AuroraOpenALSoft/Alc/ALu.c
2008-10-09 23:44:48 -07:00

1218 lines
49 KiB
C

/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#define _CRT_SECURE_NO_DEPRECATE // get rid of sprintf security warnings on VS2005
#include "config.h"
#include <math.h>
#include "alMain.h"
#include "AL/al.h"
#include "AL/alc.h"
#include "alSource.h"
#include "alBuffer.h"
#include "alThunk.h"
#include "alListener.h"
#include "alAuxEffectSlot.h"
#include "alu.h"
#include "bs2b.h"
#if defined(HAVE_STDINT_H)
#include <stdint.h>
typedef int64_t ALint64;
#elif defined(HAVE___INT64)
typedef __int64 ALint64;
#elif (SIZEOF_LONG == 8)
typedef long ALint64;
#elif (SIZEOF_LONG_LONG == 8)
typedef long long ALint64;
#endif
#ifdef HAVE_SQRTF
#define aluSqrt(x) ((ALfloat)sqrtf((float)(x)))
#else
#define aluSqrt(x) ((ALfloat)sqrt((double)(x)))
#endif
#ifdef HAVE_ACOSF
#define aluAcos(x) ((ALfloat)acosf((float)(x)))
#else
#define aluAcos(x) ((ALfloat)acos((double)(x)))
#endif
// fixes for mingw32.
#if defined(max) && !defined(__max)
#define __max max
#endif
#if defined(min) && !defined(__min)
#define __min min
#endif
#define BUFFERSIZE 24000
#define FRACTIONBITS 14
#define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
#define MAX_PITCH 65536
/* Minimum ramp length in milliseconds. The value below was chosen to
* adequately reduce clicks and pops from harsh gain changes. */
#define MIN_RAMP_LENGTH 16
ALboolean DuplicateStereo = AL_FALSE;
/* NOTE: The AL_FORMAT_REAR* enums aren't handled here be cause they're
* converted to AL_FORMAT_QUAD* when loaded */
__inline ALuint aluBytesFromFormat(ALenum format)
{
switch(format)
{
case AL_FORMAT_MONO8:
case AL_FORMAT_STEREO8:
case AL_FORMAT_QUAD8_LOKI:
case AL_FORMAT_QUAD8:
case AL_FORMAT_51CHN8:
case AL_FORMAT_61CHN8:
case AL_FORMAT_71CHN8:
return 1;
case AL_FORMAT_MONO16:
case AL_FORMAT_STEREO16:
case AL_FORMAT_QUAD16_LOKI:
case AL_FORMAT_QUAD16:
case AL_FORMAT_51CHN16:
case AL_FORMAT_61CHN16:
case AL_FORMAT_71CHN16:
return 2;
case AL_FORMAT_MONO_FLOAT32:
case AL_FORMAT_STEREO_FLOAT32:
case AL_FORMAT_QUAD32:
case AL_FORMAT_51CHN32:
case AL_FORMAT_61CHN32:
case AL_FORMAT_71CHN32:
return 4;
default:
return 0;
}
}
__inline ALuint aluChannelsFromFormat(ALenum format)
{
switch(format)
{
case AL_FORMAT_MONO8:
case AL_FORMAT_MONO16:
case AL_FORMAT_MONO_FLOAT32:
return 1;
case AL_FORMAT_STEREO8:
case AL_FORMAT_STEREO16:
case AL_FORMAT_STEREO_FLOAT32:
return 2;
case AL_FORMAT_QUAD8_LOKI:
case AL_FORMAT_QUAD16_LOKI:
case AL_FORMAT_QUAD8:
case AL_FORMAT_QUAD16:
case AL_FORMAT_QUAD32:
return 4;
case AL_FORMAT_51CHN8:
case AL_FORMAT_51CHN16:
case AL_FORMAT_51CHN32:
return 6;
case AL_FORMAT_61CHN8:
case AL_FORMAT_61CHN16:
case AL_FORMAT_61CHN32:
return 7;
case AL_FORMAT_71CHN8:
case AL_FORMAT_71CHN16:
case AL_FORMAT_71CHN32:
return 8;
default:
return 0;
}
}
static __inline ALfloat lpFilter(FILTER *iir, ALfloat input)
{
ALfloat *history = iir->history;
ALfloat a = iir->coeff;
ALfloat output = input;
output = output + (history[0]-output)*a;
history[0] = output;
output = output + (history[1]-output)*a;
history[1] = output;
output = output + (history[2]-output)*a;
history[2] = output;
output = output + (history[3]-output)*a;
history[3] = output;
return output;
}
static __inline ALshort aluF2S(ALfloat Value)
{
ALint i;
i = (ALint)Value;
i = __min( 32767, i);
i = __max(-32768, i);
return ((ALshort)i);
}
static __inline ALvoid aluCrossproduct(ALfloat *inVector1,ALfloat *inVector2,ALfloat *outVector)
{
outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
}
static __inline ALfloat aluDotproduct(ALfloat *inVector1,ALfloat *inVector2)
{
return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] +
inVector1[2]*inVector2[2];
}
static __inline ALvoid aluNormalize(ALfloat *inVector)
{
ALfloat length, inverse_length;
length = aluSqrt(aluDotproduct(inVector, inVector));
if(length != 0.0f)
{
inverse_length = 1.0f/length;
inVector[0] *= inverse_length;
inVector[1] *= inverse_length;
inVector[2] *= inverse_length;
}
}
static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat matrix[3][3])
{
ALfloat result[3];
result[0] = vector[0]*matrix[0][0] + vector[1]*matrix[1][0] + vector[2]*matrix[2][0];
result[1] = vector[0]*matrix[0][1] + vector[1]*matrix[1][1] + vector[2]*matrix[2][1];
result[2] = vector[0]*matrix[0][2] + vector[1]*matrix[1][2] + vector[2]*matrix[2][2];
memcpy(vector, result, sizeof(result));
}
static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
ALenum isMono, ALenum OutputFormat,
ALfloat *drysend, ALfloat *wetsend,
ALfloat *pitch, ALfloat *drygainhf,
ALfloat *wetgainhf)
{
ALfloat InnerAngle,OuterAngle,Angle,Distance,DryMix,WetMix=0.0f;
ALfloat Direction[3],Position[3],SourceToListener[3];
ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff,OuterGainHF;
ALfloat ConeVolume,SourceVolume,PanningFB,PanningLR,ListenerGain;
ALfloat U[3],V[3],N[3];
ALfloat DopplerFactor, DopplerVelocity, flSpeedOfSound, flMaxVelocity;
ALfloat Matrix[3][3];
ALfloat flAttenuation;
ALfloat RoomAttenuation;
ALfloat MetersPerUnit;
ALfloat RoomRolloff;
ALfloat DryGainHF = 1.0f;
ALfloat WetGainHF = 1.0f;
ALfloat cw, a, g;
//Get context properties
DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
DopplerVelocity = ALContext->DopplerVelocity;
flSpeedOfSound = ALContext->flSpeedOfSound;
//Get listener properties
ListenerGain = ALContext->Listener.Gain;
MetersPerUnit = ALContext->Listener.MetersPerUnit;
//Get source properties
SourceVolume = ALSource->flGain;
memcpy(Position, ALSource->vPosition, sizeof(ALSource->vPosition));
memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation));
MinVolume = ALSource->flMinGain;
MaxVolume = ALSource->flMaxGain;
MinDist = ALSource->flRefDistance;
MaxDist = ALSource->flMaxDistance;
Rolloff = ALSource->flRollOffFactor;
InnerAngle = ALSource->flInnerAngle;
OuterAngle = ALSource->flOuterAngle;
OuterGainHF = ALSource->OuterGainHF;
RoomRolloff = ALSource->RoomRolloffFactor;
//Only apply 3D calculations for mono buffers
if(isMono != AL_FALSE)
{
//1. Translate Listener to origin (convert to head relative)
// Note that Direction and SourceToListener are *not* transformed.
// SourceToListener is used with the source and listener velocities,
// which are untransformed, and Direction is used with SourceToListener
// for the sound cone
if(ALSource->bHeadRelative==AL_FALSE)
{
// Build transform matrix
aluCrossproduct(ALContext->Listener.Forward, ALContext->Listener.Up, U); // Right-vector
aluNormalize(U); // Normalized Right-vector
memcpy(V, ALContext->Listener.Up, sizeof(V)); // Up-vector
aluNormalize(V); // Normalized Up-vector
memcpy(N, ALContext->Listener.Forward, sizeof(N)); // At-vector
aluNormalize(N); // Normalized At-vector
Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0];
Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1];
Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2];
// Translate source position into listener space
Position[0] -= ALContext->Listener.Position[0];
Position[1] -= ALContext->Listener.Position[1];
Position[2] -= ALContext->Listener.Position[2];
SourceToListener[0] = -Position[0];
SourceToListener[1] = -Position[1];
SourceToListener[2] = -Position[2];
// Transform source position and direction into listener space
aluMatrixVector(Position, Matrix);
}
else
{
SourceToListener[0] = -Position[0];
SourceToListener[1] = -Position[1];
SourceToListener[2] = -Position[2];
}
aluNormalize(SourceToListener);
aluNormalize(Direction);
//2. Calculate distance attenuation
Distance = aluSqrt(aluDotproduct(Position, Position));
if(ALSource->Send[0].Slot)
{
if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB)
RoomRolloff += ALSource->Send[0].Slot->effect.Reverb.RoomRolloffFactor;
}
flAttenuation = 1.0f;
RoomAttenuation = 1.0f;
switch (ALContext->DistanceModel)
{
case AL_INVERSE_DISTANCE_CLAMPED:
Distance=__max(Distance,MinDist);
Distance=__min(Distance,MaxDist);
if (MaxDist < MinDist)
break;
//fall-through
case AL_INVERSE_DISTANCE:
if (MinDist > 0.0f)
{
if ((MinDist + (Rolloff * (Distance - MinDist))) > 0.0f)
flAttenuation = MinDist / (MinDist + (Rolloff * (Distance - MinDist)));
if ((MinDist + (RoomRolloff * (Distance - MinDist))) > 0.0f)
RoomAttenuation = MinDist / (MinDist + (RoomRolloff * (Distance - MinDist)));
}
break;
case AL_LINEAR_DISTANCE_CLAMPED:
Distance=__max(Distance,MinDist);
Distance=__min(Distance,MaxDist);
if (MaxDist < MinDist)
break;
//fall-through
case AL_LINEAR_DISTANCE:
Distance=__min(Distance,MaxDist);
if (MaxDist != MinDist)
{
flAttenuation = 1.0f - (Rolloff*(Distance-MinDist)/(MaxDist - MinDist));
RoomAttenuation = 1.0f - (RoomRolloff*(Distance-MinDist)/(MaxDist - MinDist));
}
break;
case AL_EXPONENT_DISTANCE_CLAMPED:
Distance=__max(Distance,MinDist);
Distance=__min(Distance,MaxDist);
if (MaxDist < MinDist)
break;
//fall-through
case AL_EXPONENT_DISTANCE:
if ((Distance > 0.0f) && (MinDist > 0.0f))
{
flAttenuation = (ALfloat)pow(Distance/MinDist, -Rolloff);
RoomAttenuation = (ALfloat)pow(Distance/MinDist, -RoomRolloff);
}
break;
case AL_NONE:
flAttenuation = 1.0f;
RoomAttenuation = 1.0f;
break;
}
// Distance-based air absorption
if(ALSource->AirAbsorptionFactor > 0.0f && ALContext->DistanceModel != AL_NONE)
{
ALfloat dist = Distance-MinDist;
ALfloat absorb;
if(dist < 0.0f) dist = 0.0f;
// Absorption calculation is done in dB
absorb = (ALSource->AirAbsorptionFactor*AIRABSORBGAINDBHF) *
(Distance*MetersPerUnit);
// Convert dB to linear gain before applying
absorb = pow(0.5, absorb/-6.0);
DryGainHF *= absorb;
WetGainHF *= absorb;
}
// Source Gain + Attenuation and clamp to Min/Max Gain
DryMix = SourceVolume * flAttenuation;
DryMix = __min(DryMix,MaxVolume);
DryMix = __max(DryMix,MinVolume);
WetMix = SourceVolume * RoomAttenuation;
WetMix = __min(WetMix,MaxVolume);
WetMix = __max(WetMix,MinVolume);
//3. Apply directional soundcones
Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f /
3.141592654f;
if(Angle >= InnerAngle && Angle <= OuterAngle)
{
ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)*scale);
DryMix *= ConeVolume;
if(ALSource->WetGainAuto)
WetMix *= ConeVolume;
if(ALSource->DryGainHFAuto)
DryGainHF *= (1.0f+(OuterGainHF-1.0f)*scale);
if(ALSource->WetGainHFAuto)
WetGainHF *= (1.0f+(OuterGainHF-1.0f)*scale);
}
else if(Angle > OuterAngle)
{
ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f));
DryMix *= ConeVolume;
if(ALSource->WetGainAuto)
WetMix *= ConeVolume;
if(ALSource->DryGainHFAuto)
DryGainHF *= (1.0f+(OuterGainHF-1.0f));
if(ALSource->WetGainHFAuto)
WetGainHF *= (1.0f+(OuterGainHF-1.0f));
}
//4. Calculate Velocity
if(DopplerFactor != 0.0f)
{
ALfloat flVSS, flVLS = 0.0f;
if(ALSource->bHeadRelative==AL_FALSE)
flVLS = aluDotproduct(ALContext->Listener.Velocity, SourceToListener);
flVSS = aluDotproduct(ALSource->vVelocity, SourceToListener);
flMaxVelocity = (DopplerVelocity * flSpeedOfSound) / DopplerFactor;
if (flVSS >= flMaxVelocity)
flVSS = (flMaxVelocity - 1.0f);
else if (flVSS <= -flMaxVelocity)
flVSS = -flMaxVelocity + 1.0f;
if (flVLS >= flMaxVelocity)
flVLS = (flMaxVelocity - 1.0f);
else if (flVLS <= -flMaxVelocity)
flVLS = -flMaxVelocity + 1.0f;
pitch[0] = ALSource->flPitch *
((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVLS)) /
((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVSS));
}
else
pitch[0] = ALSource->flPitch;
if(ALSource->Send[0].Slot &&
ALSource->Send[0].Slot->effect.type != AL_EFFECT_NULL)
{
// If the slot's auxilliary send auto is off, the data sent to the
// effect slot is the same as the dry path, sans filter effects
if(!ALSource->Send[0].Slot->AuxSendAuto)
{
WetMix = DryMix;
WetGainHF = DryGainHF;
}
// Note that these are really applied by the effect slot. However,
// it's easier to handle them here (particularly the lowpass
// filter). Applying the gain to the individual sources going to
// the effect slot should have the same effect as applying the gain
// to the accumulated sources in the effect slot.
// vol1*g + vol2*g + ... voln*g = (vol1+vol2+...voln)*g
WetMix *= ALSource->Send[0].Slot->Gain;
if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB)
{
WetMix *= ALSource->Send[0].Slot->effect.Reverb.Gain;
WetGainHF *= ALSource->Send[0].Slot->effect.Reverb.GainHF;
WetGainHF *= pow(ALSource->Send[0].Slot->effect.Reverb.AirAbsorptionGainHF,
Distance * MetersPerUnit);
}
}
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);
}