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/**
* 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
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <ctype.h>
#include <assert.h>
#include "alMain.h"
#include "AL/al.h"
#include "AL/alc.h"
#include "alSource.h"
#include "alBuffer.h"
#include "alListener.h"
#include "alAuxEffectSlot.h"
#include "alu.h"
#include "bs2b.h"
static __inline ALvoid aluCrossproduct(const ALfloat *inVector1, const 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(const ALfloat *inVector1, const 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 w,ALfloat matrix[4][4])
{
ALfloat temp[4] = {
vector[0], vector[1], vector[2], w
};
vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0];
vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1];
vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2];
}
ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
{
ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
ALbufferlistitem *BufferListItem;
enum DevFmtChannels DevChans;
enum FmtChannels Channels;
ALfloat DryGain, DryGainHF;
ALfloat WetGain[MAX_SENDS];
ALfloat WetGainHF[MAX_SENDS];
ALint NumSends, Frequency;
ALboolean DupStereo;
ALfloat Pitch;
ALfloat cw;
ALint i;
/* Get device properties */
DevChans = ALContext->Device->FmtChans;
DupStereo = ALContext->Device->DuplicateStereo;
NumSends = ALContext->Device->NumAuxSends;
Frequency = ALContext->Device->Frequency;
/* Get listener properties */
ListenerGain = ALContext->Listener.Gain;
/* Get source properties */
SourceVolume = ALSource->flGain;
MinVolume = ALSource->flMinGain;
MaxVolume = ALSource->flMaxGain;
Pitch = ALSource->flPitch;
/* Calculate the stepping value */
Channels = FmtMono;
BufferListItem = ALSource->queue;
while(BufferListItem != NULL)
{
ALbuffer *ALBuffer;
if((ALBuffer=BufferListItem->buffer) != NULL)
{
ALint maxstep = STACK_DATA_SIZE / FrameSizeFromFmt(ALBuffer->FmtChannels,
ALBuffer->FmtType);
maxstep -= ResamplerPadding[ALSource->Resampler] +
ResamplerPrePadding[ALSource->Resampler] + 1;
maxstep = min(maxstep, INT_MAX>>FRACTIONBITS);
Pitch = Pitch * ALBuffer->Frequency / Frequency;
if(Pitch > (ALfloat)maxstep)
ALSource->Params.Step = maxstep<<FRACTIONBITS;
else
{
ALSource->Params.Step = Pitch*FRACTIONONE;
if(ALSource->Params.Step == 0)
ALSource->Params.Step = 1;
}
Channels = ALBuffer->FmtChannels;
break;
}
BufferListItem = BufferListItem->next;
}
/* Calculate gains */
DryGain = SourceVolume;
DryGain = __min(DryGain,MaxVolume);
DryGain = __max(DryGain,MinVolume);
DryGainHF = 1.0f;
switch(ALSource->DirectFilter.type)
{
case AL_FILTER_LOWPASS:
DryGain *= ALSource->DirectFilter.Gain;
DryGainHF *= ALSource->DirectFilter.GainHF;
break;
}
for(i = 0;i < MAXCHANNELS;i++)
{
ALuint i2;
for(i2 = 0;i2 < MAXCHANNELS;i2++)
ALSource->Params.DryGains[i][i2] = 0.0f;
}
switch(Channels)
{
case FmtMono:
ALSource->Params.DryGains[0][FRONT_CENTER] = DryGain * ListenerGain;
break;
case FmtStereo:
if(DupStereo == AL_FALSE)
{
ALSource->Params.DryGains[0][FRONT_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][FRONT_RIGHT] = DryGain * ListenerGain;
}
else
{
switch(DevChans)
{
case DevFmtMono:
case DevFmtStereo:
ALSource->Params.DryGains[0][FRONT_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][FRONT_RIGHT] = DryGain * ListenerGain;
break;
case DevFmtQuad:
case DevFmtX51:
DryGain *= aluSqrt(2.0f/4.0f);
ALSource->Params.DryGains[0][FRONT_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][FRONT_RIGHT] = DryGain * ListenerGain;
ALSource->Params.DryGains[0][BACK_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][BACK_RIGHT] = DryGain * ListenerGain;
break;
case DevFmtX61:
DryGain *= aluSqrt(2.0f/4.0f);
ALSource->Params.DryGains[0][FRONT_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][FRONT_RIGHT] = DryGain * ListenerGain;
ALSource->Params.DryGains[0][SIDE_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][SIDE_RIGHT] = DryGain * ListenerGain;
break;
case DevFmtX71:
DryGain *= aluSqrt(2.0f/6.0f);
ALSource->Params.DryGains[0][FRONT_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][FRONT_RIGHT] = DryGain * ListenerGain;
ALSource->Params.DryGains[0][BACK_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][BACK_RIGHT] = DryGain * ListenerGain;
ALSource->Params.DryGains[0][SIDE_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][SIDE_RIGHT] = DryGain * ListenerGain;
break;
}
}
break;
case FmtRear:
ALSource->Params.DryGains[0][BACK_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][BACK_RIGHT] = DryGain * ListenerGain;
break;
case FmtQuad:
ALSource->Params.DryGains[0][FRONT_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][FRONT_RIGHT] = DryGain * ListenerGain;
ALSource->Params.DryGains[2][BACK_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[3][BACK_RIGHT] = DryGain * ListenerGain;
break;
case FmtX51:
ALSource->Params.DryGains[0][FRONT_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][FRONT_RIGHT] = DryGain * ListenerGain;
ALSource->Params.DryGains[2][FRONT_CENTER] = DryGain * ListenerGain;
ALSource->Params.DryGains[3][LFE] = DryGain * ListenerGain;
ALSource->Params.DryGains[4][BACK_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[5][BACK_RIGHT] = DryGain * ListenerGain;
break;
case FmtX61:
ALSource->Params.DryGains[0][FRONT_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][FRONT_RIGHT] = DryGain * ListenerGain;
ALSource->Params.DryGains[2][FRONT_CENTER] = DryGain * ListenerGain;
ALSource->Params.DryGains[3][LFE] = DryGain * ListenerGain;
ALSource->Params.DryGains[4][BACK_CENTER] = DryGain * ListenerGain;
ALSource->Params.DryGains[5][SIDE_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[6][SIDE_RIGHT] = DryGain * ListenerGain;
break;
case FmtX71:
ALSource->Params.DryGains[0][FRONT_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[1][FRONT_RIGHT] = DryGain * ListenerGain;
ALSource->Params.DryGains[2][FRONT_CENTER] = DryGain * ListenerGain;
ALSource->Params.DryGains[3][LFE] = DryGain * ListenerGain;
ALSource->Params.DryGains[4][BACK_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[5][BACK_RIGHT] = DryGain * ListenerGain;
ALSource->Params.DryGains[6][SIDE_LEFT] = DryGain * ListenerGain;
ALSource->Params.DryGains[7][SIDE_RIGHT] = DryGain * ListenerGain;
break;
}
for(i = 0;i < NumSends;i++)
{
WetGain[i] = SourceVolume;
WetGain[i] = __min(WetGain[i],MaxVolume);
WetGain[i] = __max(WetGain[i],MinVolume);
WetGainHF[i] = 1.0f;
switch(ALSource->Send[i].WetFilter.type)
{
case AL_FILTER_LOWPASS:
WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
break;
}
ALSource->Params.Send[i].WetGain = WetGain[i] * ListenerGain;
}
/* Update filter coefficients. Calculations based on the I3DL2
* spec. */
cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
/* We use two chained one-pole filters, so we need to take the
* square root of the squared gain, which is the same as the base
* gain. */
ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
for(i = 0;i < NumSends;i++)
{
/* We use a one-pole filter, so we need to take the squared gain */
ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
ALSource->Params.Send[i].iirFilter.coeff = a;
}
}
ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
{
const ALCdevice *Device = ALContext->Device;
ALfloat InnerAngle,OuterAngle,Angle,Distance,OrigDist;
ALfloat Direction[3],Position[3],SourceToListener[3];
ALfloat Velocity[3],ListenerVel[3];
ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff,OuterGainHF;
ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
ALfloat DopplerFactor, DopplerVelocity, SpeedOfSound;
ALfloat AirAbsorptionFactor;
ALbufferlistitem *BufferListItem;
ALfloat Attenuation, EffectiveDist;
ALfloat RoomAttenuation[MAX_SENDS];
ALfloat MetersPerUnit;
ALfloat RoomRolloff[MAX_SENDS];
ALfloat DryGain;
ALfloat DryGainHF;
ALfloat WetGain[MAX_SENDS];
ALfloat WetGainHF[MAX_SENDS];
ALfloat DirGain, AmbientGain;
const ALfloat *SpeakerGain;
ALfloat Pitch;
ALfloat length;
ALuint Frequency;
ALint NumSends;
ALint pos, s, i;
ALfloat cw;
DryGainHF = 1.0f;
for(i = 0;i < MAX_SENDS;i++)
WetGainHF[i] = 1.0f;
//Get context properties
DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
DopplerVelocity = ALContext->DopplerVelocity;
SpeedOfSound = ALContext->flSpeedOfSound;
NumSends = Device->NumAuxSends;
Frequency = Device->Frequency;
//Get listener properties
ListenerGain = ALContext->Listener.Gain;
MetersPerUnit = ALContext->Listener.MetersPerUnit;
memcpy(ListenerVel, ALContext->Listener.Velocity, sizeof(ALContext->Listener.Velocity));
//Get source properties
SourceVolume = ALSource->flGain;
memcpy(Position, ALSource->vPosition, sizeof(ALSource->vPosition));
memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation));
memcpy(Velocity, ALSource->vVelocity, sizeof(ALSource->vVelocity));
MinVolume = ALSource->flMinGain;
MaxVolume = ALSource->flMaxGain;
MinDist = ALSource->flRefDistance;
MaxDist = ALSource->flMaxDistance;
Rolloff = ALSource->flRollOffFactor;
InnerAngle = ALSource->flInnerAngle;
OuterAngle = ALSource->flOuterAngle;
OuterGainHF = ALSource->OuterGainHF;
AirAbsorptionFactor = ALSource->AirAbsorptionFactor;
//1. Translate Listener to origin (convert to head relative)
if(ALSource->bHeadRelative == AL_FALSE)
{
ALfloat U[3],V[3],N[3];
ALfloat Matrix[4][4];
// Build transform matrix
memcpy(N, ALContext->Listener.Forward, sizeof(N)); // At-vector
aluNormalize(N); // Normalized At-vector
memcpy(V, ALContext->Listener.Up, sizeof(V)); // Up-vector
aluNormalize(V); // Normalized Up-vector
aluCrossproduct(N, V, U); // Right-vector
aluNormalize(U); // Normalized Right-vector
Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0]; Matrix[0][3] = 0.0f;
Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1]; Matrix[1][3] = 0.0f;
Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2]; Matrix[2][3] = 0.0f;
Matrix[3][0] = 0.0f; Matrix[3][1] = 0.0f; Matrix[3][2] = 0.0f; Matrix[3][3] = 1.0f;
// Translate position
Position[0] -= ALContext->Listener.Position[0];
Position[1] -= ALContext->Listener.Position[1];
Position[2] -= ALContext->Listener.Position[2];
// Transform source position and direction into listener space
aluMatrixVector(Position, 1.0f, Matrix);
aluMatrixVector(Direction, 0.0f, Matrix);
// Transform source and listener velocity into listener space
aluMatrixVector(Velocity, 0.0f, Matrix);
aluMatrixVector(ListenerVel, 0.0f, Matrix);
}
else
ListenerVel[0] = ListenerVel[1] = ListenerVel[2] = 0.0f;
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));
OrigDist = Distance;
Attenuation = 1.0f;
for(i = 0;i < NumSends;i++)
{
RoomAttenuation[i] = 1.0f;
RoomRolloff[i] = ALSource->RoomRolloffFactor;
if(ALSource->Send[i].Slot &&
(ALSource->Send[i].Slot->effect.type == AL_EFFECT_REVERB ||
ALSource->Send[i].Slot->effect.type == AL_EFFECT_EAXREVERB))
RoomRolloff[i] += ALSource->Send[i].Slot->effect.Reverb.RoomRolloffFactor;
}
switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel :
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)
Attenuation = MinDist / (MinDist + (Rolloff * (Distance - MinDist)));
for(i = 0;i < NumSends;i++)
{
if((MinDist + (RoomRolloff[i] * (Distance - MinDist))) > 0.0f)
RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (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:
if(MaxDist != MinDist)
{
Attenuation = 1.0f - (Rolloff*(Distance-MinDist)/(MaxDist - MinDist));
Attenuation = __max(Attenuation, 0.0f);
for(i = 0;i < NumSends;i++)
{
RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(Distance-MinDist)/(MaxDist - MinDist));
RoomAttenuation[i] = __max(RoomAttenuation[i], 0.0f);
}
}
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)
{
Attenuation = aluPow(Distance/MinDist, -Rolloff);
for(i = 0;i < NumSends;i++)
RoomAttenuation[i] = aluPow(Distance/MinDist, -RoomRolloff[i]);
}
break;
case AL_NONE:
break;
}
// Source Gain + Attenuation
DryGain = SourceVolume * Attenuation;
for(i = 0;i < NumSends;i++)
WetGain[i] = SourceVolume * RoomAttenuation[i];
EffectiveDist = 0.0f;
if(MinDist > 0.0f && Attenuation < 1.0f)
EffectiveDist = (MinDist/Attenuation - MinDist)*MetersPerUnit;
// Distance-based air absorption
if(AirAbsorptionFactor > 0.0f && EffectiveDist > 0.0f)
{
ALfloat absorb;
// Absorption calculation is done in dB
absorb = (AirAbsorptionFactor*AIRABSORBGAINDBHF) *
EffectiveDist;
// Convert dB to linear gain before applying
absorb = aluPow(10.0f, absorb/20.0f);
DryGainHF *= absorb;
}
//3. Apply directional soundcones
Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f/M_PI;
if(Angle >= InnerAngle && Angle <= OuterAngle)
{
ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)*scale);
ConeHF = (1.0f+(OuterGainHF-1.0f)*scale);
}
else if(Angle > OuterAngle)
{
ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f));
ConeHF = (1.0f+(OuterGainHF-1.0f));
}
else
{
ConeVolume = 1.0f;
ConeHF = 1.0f;
}
// Apply some high-frequency attenuation for sources behind the listener
// NOTE: This should be aluDotproduct({0,0,-1}, ListenerToSource), however
// that is equivalent to aluDotproduct({0,0,1}, SourceToListener), which is
// the same as SourceToListener[2]
Angle = aluAcos(SourceToListener[2]) * 180.0f/M_PI;
// Sources within the minimum distance attenuate less
if(OrigDist < MinDist)
Angle *= OrigDist/MinDist;
if(Angle > 90.0f)
{
ALfloat scale = (Angle-90.0f) / (180.1f-90.0f); // .1 to account for fp errors
ConeHF *= 1.0f - (Device->HeadDampen*scale);
}
DryGain *= ConeVolume;
if(ALSource->DryGainHFAuto)
DryGainHF *= ConeHF;
// Clamp to Min/Max Gain
DryGain = __min(DryGain,MaxVolume);
DryGain = __max(DryGain,MinVolume);
for(i = 0;i < NumSends;i++)
{
ALeffectslot *Slot = ALSource->Send[i].Slot;
if(!Slot || Slot->effect.type == AL_EFFECT_NULL)
{
ALSource->Params.Send[i].WetGain = 0.0f;
WetGainHF[i] = 1.0f;
continue;
}
if(Slot->AuxSendAuto)
{
if(ALSource->WetGainAuto)
WetGain[i] *= ConeVolume;
if(ALSource->WetGainHFAuto)
WetGainHF[i] *= ConeHF;
// Clamp to Min/Max Gain
WetGain[i] = __min(WetGain[i],MaxVolume);
WetGain[i] = __max(WetGain[i],MinVolume);
if(Slot->effect.type == AL_EFFECT_REVERB ||
Slot->effect.type == AL_EFFECT_EAXREVERB)
{
/* Apply a decay-time transformation to the wet path, based on
* the attenuation of the dry path.
*
* Using the approximate (effective) source to listener
* distance, the initial decay of the reverb effect is
* calculated and applied to the wet path.
*/
WetGain[i] *= aluPow(10.0f, EffectiveDist /
(SPEEDOFSOUNDMETRESPERSEC *
Slot->effect.Reverb.DecayTime) *
-60.0 / 20.0);
WetGainHF[i] *= aluPow(Slot->effect.Reverb.AirAbsorptionGainHF,
AirAbsorptionFactor * EffectiveDist);
}
}
else
{
/* If the slot's auxiliary send auto is off, the data sent to the
* effect slot is the same as the dry path, sans filter effects */
WetGain[i] = DryGain;
WetGainHF[i] = DryGainHF;
}
switch(ALSource->Send[i].WetFilter.type)
{
case AL_FILTER_LOWPASS:
WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
break;
}
ALSource->Params.Send[i].WetGain = WetGain[i] * ListenerGain;
}
// Apply filter gains and filters
switch(ALSource->DirectFilter.type)
{
case AL_FILTER_LOWPASS:
DryGain *= ALSource->DirectFilter.Gain;
DryGainHF *= ALSource->DirectFilter.GainHF;
break;
}
DryGain *= ListenerGain;
// Calculate Velocity
Pitch = ALSource->flPitch;
if(DopplerFactor != 0.0f)
{
ALfloat VSS, VLS;
ALfloat MaxVelocity = (SpeedOfSound*DopplerVelocity) /
DopplerFactor;
VSS = aluDotproduct(Velocity, SourceToListener);
if(VSS >= MaxVelocity)
VSS = (MaxVelocity - 1.0f);
else if(VSS <= -MaxVelocity)
VSS = -MaxVelocity + 1.0f;
VLS = aluDotproduct(ListenerVel, SourceToListener);
if(VLS >= MaxVelocity)
VLS = (MaxVelocity - 1.0f);
else if(VLS <= -MaxVelocity)
VLS = -MaxVelocity + 1.0f;
Pitch *= ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VLS)) /
((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VSS));
}
BufferListItem = ALSource->queue;
while(BufferListItem != NULL)
{
ALbuffer *ALBuffer;
if((ALBuffer=BufferListItem->buffer) != NULL)
{
ALint maxstep = STACK_DATA_SIZE / FrameSizeFromFmt(ALBuffer->FmtChannels,
ALBuffer->FmtType);
maxstep -= ResamplerPadding[ALSource->Resampler] +
ResamplerPrePadding[ALSource->Resampler] + 1;
maxstep = min(maxstep, INT_MAX>>FRACTIONBITS);
Pitch = Pitch * ALBuffer->Frequency / Frequency;
if(Pitch > (ALfloat)maxstep)
ALSource->Params.Step = maxstep<<FRACTIONBITS;
else
{
ALSource->Params.Step = Pitch*FRACTIONONE;
if(ALSource->Params.Step == 0)
ALSource->Params.Step = 1;
}
break;
}
BufferListItem = BufferListItem->next;
}
// Use energy-preserving panning algorithm for multi-speaker playback
length = __max(OrigDist, MinDist);
if(length > 0.0f)
{
ALfloat invlen = 1.0f/length;
Position[0] *= invlen;
Position[1] *= invlen;
Position[2] *= invlen;
}
pos = aluCart2LUTpos(-Position[2], Position[0]);
SpeakerGain = &Device->PanningLUT[MAXCHANNELS * pos];
DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
// elevation adjustment for directional gain. this sucks, but
// has low complexity
AmbientGain = aluSqrt(1.0/Device->NumChan);
for(s = 0;s < MAXCHANNELS;s++)
{
ALuint s2;
for(s2 = 0;s2 < MAXCHANNELS;s2++)
ALSource->Params.DryGains[s][s2] = 0.0f;
}
for(s = 0;s < (ALsizei)Device->NumChan;s++)
{
Channel chan = Device->Speaker2Chan[s];
ALfloat gain = AmbientGain + (SpeakerGain[chan]-AmbientGain)*DirGain;
ALSource->Params.DryGains[0][chan] = DryGain * gain;
}
/* Update filter coefficients. */
cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
/* Spatialized sources 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. */
ALSource->Params.iirFilter.coeff = lpCoeffCalc(aluSqrt(DryGainHF), cw);
for(i = 0;i < NumSends;i++)
{
/* The wet path uses two chained one-pole filters, so take the
* base gain (square root of the squared gain) */
ALSource->Params.Send[i].iirFilter.coeff = lpCoeffCalc(WetGainHF[i], cw);
}
}
static __inline ALfloat aluF2F(ALfloat val)
{
return val;
}
static __inline ALushort aluF2US(ALfloat val)
{
if(val > 1.0f) return 65535;
if(val < -1.0f) return 0;
return (ALint)(val*32767.0f) + 32768;
}
static __inline ALshort aluF2S(ALfloat val)
{
if(val > 1.0f) return 32767;
if(val < -1.0f) return -32768;
return (ALint)(val*32767.0f);
}
static __inline ALubyte aluF2UB(ALfloat val)
{
ALushort i = aluF2US(val);
return i>>8;
}
static __inline ALbyte aluF2B(ALfloat val)
{
ALshort i = aluF2S(val);
return i>>8;
}
static const Channel MonoChans[] = { FRONT_CENTER };
static const Channel StereoChans[] = { FRONT_LEFT, FRONT_RIGHT };
static const Channel QuadChans[] = { FRONT_LEFT, FRONT_RIGHT,
BACK_LEFT, BACK_RIGHT };
static const Channel X51Chans[] = { FRONT_LEFT, FRONT_RIGHT,
FRONT_CENTER, LFE,
BACK_LEFT, BACK_RIGHT };
static const Channel X61Chans[] = { FRONT_LEFT, FRONT_LEFT,
FRONT_CENTER, LFE, BACK_CENTER,
SIDE_LEFT, SIDE_RIGHT };
static const Channel X71Chans[] = { FRONT_LEFT, FRONT_RIGHT,
FRONT_CENTER, LFE,
BACK_LEFT, BACK_RIGHT,
SIDE_LEFT, SIDE_RIGHT };
#define DECL_TEMPLATE(T, chans,N, func) \
static void Write_##T##_##chans(ALCdevice *device, T *buffer, ALuint SamplesToDo)\
{ \
ALfloat (*DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
ALfloat (*Matrix)[MAXCHANNELS] = device->ChannelMatrix; \
const ALuint *ChanMap = device->DevChannels; \
ALuint i, j, c; \
\
for(i = 0;i < SamplesToDo;i++) \
{ \
for(j = 0;j < N;j++) \
{ \
ALfloat samp = 0.0f; \
for(c = 0;c < MAXCHANNELS;c++) \
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
((T*)buffer)[ChanMap[chans[j]]] = func(samp); \
} \
buffer = ((T*)buffer) + N; \
} \
}
DECL_TEMPLATE(ALfloat, MonoChans,1, aluF2F)
DECL_TEMPLATE(ALfloat, QuadChans,4, aluF2F)
DECL_TEMPLATE(ALfloat, X51Chans,6, aluF2F)
DECL_TEMPLATE(ALfloat, X61Chans,7, aluF2F)
DECL_TEMPLATE(ALfloat, X71Chans,8, aluF2F)
DECL_TEMPLATE(ALushort, MonoChans,1, aluF2US)
DECL_TEMPLATE(ALushort, QuadChans,4, aluF2US)
DECL_TEMPLATE(ALushort, X51Chans,6, aluF2US)
DECL_TEMPLATE(ALushort, X61Chans,7, aluF2US)
DECL_TEMPLATE(ALushort, X71Chans,8, aluF2US)
DECL_TEMPLATE(ALshort, MonoChans,1, aluF2S)
DECL_TEMPLATE(ALshort, QuadChans,4, aluF2S)
DECL_TEMPLATE(ALshort, X51Chans,6, aluF2S)
DECL_TEMPLATE(ALshort, X61Chans,7, aluF2S)
DECL_TEMPLATE(ALshort, X71Chans,8, aluF2S)
DECL_TEMPLATE(ALubyte, MonoChans,1, aluF2UB)
DECL_TEMPLATE(ALubyte, QuadChans,4, aluF2UB)
DECL_TEMPLATE(ALubyte, X51Chans,6, aluF2UB)
DECL_TEMPLATE(ALubyte, X61Chans,7, aluF2UB)
DECL_TEMPLATE(ALubyte, X71Chans,8, aluF2UB)
DECL_TEMPLATE(ALbyte, MonoChans,1, aluF2B)
DECL_TEMPLATE(ALbyte, QuadChans,4, aluF2B)
DECL_TEMPLATE(ALbyte, X51Chans,6, aluF2B)
DECL_TEMPLATE(ALbyte, X61Chans,7, aluF2B)
DECL_TEMPLATE(ALbyte, X71Chans,8, aluF2B)
#undef DECL_TEMPLATE
#define DECL_TEMPLATE(T, chans,N, func) \
static void Write_##T##_##chans(ALCdevice *device, T *buffer, ALuint SamplesToDo)\
{ \
ALfloat (*DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
ALfloat (*Matrix)[MAXCHANNELS] = device->ChannelMatrix; \
const ALuint *ChanMap = device->DevChannels; \
ALuint i, j, c; \
\
if(device->Bs2b) \
{ \
for(i = 0;i < SamplesToDo;i++) \
{ \
float samples[2] = { 0.0f, 0.0f }; \
for(c = 0;c < MAXCHANNELS;c++) \
{ \
samples[0] += DryBuffer[i][c]*Matrix[c][FRONT_LEFT]; \
samples[1] += DryBuffer[i][c]*Matrix[c][FRONT_RIGHT]; \
} \
bs2b_cross_feed(device->Bs2b, samples); \
((T*)buffer)[ChanMap[FRONT_LEFT]] = func(samples[0]); \
((T*)buffer)[ChanMap[FRONT_RIGHT]] = func(samples[1]); \
buffer = ((T*)buffer) + 2; \
} \
} \
else \
{ \
for(i = 0;i < SamplesToDo;i++) \
{ \
for(j = 0;j < N;j++) \
{ \
ALfloat samp = 0.0f; \
for(c = 0;c < MAXCHANNELS;c++) \
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
((T*)buffer)[ChanMap[chans[j]]] = func(samp); \
} \
buffer = ((T*)buffer) + N; \
} \
} \
}
DECL_TEMPLATE(ALfloat, StereoChans,2, aluF2F)
DECL_TEMPLATE(ALushort, StereoChans,2, aluF2US)
DECL_TEMPLATE(ALshort, StereoChans,2, aluF2S)
DECL_TEMPLATE(ALubyte, StereoChans,2, aluF2UB)
DECL_TEMPLATE(ALbyte, StereoChans,2, aluF2B)
#undef DECL_TEMPLATE
#define DECL_TEMPLATE(T, func) \
static void Write_##T(ALCdevice *device, T *buffer, ALuint SamplesToDo) \
{ \
switch(device->FmtChans) \
{ \
case DevFmtMono: \
Write_##T##_MonoChans(device, buffer, SamplesToDo); \
break; \
case DevFmtStereo: \
Write_##T##_StereoChans(device, buffer, SamplesToDo); \
break; \
case DevFmtQuad: \
Write_##T##_QuadChans(device, buffer, SamplesToDo); \
break; \
case DevFmtX51: \
Write_##T##_X51Chans(device, buffer, SamplesToDo); \
break; \
case DevFmtX61: \
Write_##T##_X61Chans(device, buffer, SamplesToDo); \
break; \
case DevFmtX71: \
Write_##T##_X71Chans(device, buffer, SamplesToDo); \
break; \
} \
}
DECL_TEMPLATE(ALfloat, aluF2F)
DECL_TEMPLATE(ALushort, aluF2US)
DECL_TEMPLATE(ALshort, aluF2S)
DECL_TEMPLATE(ALubyte, aluF2UB)
DECL_TEMPLATE(ALbyte, aluF2B)
#undef DECL_TEMPLATE
ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
{
ALuint SamplesToDo;
ALeffectslot *ALEffectSlot;
ALCcontext **ctx, **ctx_end;
ALsource **src, **src_end;
int fpuState;
ALuint i, c;
ALsizei e;
#if defined(HAVE_FESETROUND)
fpuState = fegetround();
fesetround(FE_TOWARDZERO);
#elif defined(HAVE__CONTROLFP)
fpuState = _controlfp(_RC_CHOP, _MCW_RC);
#else
(void)fpuState;
#endif
while(size > 0)
{
/* Setup variables */
SamplesToDo = min(size, BUFFERSIZE);
/* Clear mixing buffer */
memset(device->DryBuffer, 0, SamplesToDo*MAXCHANNELS*sizeof(ALfloat));
SuspendContext(NULL);
ctx = device->Contexts;
ctx_end = ctx + device->NumContexts;
while(ctx != ctx_end)
{
SuspendContext(*ctx);
src = (*ctx)->ActiveSources;
src_end = src + (*ctx)->ActiveSourceCount;
while(src != src_end)
{
if((*src)->state != AL_PLAYING)
{
--((*ctx)->ActiveSourceCount);
*src = *(--src_end);
continue;
}
if((*src)->NeedsUpdate)
{
ALsource_Update(*src, *ctx);
(*src)->NeedsUpdate = AL_FALSE;
}
MixSource(*src, device, SamplesToDo);
src++;
}
/* effect slot processing */
for(e = 0;e < (*ctx)->EffectSlotMap.size;e++)
{
ALEffectSlot = (*ctx)->EffectSlotMap.array[e].value;
for(i = 0;i < SamplesToDo;i++)
{
ALEffectSlot->ClickRemoval[0] -= ALEffectSlot->ClickRemoval[0] / 256.0f;
ALEffectSlot->WetBuffer[i] += ALEffectSlot->ClickRemoval[0];
}
for(i = 0;i < 1;i++)
{
ALEffectSlot->ClickRemoval[i] += ALEffectSlot->PendingClicks[i];
ALEffectSlot->PendingClicks[i] = 0.0f;
}
ALEffect_Process(ALEffectSlot->EffectState, ALEffectSlot,
SamplesToDo, ALEffectSlot->WetBuffer,
device->DryBuffer);
for(i = 0;i < SamplesToDo;i++)
ALEffectSlot->WetBuffer[i] = 0.0f;
}
ProcessContext(*ctx);
ctx++;
}
ProcessContext(NULL);
//Post processing loop
for(i = 0;i < SamplesToDo;i++)
{
for(c = 0;c < MAXCHANNELS;c++)
{
device->ClickRemoval[c] -= device->ClickRemoval[c] / 256.0f;
device->DryBuffer[i][c] += device->ClickRemoval[c];
}
}
for(i = 0;i < MAXCHANNELS;i++)
{
device->ClickRemoval[i] += device->PendingClicks[i];
device->PendingClicks[i] = 0.0f;
}
switch(device->FmtType)
{
case DevFmtByte:
Write_ALbyte(device, buffer, SamplesToDo);
break;
case DevFmtUByte:
Write_ALubyte(device, buffer, SamplesToDo);
break;
case DevFmtShort:
Write_ALshort(device, buffer, SamplesToDo);
break;
case DevFmtUShort:
Write_ALushort(device, buffer, SamplesToDo);
break;
case DevFmtFloat:
Write_ALfloat(device, buffer, SamplesToDo);
break;
}
size -= SamplesToDo;
}
#if defined(HAVE_FESETROUND)
fesetround(fpuState);
#elif defined(HAVE__CONTROLFP)
_controlfp(fpuState, _MCW_RC);
#endif
}
ALvoid aluHandleDisconnect(ALCdevice *device)
{
ALuint i;
SuspendContext(NULL);
for(i = 0;i < device->NumContexts;i++)
{
ALCcontext *Context = device->Contexts[i];
ALsource *source;
ALsizei pos;
SuspendContext(Context);
for(pos = 0;pos < Context->SourceMap.size;pos++)
{
source = Context->SourceMap.array[pos].value;
if(source->state == AL_PLAYING)
{
source->state = AL_STOPPED;
source->BuffersPlayed = source->BuffersInQueue;
source->position = 0;
source->position_fraction = 0;
}
}
ProcessContext(Context);
}
device->Connected = ALC_FALSE;
ProcessContext(NULL);
}