AuroraOpenALSoft/Alc/ALu.c

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2007-11-14 02:02:18 +00:00
/**
* 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"
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#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
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#ifdef HAVE_ACOSF
#define aluAcos(x) ((ALfloat)acosf((float)(x)))
#else
#define aluAcos(x) ((ALfloat)acos((double)(x)))
#endif
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// fixes for mingw32.
#if defined(max) && !defined(__max)
#define __max max
#endif
#if defined(min) && !defined(__min)
#define __min min
#endif
#define BUFFERSIZE 24000
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#define FRACTIONBITS 14
#define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
#define MAX_PITCH 4
/* 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
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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 */
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__inline ALuint aluBytesFromFormat(ALenum format)
{
switch(format)
{
case AL_FORMAT_MONO8:
case AL_FORMAT_STEREO8:
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case AL_FORMAT_QUAD8_LOKI:
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case AL_FORMAT_QUAD8:
case AL_FORMAT_51CHN8:
case AL_FORMAT_61CHN8:
case AL_FORMAT_71CHN8:
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return 1;
case AL_FORMAT_MONO16:
case AL_FORMAT_STEREO16:
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case AL_FORMAT_QUAD16_LOKI:
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case AL_FORMAT_QUAD16:
case AL_FORMAT_51CHN16:
case AL_FORMAT_61CHN16:
case AL_FORMAT_71CHN16:
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return 2;
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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:
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return 4;
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default:
return 0;
}
}
__inline ALuint aluChannelsFromFormat(ALenum format)
{
switch(format)
{
case AL_FORMAT_MONO8:
case AL_FORMAT_MONO16:
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case AL_FORMAT_MONO_FLOAT32:
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return 1;
case AL_FORMAT_STEREO8:
case AL_FORMAT_STEREO16:
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case AL_FORMAT_STEREO_FLOAT32:
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return 2;
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case AL_FORMAT_QUAD8_LOKI:
case AL_FORMAT_QUAD16_LOKI:
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case AL_FORMAT_QUAD8:
case AL_FORMAT_QUAD16:
case AL_FORMAT_QUAD32:
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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;
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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;
}
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static __inline ALshort aluF2S(ALfloat Value)
{
ALint i;
i = (ALint)Value;
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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;
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length = aluSqrt(aluDotproduct(inVector, inVector));
if(length != 0.0f)
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{
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));
}
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static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
ALenum isMono, ALenum OutputFormat,
ALfloat *drysend, ALfloat *wetsend,
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ALfloat *pitch, ALfloat *drygainhf,
ALfloat *wetgainhf)
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{
ALfloat InnerAngle,OuterAngle,Angle,Distance,DryMix,WetMix=0.0f;
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ALfloat Direction[3],Position[3],SourceToListener[3];
ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff,OuterGainHF;
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ALfloat ConeVolume,SourceVolume,PanningFB,PanningLR,ListenerGain;
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ALfloat U[3],V[3],N[3];
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;
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//Get context properties
DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
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DopplerVelocity = ALContext->DopplerVelocity;
flSpeedOfSound = ALContext->flSpeedOfSound;
//Get listener properties
ListenerGain = ALContext->Listener.Gain;
MetersPerUnit = ALContext->Listener.MetersPerUnit;
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//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;
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//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
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if(ALSource->bHeadRelative==AL_FALSE)
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{
// 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
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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);
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}
else
{
SourceToListener[0] = -Position[0];
SourceToListener[1] = -Position[1];
SourceToListener[2] = -Position[2];
}
aluNormalize(SourceToListener);
aluNormalize(Direction);
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//2. Calculate distance attenuation
Distance = aluSqrt(aluDotproduct(Position, Position));
if(ALSource->Send[0].Slot)
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
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{
if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB)
RoomRolloff += ALSource->Send[0].Slot->effect.Reverb.RoomRolloffFactor;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
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}
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flAttenuation = 1.0f;
RoomAttenuation = 1.0f;
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switch (ALContext->DistanceModel)
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{
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)));
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}
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)
{
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flAttenuation = 1.0f - (Rolloff*(Distance-MinDist)/(MaxDist - MinDist));
RoomAttenuation = 1.0f - (RoomRolloff*(Distance-MinDist)/(MaxDist - MinDist));
}
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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))
{
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flAttenuation = (ALfloat)pow(Distance/MinDist, -Rolloff);
RoomAttenuation = (ALfloat)pow(Distance/MinDist, -RoomRolloff);
}
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break;
case AL_NONE:
flAttenuation = 1.0f;
RoomAttenuation = 1.0f;
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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);
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//3. Apply directional soundcones
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Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f /
3.141592654f;
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if(Angle >= InnerAngle && Angle <= OuterAngle)
{
ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
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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);
}
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else if(Angle > OuterAngle)
{
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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));
}
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//4. Calculate Velocity
if(DopplerFactor != 0.0f)
{
ALfloat flVSS, flVLS = 0.0f;
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if(ALSource->bHeadRelative==AL_FALSE)
flVLS = aluDotproduct(ALContext->Listener.Velocity, SourceToListener);
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flVSS = aluDotproduct(ALSource->vVelocity, SourceToListener);
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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;
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pitch[0] = ALSource->flPitch *
((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVLS)) /
((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVSS));
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}
else
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pitch[0] = ALSource->flPitch;
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if(ALSource->Send[0].Slot &&
ALSource->Send[0].Slot->effect.type != AL_EFFECT_NULL)
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
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{
// 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;
}
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
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// 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;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB)
{
WetMix *= ALSource->Send[0].Slot->effect.Reverb.Gain;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
WetGainHF *= ALSource->Send[0].Slot->effect.Reverb.GainHF;
WetGainHF *= pow(ALSource->Send[0].Slot->effect.Reverb.AirAbsorptionGainHF,
Distance * MetersPerUnit);
}
}
2008-01-19 05:39:09 +00:00
else
{
WetMix = 0.0f;
WetGainHF = 1.0f;
}
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
//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))
2007-11-14 02:02:18 +00:00
{
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] = drysend[FRONT_LEFT];
drysend[BACK_RIGHT] = drysend[FRONT_RIGHT];
drysend[SIDE_LEFT] = drysend[FRONT_LEFT];
drysend[SIDE_RIGHT] = drysend[FRONT_RIGHT];
wetsend[FRONT_LEFT] = WetMix * aluSqrt(1.0f-PanningLR); //L Room
wetsend[FRONT_RIGHT] = WetMix * aluSqrt( PanningLR); //R Room
wetsend[BACK_LEFT] = wetsend[FRONT_LEFT];
wetsend[BACK_RIGHT] = wetsend[FRONT_RIGHT];
wetsend[SIDE_LEFT] = wetsend[FRONT_LEFT];
wetsend[SIDE_RIGHT] = wetsend[FRONT_RIGHT];
2007-11-14 02:02:18 +00:00
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] = (drysend[FRONT_LEFT] +drysend[BACK_LEFT]) * 0.5f;
drysend[SIDE_RIGHT] = (drysend[FRONT_RIGHT]+drysend[BACK_RIGHT]) * 0.5f;
wetsend[FRONT_LEFT] = WetMix * aluSqrt((1.0f-PanningLR)*(1.0f-PanningFB));
wetsend[FRONT_RIGHT] = WetMix * aluSqrt(( PanningLR)*(1.0f-PanningFB));
wetsend[BACK_LEFT] = WetMix * aluSqrt((1.0f-PanningLR)*( PanningFB));
wetsend[BACK_RIGHT] = WetMix * aluSqrt(( PanningLR)*( PanningFB));
wetsend[SIDE_LEFT] = (wetsend[FRONT_LEFT] +wetsend[BACK_LEFT]) * 0.5f;
wetsend[SIDE_RIGHT] = (wetsend[FRONT_RIGHT]+wetsend[BACK_RIGHT]) * 0.5f;
2007-11-14 02:02:18 +00:00
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;
wetsend[BACK_LEFT] = WetMix * aluSqrt((1.0f-PanningLR)*(1.0f-PanningFB));
wetsend[BACK_RIGHT] = WetMix * aluSqrt(( PanningLR)*(1.0f-PanningFB));
wetsend[SIDE_LEFT] = WetMix * aluSqrt((1.0f-PanningLR)*( PanningFB));
wetsend[SIDE_RIGHT] = WetMix * aluSqrt(( PanningLR)*( PanningFB));
2008-01-19 05:39:09 +00:00
wetsend[FRONT_LEFT] = 0.0f;
wetsend[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;
wetsend[FRONT_LEFT] = WetMix * aluSqrt((1.0f-PanningLR)*(1.0f-PanningFB));
wetsend[FRONT_RIGHT] = WetMix * aluSqrt(( PanningLR)*(1.0f-PanningFB));
wetsend[SIDE_LEFT] = WetMix * aluSqrt((1.0f-PanningLR)*( PanningFB));
wetsend[SIDE_RIGHT] = WetMix * aluSqrt(( PanningLR)*( PanningFB));
2008-01-19 05:39:09 +00:00
wetsend[BACK_LEFT] = 0.0f;
wetsend[BACK_RIGHT] = 0.0f;
}
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default:
break;
}
2008-01-12 15:36:22 +00:00
// 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;
2008-01-12 15:36:22 +00:00
*drygainhf = DryGainHF;
*wetgainhf = WetGainHF;
2007-11-14 02:02:18 +00:00
}
else
{
//1. Multi-channel buffers always play "normal"
2008-01-16 05:57:50 +00:00
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[FRONT_LEFT] = 0.0f;
wetsend[FRONT_RIGHT] = 0.0f;
wetsend[SIDE_LEFT] = 0.0f;
wetsend[SIDE_RIGHT] = 0.0f;
wetsend[BACK_LEFT] = 0.0f;
wetsend[BACK_RIGHT] = 0.0f;
wetsend[CENTER] = 0.0f;
wetsend[LFE] = 0.0f;
WetGainHF = 1.0f;
*drygainhf = DryGainHF;
*wetgainhf = WetGainHF;
2007-11-14 02:02:18 +00:00
}
}
static __inline ALshort lerp(ALshort val1, ALshort val2, ALint frac)
{
return (val1*((1<<FRACTIONBITS)-frac) + val2*frac) >> FRACTIONBITS;
}
2007-11-14 02:02:18 +00:00
ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum format)
{
static float DryBuffer[BUFFERSIZE][OUTPUTCHANNELS];
static float WetBuffer[BUFFERSIZE][OUTPUTCHANNELS];
ALfloat newDrySend[OUTPUTCHANNELS] = { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f };
ALfloat newWetSend[OUTPUTCHANNELS] = { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f };
2007-12-18 05:00:52 +00:00
ALfloat DryGainHF = 0.0f;
ALfloat WetGainHF = 0.0f;
ALfloat *DrySend;
ALfloat *WetSend;
ALuint rampLength;
ALfloat dryGainStep[OUTPUTCHANNELS];
ALfloat wetGainStep[OUTPUTCHANNELS];
2007-11-14 02:02:18 +00:00
ALuint BlockAlign,BufferSize;
ALuint DataSize=0,DataPosInt=0,DataPosFrac=0;
2008-01-16 05:57:50 +00:00
ALuint Channels,Frequency,ulExtraSamples;
2007-11-14 02:02:18 +00:00
ALfloat Pitch;
ALint Looping,State;
ALint increment;
ALuint Buffer;
2007-11-14 02:02:18 +00:00
ALuint SamplesToDo;
ALsource *ALSource;
ALbuffer *ALBuffer;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
ALeffectslot *ALEffectSlot;
2007-11-14 02:02:18 +00:00
ALfloat value;
ALshort *Data;
ALuint i,j,k;
ALbufferlistitem *BufferListItem;
ALuint loop;
ALint64 DataSize64,DataPos64;
FILTER *DryFilter, *WetFilter;
int fpuState;
2007-11-14 02:02:18 +00:00
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);
2007-11-14 02:02:18 +00:00
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(DryBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat));
memset(WetBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat));
//Actual mixing loop
while(ALSource)
2007-11-14 02:02:18 +00:00
{
j = 0;
State = ALSource->state;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
while(State == AL_PLAYING && j < SamplesToDo)
{
DataSize = 0;
DataPosInt = 0;
DataPosFrac = 0;
2007-11-14 02:02:18 +00:00
//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->WetGains;
//Compute the gain steps for each output channel
for(i = 0;i < OUTPUTCHANNELS;i++)
{
dryGainStep[i] = (newDrySend[i]-DrySend[i]) / rampLength;
wetGainStep[i] = (newWetSend[i]-WetSend[i]) / rampLength;
}
//Compute 18.14 fixed point step
increment = (ALint)(Pitch*(ALfloat)(1L<<FRACTIONBITS));
if(increment > (MAX_PITCH<<FRACTIONBITS))
increment = (MAX_PITCH<<FRACTIONBITS);
else 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++)
2007-11-14 02:02:18 +00:00
{
if(BufferListItem)
BufferListItem = BufferListItem->next;
}
if (BufferListItem)
{
if (BufferListItem->next)
2007-11-14 02:02:18 +00:00
{
ALbuffer *NextBuf = (ALbuffer*)ALTHUNK_LOOKUPENTRY(BufferListItem->next->buffer);
if(NextBuf && NextBuf->data)
2007-11-14 02:02:18 +00:00
{
ulExtraSamples = min(NextBuf->size, (ALint)(ALBuffer->padding*Channels*2));
memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
2007-11-14 02:02:18 +00:00
}
}
else if (ALSource->bLooping)
{
ALbuffer *NextBuf = (ALbuffer*)ALTHUNK_LOOKUPENTRY(ALSource->queue->buffer);
if (NextBuf && NextBuf->data)
2007-11-14 02:02:18 +00:00
{
ulExtraSamples = min(NextBuf->size, (ALint)(ALBuffer->padding*Channels*2));
memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
2007-11-14 02:02:18 +00:00
}
}
else
memset(&Data[DataSize*Channels], 0, (ALBuffer->padding*Channels*2));
}
BufferSize = min(BufferSize, (SamplesToDo-j));
2007-11-14 02:02:18 +00:00
//Actual sample mixing loop
k = 0;
Data += DataPosInt*Channels;
while(BufferSize--)
{
for(i = 0;i < OUTPUTCHANNELS;i++)
{
DrySend[i] += dryGainStep[i];
WetSend[i] += wetGainStep[i];
}
if(Channels==1)
2007-11-14 02:02:18 +00:00
{
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][FRONT_LEFT] += outsamp*WetSend[FRONT_LEFT];
WetBuffer[j][FRONT_RIGHT] += outsamp*WetSend[FRONT_RIGHT];
WetBuffer[j][SIDE_LEFT] += outsamp*WetSend[SIDE_LEFT];
WetBuffer[j][SIDE_RIGHT] += outsamp*WetSend[SIDE_RIGHT];
WetBuffer[j][BACK_LEFT] += outsamp*WetSend[BACK_LEFT];
WetBuffer[j][BACK_RIGHT] += outsamp*WetSend[BACK_RIGHT];
}
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)
2007-11-14 02:02:18 +00:00
{
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++;
}
2007-11-14 02:02:18 +00:00
}
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];
}
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}
DataPosFrac += increment;
k += DataPosFrac>>FRACTIONBITS;
DataPosFrac &= FRACTIONMASK;
j++;
2007-11-14 02:02:18 +00:00
}
DataPosInt += k;
//Update source info
ALSource->position = DataPosInt;
ALSource->position_fraction = DataPosFrac;
skipmix: ;
}
2007-11-14 02:02:18 +00:00
//Handle looping sources
if(!Buffer || DataPosInt >= DataSize)
{
//queueing
if(ALSource->queue)
2007-11-14 02:02:18 +00:00
{
Looping = ALSource->bLooping;
if(ALSource->BuffersPlayed < (ALSource->BuffersInQueue-1))
2007-11-14 02:02:18 +00:00
{
BufferListItem = ALSource->queue;
for(loop = 0; loop <= ALSource->BuffersPlayed; loop++)
2007-11-14 02:02:18 +00:00
{
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;
2008-08-16 00:43:07 +00:00
ALSource->BuffersPlayed = ALSource->BuffersInQueue;
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BufferListItem = ALSource->queue;
while(BufferListItem != NULL)
2007-11-14 02:02:18 +00:00
{
BufferListItem->bufferstate = PROCESSED;
BufferListItem = BufferListItem->next;
2007-11-14 02:02:18 +00:00
}
}
else
{
/* alSourceRewind */
/* alSourcePlay */
ALSource->state = AL_PLAYING;
ALSource->inuse = AL_TRUE;
ALSource->play = AL_TRUE;
ALSource->BuffersPlayed = 0;
ALSource->BufferPosition = 0;
ALSource->lBytesPlayed = 0;
BufferListItem = ALSource->queue;
while(BufferListItem != NULL)
2007-11-14 02:02:18 +00:00
{
BufferListItem->bufferstate = PENDING;
BufferListItem = BufferListItem->next;
2007-11-14 02:02:18 +00:00
}
ALSource->ulBufferID = ALSource->queue->buffer;
2007-11-14 02:02:18 +00:00
ALSource->position = DataPosInt-DataSize;
ALSource->position_fraction = DataPosFrac;
2007-11-14 02:02:18 +00:00
}
}
}
}
//Get source state
State = ALSource->state;
2007-11-14 02:02:18 +00:00
}
ALSource = ALSource->next;
}
// effect slot processing
while(ALEffectSlot)
{
if(ALEffectSlot->effect.type == AL_EFFECT_REVERB)
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
{
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++)
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
{
sample = WetBuffer[i][FRONT_LEFT] +WetBuffer[i][SIDE_LEFT] +WetBuffer[i][BACK_LEFT];
sample += WetBuffer[i][FRONT_RIGHT]+WetBuffer[i][SIDE_RIGHT]+WetBuffer[i][BACK_RIGHT];
DelayBuffer[Pos] = sample / 6.0f;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
sample = DelayBuffer[ReflectPos] * ReflectGain;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
DelayBuffer[LatePos] *= LateReverbGain;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
Pos = (Pos+1) % Length;
lowsample = lpFilter(WetFilter, DelayBuffer[Pos]);
lowsample += (DelayBuffer[Pos]-lowsample) * DecayHFRatio;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
DelayBuffer[LatePos] += lowsample * DecayGain;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
sample += DelayBuffer[LatePos];
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
WetBuffer[i][FRONT_LEFT] = sample;
WetBuffer[i][FRONT_RIGHT] = sample;
WetBuffer[i][SIDE_LEFT] = sample;
WetBuffer[i][SIDE_RIGHT] = sample;
WetBuffer[i][BACK_LEFT] = sample;
WetBuffer[i][BACK_RIGHT] = sample;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
LatePos = (LatePos+1) % Length;
ReflectPos = (ReflectPos+1) % Length;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
}
2008-01-21 06:16:28 +00:00
ALEffectSlot->ReverbPos = Pos;
ALEffectSlot->ReverbLatePos = LatePos;
ALEffectSlot->ReverbReflectPos = ReflectPos;
Implement AL_EFFECT_REVERB Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
2008-01-19 05:25:40 +00:00
}
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]+
WetBuffer[i][FRONT_LEFT]+WetBuffer[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] +WetBuffer[i][FRONT_LEFT];
samples[1] = DryBuffer[i][FRONT_RIGHT]+WetBuffer[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;
2007-11-14 02:02:18 +00:00
}
}
else
{
for(i = 0;i < SamplesToDo;i++)
{
((ALubyte*)buffer)[0] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_LEFT] +WetBuffer[i][FRONT_LEFT])>>8)+128);
((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT]+WetBuffer[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] +WetBuffer[i][FRONT_LEFT])>>8)+128);
((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT]+WetBuffer[i][FRONT_RIGHT])>>8)+128);
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT])>>8)+128);
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[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] +WetBuffer[i][FRONT_LEFT])>>8)+128);
((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT]+WetBuffer[i][FRONT_RIGHT])>>8)+128);
#ifdef _WIN32 /* Of course, Windows can't use the same ordering... */
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][CENTER] +WetBuffer[i][CENTER])>>8)+128);
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][LFE] +WetBuffer[i][LFE])>>8)+128);
((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT])>>8)+128);
((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT])>>8)+128);
#else
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT])>>8)+128);
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT])>>8)+128);
((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][CENTER] +WetBuffer[i][CENTER])>>8)+128);
((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][LFE] +WetBuffer[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] +WetBuffer[i][FRONT_LEFT])>>8)+128);
((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT]+WetBuffer[i][FRONT_RIGHT])>>8)+128);
#ifdef _WIN32
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][LFE] +WetBuffer[i][LFE])>>8)+128);
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT])>>8)+128);
((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT])>>8)+128);
#else
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT])>>8)+128);
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT])>>8)+128);
((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][LFE] +WetBuffer[i][LFE])>>8)+128);
#endif
((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_LEFT] +WetBuffer[i][SIDE_LEFT])>>8)+128);
((ALubyte*)buffer)[6] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_RIGHT] +WetBuffer[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] +WetBuffer[i][FRONT_LEFT])>>8)+128);
((ALubyte*)buffer)[1] = (ALubyte)((aluF2S(DryBuffer[i][FRONT_RIGHT]+WetBuffer[i][FRONT_RIGHT])>>8)+128);
#ifdef _WIN32
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][CENTER] +WetBuffer[i][CENTER])>>8)+128);
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][LFE] +WetBuffer[i][LFE])>>8)+128);
((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT])>>8)+128);
((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT])>>8)+128);
#else
((ALubyte*)buffer)[2] = (ALubyte)((aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT])>>8)+128);
((ALubyte*)buffer)[3] = (ALubyte)((aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT])>>8)+128);
((ALubyte*)buffer)[4] = (ALubyte)((aluF2S(DryBuffer[i][CENTER] +WetBuffer[i][CENTER])>>8)+128);
((ALubyte*)buffer)[5] = (ALubyte)((aluF2S(DryBuffer[i][LFE] +WetBuffer[i][LFE])>>8)+128);
#endif
((ALubyte*)buffer)[6] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_LEFT] +WetBuffer[i][SIDE_LEFT])>>8)+128);
((ALubyte*)buffer)[7] = (ALubyte)((aluF2S(DryBuffer[i][SIDE_RIGHT] +WetBuffer[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]+
WetBuffer[i][FRONT_LEFT]+WetBuffer[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] +WetBuffer[i][FRONT_LEFT];
samples[1] = DryBuffer[i][FRONT_RIGHT]+WetBuffer[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] +WetBuffer[i][FRONT_LEFT]);
((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]+WetBuffer[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] +WetBuffer[i][FRONT_LEFT]);
((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]+WetBuffer[i][FRONT_RIGHT]);
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT]);
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[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] +WetBuffer[i][FRONT_LEFT]);
((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]+WetBuffer[i][FRONT_RIGHT]);
#ifdef _WIN32
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][CENTER] +WetBuffer[i][CENTER]);
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][LFE] +WetBuffer[i][LFE]);
((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT]);
((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT]);
#else
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT]);
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT]);
((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][CENTER] +WetBuffer[i][CENTER]);
((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][LFE] +WetBuffer[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] +WetBuffer[i][FRONT_LEFT]);
((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]+WetBuffer[i][FRONT_RIGHT]);
#ifdef _WIN32
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][LFE] +WetBuffer[i][LFE]);
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT]);
((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT]);
#else
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT]);
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT]);
((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][LFE] +WetBuffer[i][LFE]);
#endif
((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][SIDE_LEFT] +WetBuffer[i][SIDE_LEFT]);
((ALshort*)buffer)[6] = aluF2S(DryBuffer[i][SIDE_RIGHT] +WetBuffer[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] +WetBuffer[i][FRONT_LEFT]);
((ALshort*)buffer)[1] = aluF2S(DryBuffer[i][FRONT_RIGHT]+WetBuffer[i][FRONT_RIGHT]);
#ifdef _WIN32
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][CENTER] +WetBuffer[i][CENTER]);
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][LFE] +WetBuffer[i][LFE]);
((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT]);
((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT]);
#else
((ALshort*)buffer)[2] = aluF2S(DryBuffer[i][BACK_LEFT] +WetBuffer[i][BACK_LEFT]);
((ALshort*)buffer)[3] = aluF2S(DryBuffer[i][BACK_RIGHT] +WetBuffer[i][BACK_RIGHT]);
((ALshort*)buffer)[4] = aluF2S(DryBuffer[i][CENTER] +WetBuffer[i][CENTER]);
((ALshort*)buffer)[5] = aluF2S(DryBuffer[i][LFE] +WetBuffer[i][LFE]);
#endif
((ALshort*)buffer)[6] = aluF2S(DryBuffer[i][SIDE_LEFT] +WetBuffer[i][SIDE_LEFT]);
((ALshort*)buffer)[7] = aluF2S(DryBuffer[i][SIDE_RIGHT] +WetBuffer[i][SIDE_RIGHT]);
buffer = ((ALshort*)buffer) + 8;
}
break;
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default:
break;
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}
size -= SamplesToDo;
2007-11-14 02:02:18 +00:00
}
#if defined(HAVE_FESETROUND)
fesetround(fpuState);
#elif defined(HAVE__CONTROLFP)
_controlfp(fpuState, 0xfffff);
#endif
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ProcessContext(ALContext);
}