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AuroraOpenALSoft/Alc/ALu.c
Chris Robinson 1fb9311d82 Lock the device before calling aluHandleDisconnect 
PulseAudio causes an assert if being relocked inside a callback on the worker
thread, where aluHandleDisconnect is called. We can assume it's already locked
there, so just make sure the device is locked before being calling it.
2012-12-02 11:30:23 -08:00

1197 lines
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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 "alSource.h"
#include "alBuffer.h"
#include "alListener.h"
#include "alAuxEffectSlot.h"
#include "alu.h"
#include "bs2b.h"
#include "mixer_defs.h"
struct ChanMap {
enum Channel channel;
ALfloat angle;
};
/* Cone scalar */
ALfloat ConeScale = 1.0f;
/* Localized Z scalar for mono sources */
ALfloat ZScale = 1.0f;
static ResamplerFunc SelectResampler(enum Resampler Resampler, ALuint increment)
{
if(increment == FRACTIONONE)
return Resample_copy32_C;
switch(Resampler)
{
case PointResampler:
return Resample_point32_C;
case LinearResampler:
return Resample_lerp32_C;
case CubicResampler:
return Resample_cubic32_C;
case ResamplerMax:
/* Shouldn't happen */
break;
}
return Resample_point32_C;
}
static DryMixerFunc SelectHrtfMixer(void)
{
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return MixDirect_Hrtf_SSE;
#endif
#ifdef HAVE_NEON
if((CPUCapFlags&CPU_CAP_NEON))
return MixDirect_Hrtf_Neon;
#endif
return MixDirect_Hrtf_C;
}
static DryMixerFunc SelectDirectMixer(void)
{
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return MixDirect_SSE;
#endif
return MixDirect_C;
}
static WetMixerFunc SelectSendMixer(void)
{
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return MixSend_SSE;
#endif
return MixSend_C;
}
static __inline void 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 void aluNormalize(ALfloat *inVector)
{
ALfloat lengthsqr = aluDotproduct(inVector, inVector);
if(lengthsqr > 0.0f)
{
ALfloat inv_length = 1.0f/sqrtf(lengthsqr);
inVector[0] *= inv_length;
inVector[1] *= inv_length;
inVector[2] *= inv_length;
}
}
static __inline ALvoid aluMatrixVector(ALfloat *vector, ALfloat w, ALfloat (*RESTRICT matrix)[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];
}
static ALvoid CalcListenerParams(ALlistener *Listener)
{
ALfloat N[3], V[3], U[3], P[3];
/* AT then UP */
N[0] = Listener->Forward[0];
N[1] = Listener->Forward[1];
N[2] = Listener->Forward[2];
aluNormalize(N);
V[0] = Listener->Up[0];
V[1] = Listener->Up[1];
V[2] = Listener->Up[2];
aluNormalize(V);
/* Build and normalize right-vector */
aluCrossproduct(N, V, U);
aluNormalize(U);
Listener->Params.Matrix[0][0] = U[0];
Listener->Params.Matrix[0][1] = V[0];
Listener->Params.Matrix[0][2] = -N[0];
Listener->Params.Matrix[0][3] = 0.0f;
Listener->Params.Matrix[1][0] = U[1];
Listener->Params.Matrix[1][1] = V[1];
Listener->Params.Matrix[1][2] = -N[1];
Listener->Params.Matrix[1][3] = 0.0f;
Listener->Params.Matrix[2][0] = U[2];
Listener->Params.Matrix[2][1] = V[2];
Listener->Params.Matrix[2][2] = -N[2];
Listener->Params.Matrix[2][3] = 0.0f;
Listener->Params.Matrix[3][0] = 0.0f;
Listener->Params.Matrix[3][1] = 0.0f;
Listener->Params.Matrix[3][2] = 0.0f;
Listener->Params.Matrix[3][3] = 1.0f;
P[0] = Listener->Position[0];
P[1] = Listener->Position[1];
P[2] = Listener->Position[2];
aluMatrixVector(P, 1.0f, Listener->Params.Matrix);
Listener->Params.Matrix[3][0] = -P[0];
Listener->Params.Matrix[3][1] = -P[1];
Listener->Params.Matrix[3][2] = -P[2];
Listener->Params.Velocity[0] = Listener->Velocity[0];
Listener->Params.Velocity[1] = Listener->Velocity[1];
Listener->Params.Velocity[2] = Listener->Velocity[2];
aluMatrixVector(Listener->Params.Velocity, 0.0f, Listener->Params.Matrix);
}
ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
{
static const struct ChanMap MonoMap[1] = { { FrontCenter, 0.0f } };
static const struct ChanMap StereoMap[2] = {
{ FrontLeft, -30.0f * F_PI/180.0f },
{ FrontRight, 30.0f * F_PI/180.0f }
};
static const struct ChanMap StereoWideMap[2] = {
{ FrontLeft, -90.0f * F_PI/180.0f },
{ FrontRight, 90.0f * F_PI/180.0f }
};
static const struct ChanMap RearMap[2] = {
{ BackLeft, -150.0f * F_PI/180.0f },
{ BackRight, 150.0f * F_PI/180.0f }
};
static const struct ChanMap QuadMap[4] = {
{ FrontLeft, -45.0f * F_PI/180.0f },
{ FrontRight, 45.0f * F_PI/180.0f },
{ BackLeft, -135.0f * F_PI/180.0f },
{ BackRight, 135.0f * F_PI/180.0f }
};
static const struct ChanMap X51Map[6] = {
{ FrontLeft, -30.0f * F_PI/180.0f },
{ FrontRight, 30.0f * F_PI/180.0f },
{ FrontCenter, 0.0f * F_PI/180.0f },
{ LFE, 0.0f },
{ BackLeft, -110.0f * F_PI/180.0f },
{ BackRight, 110.0f * F_PI/180.0f }
};
static const struct ChanMap X61Map[7] = {
{ FrontLeft, -30.0f * F_PI/180.0f },
{ FrontRight, 30.0f * F_PI/180.0f },
{ FrontCenter, 0.0f * F_PI/180.0f },
{ LFE, 0.0f },
{ BackCenter, 180.0f * F_PI/180.0f },
{ SideLeft, -90.0f * F_PI/180.0f },
{ SideRight, 90.0f * F_PI/180.0f }
};
static const struct ChanMap X71Map[8] = {
{ FrontLeft, -30.0f * F_PI/180.0f },
{ FrontRight, 30.0f * F_PI/180.0f },
{ FrontCenter, 0.0f * F_PI/180.0f },
{ LFE, 0.0f },
{ BackLeft, -150.0f * F_PI/180.0f },
{ BackRight, 150.0f * F_PI/180.0f },
{ SideLeft, -90.0f * F_PI/180.0f },
{ SideRight, 90.0f * F_PI/180.0f }
};
ALCdevice *Device = ALContext->Device;
ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
ALbufferlistitem *BufferListItem;
enum FmtChannels Channels;
ALfloat (*SrcMatrix)[MaxChannels];
ALfloat DryGain, DryGainHF;
ALfloat WetGain[MAX_SENDS];
ALfloat WetGainHF[MAX_SENDS];
ALint NumSends, Frequency;
const struct ChanMap *chans = NULL;
enum Resampler Resampler;
ALint num_channels = 0;
ALboolean DirectChannels;
ALfloat hwidth = 0.0f;
ALfloat Pitch;
ALfloat cw;
ALint i, c;
/* Get device properties */
NumSends = Device->NumAuxSends;
Frequency = Device->Frequency;
/* Get listener properties */
ListenerGain = ALContext->Listener->Gain;
/* Get source properties */
SourceVolume = ALSource->Gain;
MinVolume = ALSource->MinGain;
MaxVolume = ALSource->MaxGain;
Pitch = ALSource->Pitch;
Resampler = ALSource->Resampler;
DirectChannels = ALSource->DirectChannels;
/* Calculate the stepping value */
Channels = FmtMono;
BufferListItem = ALSource->queue;
while(BufferListItem != NULL)
{
ALbuffer *ALBuffer;
if((ALBuffer=BufferListItem->buffer) != NULL)
{
ALsizei maxstep = BUFFERSIZE;
maxstep -= ResamplerPadding[Resampler] +
ResamplerPrePadding[Resampler] + 1;
maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
Pitch = Pitch * ALBuffer->Frequency / Frequency;
if(Pitch > (ALfloat)maxstep)
ALSource->Params.Step = maxstep<<FRACTIONBITS;
else
{
ALSource->Params.Step = fastf2i(Pitch*FRACTIONONE);
if(ALSource->Params.Step == 0)
ALSource->Params.Step = 1;
}
ALSource->Params.Resample = SelectResampler(Resampler, ALSource->Params.Step);
Channels = ALBuffer->FmtChannels;
break;
}
BufferListItem = BufferListItem->next;
}
if(!DirectChannels && Device->Hrtf)
ALSource->Params.DryMix = SelectHrtfMixer();
else
ALSource->Params.DryMix = SelectDirectMixer();
ALSource->Params.WetMix = SelectSendMixer();
/* Calculate gains */
DryGain = clampf(SourceVolume, MinVolume, MaxVolume);
DryGain *= ALSource->DirectGain * ListenerGain;
DryGainHF = ALSource->DirectGainHF;
for(i = 0;i < NumSends;i++)
{
WetGain[i] = clampf(SourceVolume, MinVolume, MaxVolume);
WetGain[i] *= ALSource->Send[i].Gain * ListenerGain;
WetGainHF[i] = ALSource->Send[i].GainHF;
}
SrcMatrix = ALSource->Params.Direct.Gains;
for(i = 0;i < MaxChannels;i++)
{
for(c = 0;c < MaxChannels;c++)
SrcMatrix[i][c] = 0.0f;
}
switch(Channels)
{
case FmtMono:
chans = MonoMap;
num_channels = 1;
break;
case FmtStereo:
if(!(Device->Flags&DEVICE_WIDE_STEREO))
chans = StereoMap;
else
{
chans = StereoWideMap;
hwidth = 60.0f * F_PI/180.0f;
}
num_channels = 2;
break;
case FmtRear:
chans = RearMap;
num_channels = 2;
break;
case FmtQuad:
chans = QuadMap;
num_channels = 4;
break;
case FmtX51:
chans = X51Map;
num_channels = 6;
break;
case FmtX61:
chans = X61Map;
num_channels = 7;
break;
case FmtX71:
chans = X71Map;
num_channels = 8;
break;
}
if(DirectChannels != AL_FALSE)
{
for(c = 0;c < num_channels;c++)
{
for(i = 0;i < (ALint)Device->NumChan;i++)
{
enum Channel chan = Device->Speaker2Chan[i];
if(chan == chans[c].channel)
{
SrcMatrix[c][chan] = DryGain;
break;
}
}
}
}
else if(Device->Hrtf)
{
for(c = 0;c < num_channels;c++)
{
if(chans[c].channel == LFE)
{
/* Skip LFE */
ALSource->Params.Direct.Hrtf.Params.Delay[c][0] = 0;
ALSource->Params.Direct.Hrtf.Params.Delay[c][1] = 0;
for(i = 0;i < HRIR_LENGTH;i++)
{
ALSource->Params.Direct.Hrtf.Params.Coeffs[c][i][0] = 0.0f;
ALSource->Params.Direct.Hrtf.Params.Coeffs[c][i][1] = 0.0f;
}
}
else
{
/* Get the static HRIR coefficients and delays for this
* channel. */
GetLerpedHrtfCoeffs(Device->Hrtf,
0.0f, chans[c].angle, DryGain,
ALSource->Params.Direct.Hrtf.Params.Coeffs[c],
ALSource->Params.Direct.Hrtf.Params.Delay[c]);
}
}
ALSource->Hrtf.Counter = 0;
ALSource->Params.Direct.Hrtf.Params.IrSize = GetHrtfIrSize(Device->Hrtf);
ALSource->Params.Direct.Hrtf.State = &ALSource->Hrtf;
}
else
{
DryGain *= lerp(1.0f, 1.0f/sqrtf((float)Device->NumChan), hwidth/F_PI);
for(c = 0;c < num_channels;c++)
{
/* Special-case LFE */
if(chans[c].channel == LFE)
{
SrcMatrix[c][chans[c].channel] = DryGain;
continue;
}
ComputeAngleGains(Device, chans[c].angle, hwidth, DryGain,
SrcMatrix[c]);
}
}
ALSource->Params.Direct.OutBuffer = Device->DryBuffer;
ALSource->Params.Direct.ClickRemoval = Device->ClickRemoval;
ALSource->Params.Direct.PendingClicks = Device->PendingClicks;
for(i = 0;i < NumSends;i++)
{
ALeffectslot *Slot = ALSource->Send[i].Slot;
if(!Slot && i == 0)
Slot = Device->DefaultSlot;
if(Slot && Slot->effect.type == AL_EFFECT_NULL)
Slot = NULL;
ALSource->Params.Send[i].Slot = Slot;
ALSource->Params.Send[i].Gain = WetGain[i];
}
/* Update filter coefficients. Calculations based on the I3DL2
* spec. */
cw = cosf(F_PI*2.0f * LOWPASSFREQREF / 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.Direct.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
for(i = 0;i < NumSends;i++)
{
ALfloat a = lpCoeffCalc(WetGainHF[i], cw);
ALSource->Params.Send[i].iirFilter.coeff = a;
}
}
ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
{
ALCdevice *Device = ALContext->Device;
ALfloat Velocity[3],Direction[3],Position[3],SourceToListener[3];
ALfloat InnerAngle,OuterAngle,Angle,Distance,ClampedDist;
ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff;
ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
ALfloat DopplerFactor, SpeedOfSound;
ALfloat AirAbsorptionFactor;
ALfloat RoomAirAbsorption[MAX_SENDS];
ALbufferlistitem *BufferListItem;
ALfloat Attenuation;
ALfloat RoomAttenuation[MAX_SENDS];
ALfloat MetersPerUnit;
ALfloat RoomRolloffBase;
ALfloat RoomRolloff[MAX_SENDS];
ALfloat DecayDistance[MAX_SENDS];
ALfloat DryGain;
ALfloat DryGainHF;
ALboolean DryGainHFAuto;
ALfloat WetGain[MAX_SENDS];
ALfloat WetGainHF[MAX_SENDS];
ALboolean WetGainAuto;
ALboolean WetGainHFAuto;
enum Resampler Resampler;
ALfloat Pitch;
ALuint Frequency;
ALint NumSends;
ALfloat cw;
ALint i, j;
DryGainHF = 1.0f;
for(i = 0;i < MAX_SENDS;i++)
WetGainHF[i] = 1.0f;
/* Get context/device properties */
DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
SpeedOfSound = ALContext->SpeedOfSound * ALContext->DopplerVelocity;
NumSends = Device->NumAuxSends;
Frequency = Device->Frequency;
/* Get listener properties */
ListenerGain = ALContext->Listener->Gain;
MetersPerUnit = ALContext->Listener->MetersPerUnit;
/* Get source properties */
SourceVolume = ALSource->Gain;
MinVolume = ALSource->MinGain;
MaxVolume = ALSource->MaxGain;
Pitch = ALSource->Pitch;
Resampler = ALSource->Resampler;
Position[0] = ALSource->Position[0];
Position[1] = ALSource->Position[1];
Position[2] = ALSource->Position[2];
Direction[0] = ALSource->Orientation[0];
Direction[1] = ALSource->Orientation[1];
Direction[2] = ALSource->Orientation[2];
Velocity[0] = ALSource->Velocity[0];
Velocity[1] = ALSource->Velocity[1];
Velocity[2] = ALSource->Velocity[2];
MinDist = ALSource->RefDistance;
MaxDist = ALSource->MaxDistance;
Rolloff = ALSource->RollOffFactor;
InnerAngle = ALSource->InnerAngle;
OuterAngle = ALSource->OuterAngle;
AirAbsorptionFactor = ALSource->AirAbsorptionFactor;
DryGainHFAuto = ALSource->DryGainHFAuto;
WetGainAuto = ALSource->WetGainAuto;
WetGainHFAuto = ALSource->WetGainHFAuto;
RoomRolloffBase = ALSource->RoomRolloffFactor;
ALSource->Params.Direct.OutBuffer = Device->DryBuffer;
ALSource->Params.Direct.ClickRemoval = Device->ClickRemoval;
ALSource->Params.Direct.PendingClicks = Device->PendingClicks;
for(i = 0;i < NumSends;i++)
{
ALeffectslot *Slot = ALSource->Send[i].Slot;
if(!Slot && i == 0)
Slot = Device->DefaultSlot;
if(!Slot || Slot->effect.type == AL_EFFECT_NULL)
{
Slot = NULL;
RoomRolloff[i] = 0.0f;
DecayDistance[i] = 0.0f;
RoomAirAbsorption[i] = 1.0f;
}
else if(Slot->AuxSendAuto)
{
RoomRolloff[i] = RoomRolloffBase;
if(IsReverbEffect(Slot->effect.type))
{
RoomRolloff[i] += Slot->effect.Reverb.RoomRolloffFactor;
DecayDistance[i] = Slot->effect.Reverb.DecayTime *
SPEEDOFSOUNDMETRESPERSEC;
RoomAirAbsorption[i] = Slot->effect.Reverb.AirAbsorptionGainHF;
}
else
{
DecayDistance[i] = 0.0f;
RoomAirAbsorption[i] = 1.0f;
}
}
else
{
/* If the slot's auxiliary send auto is off, the data sent to the
* effect slot is the same as the dry path, sans filter effects */
RoomRolloff[i] = Rolloff;
DecayDistance[i] = 0.0f;
RoomAirAbsorption[i] = AIRABSORBGAINHF;
}
ALSource->Params.Send[i].Slot = Slot;
}
/* Transform source to listener space (convert to head relative) */
if(ALSource->HeadRelative == AL_FALSE)
{
ALfloat (*RESTRICT Matrix)[4] = ALContext->Listener->Params.Matrix;
/* Transform source vectors */
aluMatrixVector(Position, 1.0f, Matrix);
aluMatrixVector(Direction, 0.0f, Matrix);
aluMatrixVector(Velocity, 0.0f, Matrix);
}
else
{
const ALfloat *ListenerVel = ALContext->Listener->Params.Velocity;
/* Offset the source velocity to be relative of the listener velocity */
Velocity[0] += ListenerVel[0];
Velocity[1] += ListenerVel[1];
Velocity[2] += ListenerVel[2];
}
SourceToListener[0] = -Position[0];
SourceToListener[1] = -Position[1];
SourceToListener[2] = -Position[2];
aluNormalize(SourceToListener);
aluNormalize(Direction);
/* Calculate distance attenuation */
Distance = sqrtf(aluDotproduct(Position, Position));
ClampedDist = Distance;
Attenuation = 1.0f;
for(i = 0;i < NumSends;i++)
RoomAttenuation[i] = 1.0f;
switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel :
ALContext->DistanceModel)
{
case InverseDistanceClamped:
ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
if(MaxDist < MinDist)
break;
/*fall-through*/
case InverseDistance:
if(MinDist > 0.0f)
{
if((MinDist + (Rolloff * (ClampedDist - MinDist))) > 0.0f)
Attenuation = MinDist / (MinDist + (Rolloff * (ClampedDist - MinDist)));
for(i = 0;i < NumSends;i++)
{
if((MinDist + (RoomRolloff[i] * (ClampedDist - MinDist))) > 0.0f)
RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (ClampedDist - MinDist)));
}
}
break;
case LinearDistanceClamped:
ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
if(MaxDist < MinDist)
break;
/*fall-through*/
case LinearDistance:
if(MaxDist != MinDist)
{
Attenuation = 1.0f - (Rolloff*(ClampedDist-MinDist)/(MaxDist - MinDist));
Attenuation = maxf(Attenuation, 0.0f);
for(i = 0;i < NumSends;i++)
{
RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(ClampedDist-MinDist)/(MaxDist - MinDist));
RoomAttenuation[i] = maxf(RoomAttenuation[i], 0.0f);
}
}
break;
case ExponentDistanceClamped:
ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
if(MaxDist < MinDist)
break;
/*fall-through*/
case ExponentDistance:
if(ClampedDist > 0.0f && MinDist > 0.0f)
{
Attenuation = powf(ClampedDist/MinDist, -Rolloff);
for(i = 0;i < NumSends;i++)
RoomAttenuation[i] = powf(ClampedDist/MinDist, -RoomRolloff[i]);
}
break;
case DisableDistance:
ClampedDist = MinDist;
break;
}
/* Source Gain + Attenuation */
DryGain = SourceVolume * Attenuation;
for(i = 0;i < NumSends;i++)
WetGain[i] = SourceVolume * RoomAttenuation[i];
/* Distance-based air absorption */
if(AirAbsorptionFactor > 0.0f && ClampedDist > MinDist)
{
ALfloat meters = maxf(ClampedDist-MinDist, 0.0f) * MetersPerUnit;
DryGainHF *= powf(AIRABSORBGAINHF, AirAbsorptionFactor*meters);
for(i = 0;i < NumSends;i++)
WetGainHF[i] *= powf(RoomAirAbsorption[i], AirAbsorptionFactor*meters);
}
if(WetGainAuto)
{
ALfloat ApparentDist = 1.0f/maxf(Attenuation, 0.00001f) - 1.0f;
/* Apply a decay-time transformation to the wet path, based on the
* attenuation of the dry path.
*
* Using the apparent distance, based on the distance attenuation, the
* initial decay of the reverb effect is calculated and applied to the
* wet path.
*/
for(i = 0;i < NumSends;i++)
{
if(DecayDistance[i] > 0.0f)
WetGain[i] *= powf(0.001f/*-60dB*/, ApparentDist/DecayDistance[i]);
}
}
/* Calculate directional soundcones */
Angle = acosf(aluDotproduct(Direction,SourceToListener)) * ConeScale * (360.0f/F_PI);
if(Angle > InnerAngle && Angle <= OuterAngle)
{
ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
ConeVolume = lerp(1.0f, ALSource->OuterGain, scale);
ConeHF = lerp(1.0f, ALSource->OuterGainHF, scale);
}
else if(Angle > OuterAngle)
{
ConeVolume = ALSource->OuterGain;
ConeHF = ALSource->OuterGainHF;
}
else
{
ConeVolume = 1.0f;
ConeHF = 1.0f;
}
DryGain *= ConeVolume;
if(WetGainAuto)
{
for(i = 0;i < NumSends;i++)
WetGain[i] *= ConeVolume;
}
if(DryGainHFAuto)
DryGainHF *= ConeHF;
if(WetGainHFAuto)
{
for(i = 0;i < NumSends;i++)
WetGainHF[i] *= ConeHF;
}
/* Clamp to Min/Max Gain */
DryGain = clampf(DryGain, MinVolume, MaxVolume);
for(i = 0;i < NumSends;i++)
WetGain[i] = clampf(WetGain[i], MinVolume, MaxVolume);
/* Apply gain and frequency filters */
DryGain *= ALSource->DirectGain * ListenerGain;
DryGainHF *= ALSource->DirectGainHF;
for(i = 0;i < NumSends;i++)
{
WetGain[i] *= ALSource->Send[i].Gain * ListenerGain;
WetGainHF[i] *= ALSource->Send[i].GainHF;
}
/* Calculate velocity-based doppler effect */
if(DopplerFactor > 0.0f)
{
const ALfloat *ListenerVel = ALContext->Listener->Params.Velocity;
ALfloat VSS, VLS;
if(SpeedOfSound < 1.0f)
{
DopplerFactor *= 1.0f/SpeedOfSound;
SpeedOfSound = 1.0f;
}
VSS = aluDotproduct(Velocity, SourceToListener) * DopplerFactor;
VLS = aluDotproduct(ListenerVel, SourceToListener) * DopplerFactor;
Pitch *= clampf(SpeedOfSound-VLS, 1.0f, SpeedOfSound*2.0f - 1.0f) /
clampf(SpeedOfSound-VSS, 1.0f, SpeedOfSound*2.0f - 1.0f);
}
BufferListItem = ALSource->queue;
while(BufferListItem != NULL)
{
ALbuffer *ALBuffer;
if((ALBuffer=BufferListItem->buffer) != NULL)
{
/* Calculate fixed-point stepping value, based on the pitch, buffer
* frequency, and output frequency. */
ALsizei maxstep = BUFFERSIZE;
maxstep -= ResamplerPadding[Resampler] +
ResamplerPrePadding[Resampler] + 1;
maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
Pitch = Pitch * ALBuffer->Frequency / Frequency;
if(Pitch > (ALfloat)maxstep)
ALSource->Params.Step = maxstep<<FRACTIONBITS;
else
{
ALSource->Params.Step = fastf2i(Pitch*FRACTIONONE);
if(ALSource->Params.Step == 0)
ALSource->Params.Step = 1;
}
ALSource->Params.Resample = SelectResampler(Resampler, ALSource->Params.Step);
break;
}
BufferListItem = BufferListItem->next;
}
if(Device->Hrtf)
ALSource->Params.DryMix = SelectHrtfMixer();
else
ALSource->Params.DryMix = SelectDirectMixer();
ALSource->Params.WetMix = SelectSendMixer();
if(Device->Hrtf)
{
/* Use a binaural HRTF algorithm for stereo headphone playback */
ALfloat delta, ev = 0.0f, az = 0.0f;
if(Distance > FLT_EPSILON)
{
ALfloat invlen = 1.0f/Distance;
Position[0] *= invlen;
Position[1] *= invlen;
Position[2] *= invlen;
/* Calculate elevation and azimuth only when the source is not at
* the listener. This prevents +0 and -0 Z from producing
* inconsistent panning. Also, clamp Y in case FP precision errors
* cause it to land outside of -1..+1. */
ev = asinf(clampf(Position[1], -1.0f, 1.0f));
az = atan2f(Position[0], -Position[2]*ZScale);
}
/* Check to see if the HRIR is already moving. */
if(ALSource->Hrtf.Moving)
{
/* Calculate the normalized HRTF transition factor (delta). */
delta = CalcHrtfDelta(ALSource->Params.Direct.Hrtf.Params.Gain, DryGain,
ALSource->Params.Direct.Hrtf.Params.Dir, Position);
/* If the delta is large enough, get the moving HRIR target
* coefficients, target delays, steppping values, and counter. */
if(delta > 0.001f)
{
ALSource->Hrtf.Counter = GetMovingHrtfCoeffs(Device->Hrtf,
ev, az, DryGain, delta,
ALSource->Hrtf.Counter,
ALSource->Params.Direct.Hrtf.Params.Coeffs[0],
ALSource->Params.Direct.Hrtf.Params.Delay[0],
ALSource->Params.Direct.Hrtf.Params.CoeffStep,
ALSource->Params.Direct.Hrtf.Params.DelayStep);
ALSource->Params.Direct.Hrtf.Params.Gain = DryGain;
ALSource->Params.Direct.Hrtf.Params.Dir[0] = Position[0];
ALSource->Params.Direct.Hrtf.Params.Dir[1] = Position[1];
ALSource->Params.Direct.Hrtf.Params.Dir[2] = Position[2];
}
}
else
{
/* Get the initial (static) HRIR coefficients and delays. */
GetLerpedHrtfCoeffs(Device->Hrtf, ev, az, DryGain,
ALSource->Params.Direct.Hrtf.Params.Coeffs[0],
ALSource->Params.Direct.Hrtf.Params.Delay[0]);
ALSource->Hrtf.Counter = 0;
ALSource->Hrtf.Moving = AL_TRUE;
ALSource->Params.Direct.Hrtf.Params.Gain = DryGain;
ALSource->Params.Direct.Hrtf.Params.Dir[0] = Position[0];
ALSource->Params.Direct.Hrtf.Params.Dir[1] = Position[1];
ALSource->Params.Direct.Hrtf.Params.Dir[2] = Position[2];
}
ALSource->Params.Direct.Hrtf.Params.IrSize = GetHrtfIrSize(Device->Hrtf);
ALSource->Params.Direct.Hrtf.State = &ALSource->Hrtf;
}
else
{
ALfloat (*Matrix)[MaxChannels] = ALSource->Params.Direct.Gains;
ALfloat DirGain = 0.0f;
ALfloat AmbientGain;
for(i = 0;i < MaxChannels;i++)
{
for(j = 0;j < MaxChannels;j++)
Matrix[i][j] = 0.0f;
}
/* Normalize the length, and compute panned gains. */
if(Distance > FLT_EPSILON)
{
ALfloat invlen = 1.0f/Distance;
Position[0] *= invlen;
Position[1] *= invlen;
Position[2] *= invlen;
DirGain = sqrtf(Position[0]*Position[0] + Position[2]*Position[2]);
ComputeAngleGains(Device, atan2f(Position[0], -Position[2]*ZScale), 0.0f,
DryGain*DirGain, Matrix[0]);
}
/* Adjustment for vertical offsets. Not the greatest, but simple
* enough. */
AmbientGain = DryGain * sqrtf(1.0f/Device->NumChan) * (1.0f-DirGain);
for(i = 0;i < (ALint)Device->NumChan;i++)
{
enum Channel chan = Device->Speaker2Chan[i];
Matrix[0][chan] = maxf(Matrix[0][chan], AmbientGain);
}
}
for(i = 0;i < NumSends;i++)
ALSource->Params.Send[i].Gain = WetGain[i];
/* Update filter coefficients. */
cw = cosf(F_PI*2.0f * LOWPASSFREQREF / Frequency);
ALSource->Params.Direct.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
for(i = 0;i < NumSends;i++)
{
ALfloat a = lpCoeffCalc(WetGainHF[i], cw);
ALSource->Params.Send[i].iirFilter.coeff = a;
}
}
static __inline ALfloat aluF2F(ALfloat val)
{ return val; }
static __inline ALint aluF2I(ALfloat val)
{
/* Clamp the value between -1 and +1. This handles that without branching. */
val = val+1.0f - fabsf(val-1.0f);
val = (val-2.0f + fabsf(val+2.0f)) * 0.25f;
/* Convert to a signed integer, between -2147483647 and +2147483647. */
return fastf2i((ALfloat)(val*2147483647.0));
}
static __inline ALuint aluF2UI(ALfloat val)
{ return aluF2I(val)+2147483648u; }
static __inline ALshort aluF2S(ALfloat val)
{ return aluF2I(val)>>16; }
static __inline ALushort aluF2US(ALfloat val)
{ return aluF2S(val)+32768; }
static __inline ALbyte aluF2B(ALfloat val)
{ return aluF2I(val)>>24; }
static __inline ALubyte aluF2UB(ALfloat val)
{ return aluF2B(val)+128; }
#define DECL_TEMPLATE(T, func) \
static int Write_##T(ALCdevice *device, T *RESTRICT buffer, \
ALuint SamplesToDo) \
{ \
ALfloat (*RESTRICT DryBuffer)[BUFFERSIZE] = device->DryBuffer; \
ALuint numchans = ChannelsFromDevFmt(device->FmtChans); \
const ALuint *offsets = device->ChannelOffsets; \
ALuint i, j; \
\
for(j = 0;j < MaxChannels;j++) \
{ \
T *RESTRICT out; \
\
if(offsets[j] == INVALID_OFFSET) \
continue; \
\
out = buffer + offsets[j]; \
for(i = 0;i < SamplesToDo;i++) \
out[i*numchans] = func(DryBuffer[j][i]); \
} \
return SamplesToDo*numchans*sizeof(T); \
}
DECL_TEMPLATE(ALfloat, aluF2F)
DECL_TEMPLATE(ALuint, aluF2UI)
DECL_TEMPLATE(ALint, aluF2I)
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 **slot, **slot_end;
ALsource **src, **src_end;
ALCcontext *ctx;
FPUCtl oldMode;
ALuint i, c;
SetMixerFPUMode(&oldMode);
while(size > 0)
{
SamplesToDo = minu(size, BUFFERSIZE);
for(c = 0;c < MaxChannels;c++)
memset(device->DryBuffer[c], 0, SamplesToDo*sizeof(ALfloat));
ALCdevice_Lock(device);
ctx = device->ContextList;
while(ctx)
{
ALenum DeferUpdates = ctx->DeferUpdates;
ALenum UpdateSources = AL_FALSE;
if(!DeferUpdates)
UpdateSources = ExchangeInt(&ctx->UpdateSources, AL_FALSE);
if(UpdateSources)
CalcListenerParams(ctx->Listener);
/* source processing */
src = ctx->ActiveSources;
src_end = src + ctx->ActiveSourceCount;
while(src != src_end)
{
if((*src)->state != AL_PLAYING)
{
--(ctx->ActiveSourceCount);
*src = *(--src_end);
continue;
}
if(!DeferUpdates && (ExchangeInt(&(*src)->NeedsUpdate, AL_FALSE) ||
UpdateSources))
ALsource_Update(*src, ctx);
MixSource(*src, device, SamplesToDo);
src++;
}
/* effect slot processing */
slot = ctx->ActiveEffectSlots;
slot_end = slot + ctx->ActiveEffectSlotCount;
while(slot != slot_end)
{
ALfloat offset = (*slot)->ClickRemoval[0];
if(offset < (1.0f/32768.0f))
offset = 0.0f;
else for(i = 0;i < SamplesToDo;i++)
{
(*slot)->WetBuffer[0][i] += offset;
offset -= offset * (1.0f/256.0f);
}
(*slot)->ClickRemoval[0] = offset + (*slot)->PendingClicks[0];
(*slot)->PendingClicks[0] = 0.0f;
if(!DeferUpdates && ExchangeInt(&(*slot)->NeedsUpdate, AL_FALSE))
ALeffectState_Update((*slot)->EffectState, device, *slot);
ALeffectState_Process((*slot)->EffectState, SamplesToDo,
(*slot)->WetBuffer[0], device->DryBuffer);
for(i = 0;i < SamplesToDo;i++)
(*slot)->WetBuffer[0][i] = 0.0f;
slot++;
}
ctx = ctx->next;
}
slot = &device->DefaultSlot;
if(*slot != NULL)
{
ALfloat offset = (*slot)->ClickRemoval[0];
if(offset < (1.0f/32768.0f))
offset = 0.0f;
else for(i = 0;i < SamplesToDo;i++)
{
(*slot)->WetBuffer[0][i] += offset;
offset -= offset * (1.0f/256.0f);
}
(*slot)->ClickRemoval[0] = offset + (*slot)->PendingClicks[0];
(*slot)->PendingClicks[0] = 0.0f;
if(ExchangeInt(&(*slot)->NeedsUpdate, AL_FALSE))
ALeffectState_Update((*slot)->EffectState, device, *slot);
ALeffectState_Process((*slot)->EffectState, SamplesToDo,
(*slot)->WetBuffer[0], device->DryBuffer);
for(i = 0;i < SamplesToDo;i++)
(*slot)->WetBuffer[0][i] = 0.0f;
}
ALCdevice_Unlock(device);
/* Click-removal. Could do better; this only really handles immediate
* changes between updates where a predictive sample could be
* generated. Delays caused by effects and HRTF aren't caught. */
if(device->FmtChans == DevFmtMono)
{
ALfloat offset = device->ClickRemoval[FrontCenter];
if(offset < (1.0f/32768.0f))
offset = 0.0f;
else for(i = 0;i < SamplesToDo;i++)
{
device->DryBuffer[FrontCenter][i] += offset;
offset -= offset * (1.0f/256.0f);
}
device->ClickRemoval[FrontCenter] = offset + device->PendingClicks[FrontCenter];
device->PendingClicks[FrontCenter] = 0.0f;
}
else if(device->FmtChans == DevFmtStereo)
{
/* Assumes the first two channels are FrontLeft and FrontRight */
for(c = 0;c < 2;c++)
{
ALfloat offset = device->ClickRemoval[c];
if(offset < (1.0f/32768.0f))
offset = 0.0f;
else for(i = 0;i < SamplesToDo;i++)
{
device->DryBuffer[c][i] += offset;
offset -= offset * (1.0f/256.0f);
}
device->ClickRemoval[c] = offset + device->PendingClicks[c];
device->PendingClicks[c] = 0.0f;
}
if(device->Bs2b)
{
float samples[2];
for(i = 0;i < SamplesToDo;i++)
{
samples[0] = device->DryBuffer[FrontLeft][i];
samples[1] = device->DryBuffer[FrontRight][i];
bs2b_cross_feed(device->Bs2b, samples);
device->DryBuffer[FrontLeft][i] = samples[0];
device->DryBuffer[FrontRight][i] = samples[1];
}
}
}
else
{
for(c = 0;c < MaxChannels;c++)
{
ALfloat offset = device->ClickRemoval[c];
if(offset < (1.0f/32768.0f))
offset = 0.0f;
else for(i = 0;i < SamplesToDo;i++)
{
device->DryBuffer[c][i] += offset;
offset -= offset * (1.0f/256.0f);
}
device->ClickRemoval[c] = offset + device->PendingClicks[c];
device->PendingClicks[c] = 0.0f;
}
}
if(buffer)
{
int bytes = 0;
switch(device->FmtType)
{
case DevFmtByte:
bytes = Write_ALbyte(device, buffer, SamplesToDo);
break;
case DevFmtUByte:
bytes = Write_ALubyte(device, buffer, SamplesToDo);
break;
case DevFmtShort:
bytes = Write_ALshort(device, buffer, SamplesToDo);
break;
case DevFmtUShort:
bytes = Write_ALushort(device, buffer, SamplesToDo);
break;
case DevFmtInt:
bytes = Write_ALint(device, buffer, SamplesToDo);
break;
case DevFmtUInt:
bytes = Write_ALuint(device, buffer, SamplesToDo);
break;
case DevFmtFloat:
bytes = Write_ALfloat(device, buffer, SamplesToDo);
break;
}
buffer = (ALubyte*)buffer + bytes;
}
size -= SamplesToDo;
}
RestoreFPUMode(&oldMode);
}
ALvoid aluHandleDisconnect(ALCdevice *device)
{
ALCcontext *Context;
device->Connected = ALC_FALSE;
Context = device->ContextList;
while(Context)
{
ALsource **src, **src_end;
src = Context->ActiveSources;
src_end = src + Context->ActiveSourceCount;
while(src != src_end)
{
if((*src)->state == AL_PLAYING)
{
(*src)->state = AL_STOPPED;
(*src)->BuffersPlayed = (*src)->BuffersInQueue;
(*src)->position = 0;
(*src)->position_fraction = 0;
}
src++;
}
Context->ActiveSourceCount = 0;
Context = Context->next;
}
}
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