when the source is not at
|
|
* the listener. This prevents +0 and -0 Z from producing
|
|
* inconsistent panning. Also, clamp Y in case FP precision errors
|
|
* cause it to land outside of -1..+1. */
|
|
ev = asinf(clampf(dir.v[1], -1.0f, 1.0f));
|
|
az = atan2f(dir.v[0], -dir.v[2]);
|
|
}
|
|
if(radius > 0.0f)
|
|
{
|
|
if(radius >= Distance)
|
|
dirfact *= Distance / radius * 0.5f;
|
|
else
|
|
dirfact *= 1.0f - (asinf(radius / Distance) / F_PI);
|
|
}
|
|
|
|
/* Get the HRIR coefficients and delays. */
|
|
GetLerpedHrtfCoeffs(Device->Hrtf, ev, az, dirfact, DryGain,
|
|
voice->Direct.Hrtf[0].Target.Coeffs,
|
|
voice->Direct.Hrtf[0].Target.Delay);
|
|
|
|
dir.v[0] *= dirfact;
|
|
dir.v[1] *= dirfact;
|
|
dir.v[2] *= dirfact;
|
|
CalcDirectionCoeffs(dir.v, coeffs);
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
if(!SendSlots[i])
|
|
{
|
|
ALuint j;
|
|
for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
|
|
voice->Send[i].Gains[0].Target[j] = 0.0f;
|
|
}
|
|
else
|
|
{
|
|
const ALeffectslot *Slot = SendSlots[i];
|
|
ComputePanningGains(Slot->AmbiCoeffs, Slot->NumChannels, coeffs,
|
|
WetGain[i], voice->Send[i].Gains[0].Target);
|
|
}
|
|
}
|
|
|
|
voice->IsHrtf = AL_TRUE;
|
|
}
|
|
else
|
|
{
|
|
/* Non-HRTF rendering. */
|
|
ALfloat dir[3] = { 0.0f, 0.0f, -1.0f };
|
|
ALfloat radius = ALSource->Radius;
|
|
ALfloat coeffs[MAX_AMBI_COEFFS];
|
|
|
|
/* Get the localized direction, and compute panned gains. */
|
|
if(Distance > FLT_EPSILON)
|
|
{
|
|
dir[0] = -SourceToListener.v[0];
|
|
dir[1] = -SourceToListener.v[1];
|
|
dir[2] = -SourceToListener.v[2] * ZScale;
|
|
}
|
|
if(radius > 0.0f)
|
|
{
|
|
ALfloat dirfact;
|
|
if(radius >= Distance)
|
|
dirfact = Distance / radius * 0.5f;
|
|
else
|
|
dirfact = 1.0f - (asinf(radius / Distance) / F_PI);
|
|
dir[0] *= dirfact;
|
|
dir[1] *= dirfact;
|
|
dir[2] *= dirfact;
|
|
}
|
|
|
|
if(Device->Render_Mode == StereoPair)
|
|
{
|
|
/* Clamp X so it remains within 30 degrees of 0 or 180 degree azimuth. */
|
|
coeffs[0] = clampf(-dir[0], -0.5f, 0.5f) + 0.5;
|
|
voice->Direct.Gains[0].Target[0] = coeffs[0] * DryGain;
|
|
voice->Direct.Gains[0].Target[1] = (1.0f-coeffs[0]) * DryGain;
|
|
for(i = 2;i < MAX_OUTPUT_CHANNELS;i++)
|
|
voice->Direct.Gains[0].Target[i] = 0.0f;
|
|
|
|
CalcDirectionCoeffs(dir, coeffs);
|
|
}
|
|
else
|
|
{
|
|
CalcDirectionCoeffs(dir, coeffs);
|
|
ComputePanningGains(Device->Dry.AmbiCoeffs, Device->Dry.NumChannels, coeffs,
|
|
DryGain, voice->Direct.Gains[0].Target);
|
|
}
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
if(!SendSlots[i])
|
|
{
|
|
ALuint j;
|
|
for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
|
|
voice->Send[i].Gains[0].Target[j] = 0.0f;
|
|
}
|
|
else
|
|
{
|
|
const ALeffectslot *Slot = SendSlots[i];
|
|
ComputePanningGains(Slot->AmbiCoeffs, Slot->NumChannels, coeffs,
|
|
WetGain[i], voice->Send[i].Gains[0].Target);
|
|
}
|
|
}
|
|
|
|
voice->IsHrtf = AL_FALSE;
|
|
}
|
|
|
|
{
|
|
ALfloat hfscale = ALSource->Direct.HFReference / Frequency;
|
|
ALfloat lfscale = ALSource->Direct.LFReference / Frequency;
|
|
DryGainHF = maxf(DryGainHF, 0.0001f);
|
|
DryGainLF = maxf(DryGainLF, 0.0001f);
|
|
voice->Direct.Filters[0].ActiveType = AF_None;
|
|
if(DryGainHF != 1.0f) voice->Direct.Filters[0].ActiveType |= AF_LowPass;
|
|
if(DryGainLF != 1.0f) voice->Direct.Filters[0].ActiveType |= AF_HighPass;
|
|
ALfilterState_setParams(
|
|
&voice->Direct.Filters[0].LowPass, ALfilterType_HighShelf,
|
|
DryGainHF, hfscale, calc_rcpQ_from_slope(DryGainHF, 0.75f)
|
|
);
|
|
ALfilterState_setParams(
|
|
&voice->Direct.Filters[0].HighPass, ALfilterType_LowShelf,
|
|
DryGainLF, lfscale, calc_rcpQ_from_slope(DryGainLF, 0.75f)
|
|
);
|
|
}
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
ALfloat hfscale = ALSource->Send[i].HFReference / Frequency;
|
|
ALfloat lfscale = ALSource->Send[i].LFReference / Frequency;
|
|
WetGainHF[i] = maxf(WetGainHF[i], 0.0001f);
|
|
WetGainLF[i] = maxf(WetGainLF[i], 0.0001f);
|
|
voice->Send[i].Filters[0].ActiveType = AF_None;
|
|
if(WetGainHF[i] != 1.0f) voice->Send[i].Filters[0].ActiveType |= AF_LowPass;
|
|
if(WetGainLF[i] != 1.0f) voice->Send[i].Filters[0].ActiveType |= AF_HighPass;
|
|
ALfilterState_setParams(
|
|
&voice->Send[i].Filters[0].LowPass, ALfilterType_HighShelf,
|
|
WetGainHF[i], hfscale, calc_rcpQ_from_slope(WetGainHF[i], 0.75f)
|
|
);
|
|
ALfilterState_setParams(
|
|
&voice->Send[i].Filters[0].HighPass, ALfilterType_LowShelf,
|
|
WetGainLF[i], lfscale, calc_rcpQ_from_slope(WetGainLF[i], 0.75f)
|
|
);
|
|
}
|
|
}
|
|
|
|
|
|
void UpdateContextSources(ALCcontext *ctx)
|
|
{
|
|
ALvoice *voice, *voice_end;
|
|
ALsource *source;
|
|
|
|
if(ATOMIC_EXCHANGE(ALenum, &ctx->UpdateSources, AL_FALSE))
|
|
{
|
|
CalcListenerParams(ctx->Listener);
|
|
|
|
voice = ctx->Voices;
|
|
voice_end = voice + ctx->VoiceCount;
|
|
for(;voice != voice_end;++voice)
|
|
{
|
|
if(!(source=voice->Source)) continue;
|
|
if(source->state != AL_PLAYING && source->state != AL_PAUSED)
|
|
voice->Source = NULL;
|
|
else
|
|
{
|
|
ATOMIC_STORE(&source->NeedsUpdate, AL_FALSE);
|
|
voice->Update(voice, source, ctx);
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
voice = ctx->Voices;
|
|
voice_end = voice + ctx->VoiceCount;
|
|
for(;voice != voice_end;++voice)
|
|
{
|
|
if(!(source=voice->Source)) continue;
|
|
if(source->state != AL_PLAYING && source->state != AL_PAUSED)
|
|
voice->Source = NULL;
|
|
else if(ATOMIC_EXCHANGE(ALenum, &source->NeedsUpdate, AL_FALSE))
|
|
voice->Update(voice, source, ctx);
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/* Specialized function to clamp to [-1, +1] with only one branch. This also
|
|
* converts NaN to 0. */
|
|
static inline ALfloat aluClampf(ALfloat val)
|
|
{
|
|
if(fabsf(val) <= 1.0f) return val;
|
|
return (ALfloat)((0.0f < val) - (val < 0.0f));
|
|
}
|
|
|
|
static inline ALfloat aluF2F(ALfloat val)
|
|
{ return val; }
|
|
|
|
static inline ALint aluF2I(ALfloat val)
|
|
{
|
|
/* Floats only have a 24-bit mantissa, so [-16777215, +16777215] is the max
|
|
* integer range normalized floats can be safely converted to.
|
|
*/
|
|
return fastf2i(aluClampf(val)*16777215.0f)<<7;
|
|
}
|
|
static inline ALuint aluF2UI(ALfloat val)
|
|
{ return aluF2I(val)+2147483648u; }
|
|
|
|
static inline ALshort aluF2S(ALfloat val)
|
|
{ return fastf2i(aluClampf(val)*32767.0f); }
|
|
static inline ALushort aluF2US(ALfloat val)
|
|
{ return aluF2S(val)+32768; }
|
|
|
|
static inline ALbyte aluF2B(ALfloat val)
|
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{ return fastf2i(aluClampf(val)*127.0f); }
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static inline ALubyte aluF2UB(ALfloat val)
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{ return aluF2B(val)+128; }
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#define DECL_TEMPLATE(T, func) \
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static void Write_##T(ALfloatBUFFERSIZE *InBuffer, ALvoid *OutBuffer, \
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ALuint SamplesToDo, ALuint numchans) \
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{ \
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ALuint i, j; \
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for(j = 0;j < numchans;j++) \
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{ \
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const ALfloat *in = InBuffer[j]; \
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T *restrict out = (T*)OutBuffer + j; \
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for(i = 0;i < SamplesToDo;i++) \
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out[i*numchans] = func(in[i]); \
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} \
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}
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DECL_TEMPLATE(ALfloat, aluF2F)
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DECL_TEMPLATE(ALuint, aluF2UI)
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DECL_TEMPLATE(ALint, aluF2I)
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DECL_TEMPLATE(ALushort, aluF2US)
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DECL_TEMPLATE(ALshort, aluF2S)
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DECL_TEMPLATE(ALubyte, aluF2UB)
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DECL_TEMPLATE(ALbyte, aluF2B)
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#undef DECL_TEMPLATE
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ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
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{
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ALuint SamplesToDo;
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ALvoice *voice, *voice_end;
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ALeffectslot *slot;
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ALsource *source;
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ALCcontext *ctx;
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FPUCtl oldMode;
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ALuint i, c;
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SetMixerFPUMode(&oldMode);
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while(size > 0)
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{
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IncrementRef(&device->MixCount);
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SamplesToDo = minu(size, BUFFERSIZE);
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for(c = 0;c < device->VirtOut.NumChannels;c++)
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memset(device->VirtOut.Buffer[c], 0, SamplesToDo*sizeof(ALfloat));
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for(c = 0;c < device->RealOut.NumChannels;c++)
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memset(device->RealOut.Buffer[c], 0, SamplesToDo*sizeof(ALfloat));
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V0(device->Backend,lock)();
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if((slot=device->DefaultSlot) != NULL)
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{
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if(ATOMIC_EXCHANGE(ALenum, &slot->NeedsUpdate, AL_FALSE))
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V(slot->EffectState,update)(device, slot);
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for(i = 0;i < slot->NumChannels;i++)
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memset(slot->WetBuffer[i], 0, SamplesToDo*sizeof(ALfloat));
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}
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ctx = ATOMIC_LOAD(&device->ContextList);
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while(ctx)
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{
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if(!ctx->DeferUpdates)
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{
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UpdateContextSources(ctx);
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#define UPDATE_SLOT(iter) do { \
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if(ATOMIC_EXCHANGE(ALenum, &(*iter)->NeedsUpdate, AL_FALSE)) \
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V((*iter)->EffectState,update)(device, *iter); \
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for(i = 0;i < (*iter)->NumChannels;i++) \
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memset((*iter)->WetBuffer[i], 0, SamplesToDo*sizeof(ALfloat)); \
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} while(0)
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VECTOR_FOR_EACH(ALeffectslot*, ctx->ActiveAuxSlots, UPDATE_SLOT);
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#undef UPDATE_SLOT
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}
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else
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{
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#define CLEAR_WET_BUFFER(iter) do { \
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for(i = 0;i < (*iter)->NumChannels;i++) \
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memset((*iter)->WetBuffer[i], 0, SamplesToDo*sizeof(ALfloat)); \
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} while(0)
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VECTOR_FOR_EACH(ALeffectslot*, ctx->ActiveAuxSlots, CLEAR_WET_BUFFER);
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#undef CLEAR_WET_BUFFER
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}
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/* source processing */
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voice = ctx->Voices;
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voice_end = voice + ctx->VoiceCount;
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for(;voice != voice_end;++voice)
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{
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source = voice->Source;
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if(source && source->state == AL_PLAYING)
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MixSource(voice, source, device, SamplesToDo);
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}
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/* effect slot processing */
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c = VECTOR_SIZE(ctx->ActiveAuxSlots);
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for(i = 0;i < c;i++)
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{
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const ALeffectslot *slot = VECTOR_ELEM(ctx->ActiveAuxSlots, i);
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ALeffectState *state = slot->EffectState;
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V(state,process)(SamplesToDo, slot->WetBuffer, device->Dry.Buffer,
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device->Dry.NumChannels);
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}
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ctx = ctx->next;
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}
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if(device->DefaultSlot != NULL)
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{
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const ALeffectslot *slot = device->DefaultSlot;
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ALeffectState *state = slot->EffectState;
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V(state,process)(SamplesToDo, slot->WetBuffer, device->Dry.Buffer,
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device->Dry.NumChannels);
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}
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/* Increment the clock time. Every second's worth of samples is
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* converted and added to clock base so that large sample counts don't
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* overflow during conversion. This also guarantees an exact, stable
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* conversion. */
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device->SamplesDone += SamplesToDo;
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device->ClockBase += (device->SamplesDone/device->Frequency) * DEVICE_CLOCK_RES;
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device->SamplesDone %= device->Frequency;
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V0(device->Backend,unlock)();
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if(device->Hrtf)
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{
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int lidx = GetChannelIdxByName(device->RealOut, FrontLeft);
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int ridx = GetChannelIdxByName(device->RealOut, FrontRight);
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if(lidx != -1 && ridx != -1)
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{
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HrtfMixerFunc HrtfMix = SelectHrtfMixer();
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ALuint irsize = GetHrtfIrSize(device->Hrtf);
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MixHrtfParams hrtfparams;
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memset(&hrtfparams, 0, sizeof(hrtfparams));
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for(c = 0;c < device->VirtOut.NumChannels;c++)
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{
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hrtfparams.Current = &device->Hrtf_Params[c];
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hrtfparams.Target = &device->Hrtf_Params[c];
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HrtfMix(device->RealOut.Buffer, lidx, ridx,
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device->VirtOut.Buffer[c], 0, device->Hrtf_Offset, 0,
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irsize, &hrtfparams, &device->Hrtf_State[c], SamplesToDo
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);
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}
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device->Hrtf_Offset += SamplesToDo;
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}
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}
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else
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{
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if(device->Uhj_Encoder)
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{
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int lidx = GetChannelIdxByName(device->RealOut, FrontLeft);
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int ridx = GetChannelIdxByName(device->RealOut, FrontRight);
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if(lidx != -1 && ridx != -1)
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{
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/* Encode to stereo-compatible 2-channel UHJ output. */
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EncodeUhj2(device->Uhj_Encoder,
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device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx],
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device->VirtOut.Buffer, SamplesToDo
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);
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}
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}
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if(device->Bs2b)
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{
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/* Apply binaural/crossfeed filter */
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for(i = 0;i < SamplesToDo;i++)
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{
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float samples[2];
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samples[0] = device->RealOut.Buffer[0][i];
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samples[1] = device->RealOut.Buffer[1][i];
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bs2b_cross_feed(device->Bs2b, samples);
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device->RealOut.Buffer[0][i] = samples[0];
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device->RealOut.Buffer[1][i] = samples[1];
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}
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}
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}
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if(buffer)
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{
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ALfloat (*OutBuffer)[BUFFERSIZE] = device->RealOut.Buffer;
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ALuint OutChannels = device->RealOut.NumChannels;;
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#define WRITE(T, a, b, c, d) do { \
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Write_##T((a), (b), (c), (d)); \
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buffer = (T*)buffer + (c)*(d); \
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} while(0)
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switch(device->FmtType)
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{
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case DevFmtByte:
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WRITE(ALbyte, OutBuffer, buffer, SamplesToDo, OutChannels);
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break;
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case DevFmtUByte:
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WRITE(ALubyte, OutBuffer, buffer, SamplesToDo, OutChannels);
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break;
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case DevFmtShort:
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WRITE(ALshort, OutBuffer, buffer, SamplesToDo, OutChannels);
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break;
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case DevFmtUShort:
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WRITE(ALushort, OutBuffer, buffer, SamplesToDo, OutChannels);
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break;
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case DevFmtInt:
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WRITE(ALint, OutBuffer, buffer, SamplesToDo, OutChannels);
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break;
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case DevFmtUInt:
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WRITE(ALuint, OutBuffer, buffer, SamplesToDo, OutChannels);
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break;
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case DevFmtFloat:
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WRITE(ALfloat, OutBuffer, buffer, SamplesToDo, OutChannels);
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break;
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}
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#undef WRITE
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}
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size -= SamplesToDo;
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IncrementRef(&device->MixCount);
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}
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RestoreFPUMode(&oldMode);
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}
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ALvoid aluHandleDisconnect(ALCdevice *device)
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{
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ALCcontext *Context;
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device->Connected = ALC_FALSE;
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Context = ATOMIC_LOAD(&device->ContextList);
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while(Context)
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{
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ALvoice *voice, *voice_end;
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voice = Context->Voices;
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voice_end = voice + Context->VoiceCount;
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while(voice != voice_end)
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{
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ALsource *source = voice->Source;
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voice->Source = NULL;
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if(source && source->state == AL_PLAYING)
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{
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source->state = AL_STOPPED;
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ATOMIC_STORE(&source->current_buffer, NULL);
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source->position = 0;
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source->position_fraction = 0;
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}
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voice++;
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}
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Context->VoiceCount = 0;
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Context = Context->next;
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}
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}
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