static ALvoid EchoUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffect *Effect)
{
ALechoState *state = (ALechoState*)effect;
ALuint frequency = Context->Device->Frequency;
ALfp lrpan, cw, a, g;
state->Tap[0].delay = (ALuint)ALfp2int((ALfpMult(Effect->Echo.Delay, int2ALfp(frequency)) + int2ALfp(1)));
state->Tap[1].delay = (ALuint)ALfp2int(ALfpMult(Effect->Echo.LRDelay, int2ALfp(frequency)));
state->Tap[1].delay += state->Tap[0].delay;
lrpan = (ALfpMult(Effect->Echo.Spread, float2ALfp(0.5f)) + float2ALfp(0.5f));
state->GainL = aluSqrt( lrpan);
state->GainR = aluSqrt((int2ALfp(1)-lrpan));
state->FeedGain = Effect->Echo.Feedback;
cw = __cos(ALfpDiv(float2ALfp(2.0*M_PI * LOWPASSFREQCUTOFF), int2ALfp(frequency)));
g = (int2ALfp(1) - Effect->Echo.Damping);
a = int2ALfp(0);
if(g < float2ALfp(0.9999f)) { /* 1-epsilon */
// a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g);
a = ALfpDiv((int2ALfp(1) - ALfpMult(g,cw) - aluSqrt((ALfpMult(ALfpMult(int2ALfp(2),g),(int2ALfp(1)-cw)) -
ALfpMult(ALfpMult(g,g),(int2ALfp(1) - ALfpMult(cw,cw)))))),
(int2ALfp(1) - g));
}
state->iirFilter.coeff = a;
}
static ALvoid ModulatorUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
{
ALmodulatorState *state = (ALmodulatorState*)effect;
ALfloat gain, cw, a = 0.0f;
ALuint index;
if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
state->Waveform = SINUSOID;
else if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH)
state->Waveform = SAWTOOTH;
else if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)
state->Waveform = SQUARE;
state->step = fastf2u(Slot->effect.Modulator.Frequency*WAVEFORM_FRACONE /
Device->Frequency);
if(state->step == 0) state->step = 1;
cw = aluCos(F_PI*2.0f * Slot->effect.Modulator.HighPassCutoff /
Device->Frequency);
a = (2.0f-cw) - aluSqrt(aluPow(2.0f-cw, 2.0f) - 1.0f);
state->iirFilter.coeff = a;
gain = aluSqrt(1.0f/Device->NumChan);
gain *= Slot->Gain;
for(index = 0;index < MAXCHANNELS;index++)
state->Gain[index] = 0.0f;
for(index = 0;index < Device->NumChan;index++)
{
enum Channel chan = Device->Speaker2Chan[index];
state->Gain[chan] = gain;
}
}
static ALvoid EchoUpdate( ALeffectState* effect, ALCcontext* Context, const ALeffect* Effect )
{
ALechoState* state = ( ALechoState* )effect;
ALuint frequency = Context->Device->Frequency;
ALfloat lrpan, cw, a, g;
state->Tap[0].delay = ( ALuint )( Effect->Echo.Delay * frequency ) + 1;
state->Tap[1].delay = ( ALuint )( Effect->Echo.LRDelay * frequency );
state->Tap[1].delay += state->Tap[0].delay;
lrpan = Effect->Echo.Spread * 0.5f + 0.5f;
state->GainL = aluSqrt( lrpan );
state->GainR = aluSqrt( 1.0f - lrpan );
state->FeedGain = Effect->Echo.Feedback;
cw = cos( 2.0 * M_PI * LOWPASSFREQCUTOFF / frequency );
g = 1.0f - Effect->Echo.Damping;
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 );
}
state->iirFilter.coeff = a;
}
static ALvoid EchoUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffectslot *Slot)
{
ALechoState *state = (ALechoState*)effect;
ALCdevice *Device = Context->Device;
ALuint frequency = Device->Frequency;
ALfloat lrpan, cw, g, gain;
ALuint i;
state->Tap[0].delay = (ALuint)(Slot->effect.Echo.Delay * frequency) + 1;
state->Tap[1].delay = (ALuint)(Slot->effect.Echo.LRDelay * frequency);
state->Tap[1].delay += state->Tap[0].delay;
lrpan = Slot->effect.Echo.Spread*0.5f + 0.5f;
state->GainL = aluSqrt( lrpan);
state->GainR = aluSqrt(1.0f-lrpan);
state->FeedGain = Slot->effect.Echo.Feedback;
cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / frequency);
g = 1.0f - Slot->effect.Echo.Damping;
state->iirFilter.coeff = lpCoeffCalc(g, cw);
gain = Slot->Gain;
for(i = 0;i < MAXCHANNELS;i++)
state->Gain[i] = 0.0f;
for(i = 0;i < Device->NumChan;i++)
{
enum Channel chan = Device->Speaker2Chan[i];
state->Gain[chan] = gain;
}
}
static ALboolean EchoDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
{
ALechoState *state = (ALechoState*)effect;
ALuint maxlen, i;
// Use the next power of 2 for the buffer length, so the tap offsets can be
// wrapped using a mask instead of a modulo
maxlen = (ALuint)(AL_ECHO_MAX_DELAY * Device->Frequency) + 1;
maxlen += (ALuint)(AL_ECHO_MAX_LRDELAY * Device->Frequency) + 1;
maxlen = NextPowerOf2(maxlen);
if(maxlen != state->BufferLength)
{
void *temp;
temp = realloc(state->SampleBuffer, maxlen * sizeof(ALfloat));
if(!temp)
return AL_FALSE;
state->SampleBuffer = temp;
state->BufferLength = maxlen;
}
for(i = 0;i < state->BufferLength;i++)
state->SampleBuffer[i] = 0.0f;
state->Scale = aluSqrt(Device->NumChan / 6.0f);
state->Scale = __min(state->Scale, 1.0f);
return AL_TRUE;
}
// This updates the device-dependant reverb state. This is called on
// initialization and any time the device parameters (eg. playback frequency,
// or format) have been changed.
static ALboolean VerbDeviceUpdate( ALeffectState* effect, ALCdevice* Device )
{
ALverbState* State = ( ALverbState* )effect;
ALuint frequency = Device->Frequency, index;
// Allocate the delay lines.
if ( !AllocLines( AL_FALSE, frequency, State ) )
{
return AL_FALSE;
}
State->Scale = aluSqrt( Device->NumChan / 8.0f );
// The early reflection and late all-pass filter line lengths are static,
// so their offsets only need to be calculated once.
for ( index = 0; index < 4; index++ )
{
State->Early.Offset[index] = ( ALuint )( EARLY_LINE_LENGTH[index] *
frequency );
State->Late.ApOffset[index] = ( ALuint )( ALLPASS_LINE_LENGTH[index] *
frequency );
}
return AL_TRUE;
}
static ALvoid ModulatorUpdate( ALeffectState* effect, ALCcontext* Context, const ALeffect* Effect )
{
ALmodulatorState* state = ( ALmodulatorState* )effect;
ALfloat cw, a = 0.0f;
if ( Effect->Modulator.Waveform == AL_RING_MODULATOR_SINUSOID )
{
state->Waveform = SINUSOID;
}
else if ( Effect->Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH )
{
state->Waveform = SAWTOOTH;
}
else if ( Effect->Modulator.Waveform == AL_RING_MODULATOR_SQUARE )
{
state->Waveform = SQUARE;
}
state->step = Effect->Modulator.Frequency * ( 1 << WAVEFORM_FRACBITS ) /
Context->Device->Frequency;
if ( !state->step )
{
state->step = 1;
}
cw = cos( 2.0 * M_PI * Effect->Modulator.HighPassCutoff / Context->Device->Frequency );
a = ( 2.0f - cw ) - aluSqrt( aluPow( 2.0f - cw, 2.0f ) - 1.0f );
state->iirFilter.coeff = a;
}
static ALboolean ModulatorDeviceUpdate( ALeffectState* effect, ALCdevice* Device )
{
ALmodulatorState* state = ( ALmodulatorState* )effect;
state->Scale = aluSqrt( Device->NumChan / 8.0f );
return AL_TRUE;
}
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;
}
}
// Calculate the mixing matrix coefficients given a diffusion factor.
static __inline ALvoid CalcMatrixCoeffs(ALfloat diffusion, ALfloat *x, ALfloat *y)
{
ALfloat n, t;
// The matrix is of order 4, so n is sqrt (4 - 1).
n = aluSqrt(3.0f);
t = diffusion * atan(n);
// Calculate the first mixing matrix coefficient.
*x = cos(t);
// Calculate the second mixing matrix coefficient.
*y = sin(t) / n;
}
ALfloat lpCoeffCalc(ALfloat g, ALfloat cw)
{
ALfloat a = 0.0f;
/* Be careful with gains < 0.01, as that causes the coefficient
* head towards 1, which will flatten the signal */
if(g < 0.9999f) /* 1-epsilon */
{
g = maxf(g, 0.01f);
a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) /
(1 - g);
}
return a;
}
// Calculate an attenuation to be applied to the input of any echo models to
// compensate for modal density and decay time.
static __inline ALfloat CalcDensityGain(ALfloat a)
{
/* The energy of a signal can be obtained by finding the area under the
* squared signal. This takes the form of Sum(x_n^2), where x is the
* amplitude for the sample n.
*
* Decaying feedback matches exponential decay of the form Sum(a^n),
* where a is the attenuation coefficient, and n is the sample. The area
* under this decay curve can be calculated as: 1 / (1 - a).
*
* Modifying the above equation to find the squared area under the curve
* (for energy) yields: 1 / (1 - a^2). Input attenuation can then be
* calculated by inverting the square root of this approximation,
* yielding: 1 / sqrt(1 / (1 - a^2)), simplified to: sqrt(1 - a^2).
*/
return aluSqrt(1.0f - (a * a));
}
// This updates the device-dependant EAX reverb state. This is called on
// initialization and any time the device parameters (eg. playback frequency,
// format) have been changed.
static ALboolean EAXVerbDeviceUpdate( ALeffectState* effect, ALCdevice* Device )
{
ALverbState* State = ( ALverbState* )effect;
ALuint frequency = Device->Frequency, index;
// Allocate the delay lines.
if ( !AllocLines( AL_TRUE, frequency, State ) )
{
return AL_FALSE;
}
State->Scale = aluSqrt( Device->NumChan / 8.0f );
// Calculate the modulation filter coefficient. Notice that the exponent
// is calculated given the current sample rate. This ensures that the
// resulting filter response over time is consistent across all sample
// rates.
State->Mod.Coeff = aluPow( MODULATION_FILTER_COEFF,
MODULATION_FILTER_CONST / frequency );
// The early reflection and late all-pass filter line lengths are static,
// so their offsets only need to be calculated once.
for ( index = 0; index < 4; index++ )
{
State->Early.Offset[index] = ( ALuint )( EARLY_LINE_LENGTH[index] *
frequency );
State->Late.ApOffset[index] = ( ALuint )( ALLPASS_LINE_LENGTH[index] *
frequency );
}
// The echo all-pass filter line length is static, so its offset only
// needs to be calculated once.
State->Echo.ApOffset = ( ALuint )( ECHO_ALLPASS_LENGTH * frequency );
return AL_TRUE;
}
ALvoid aluInitPanning(ALCdevice *Device)
{
ALfloat SpeakerAngle[MAXCHANNELS];
enum Channel *Speaker2Chan;
ALfloat Alpha, Theta;
ALint pos;
ALuint s;
Speaker2Chan = Device->Speaker2Chan;
switch(Device->FmtChans)
{
case DevFmtMono:
Device->NumChan = 1;
Speaker2Chan[0] = FRONT_CENTER;
SpeakerAngle[0] = F_PI/180.0f * 0.0f;
break;
case DevFmtStereo:
Device->NumChan = 2;
Speaker2Chan[0] = FRONT_LEFT;
Speaker2Chan[1] = FRONT_RIGHT;
SpeakerAngle[0] = F_PI/180.0f * -90.0f;
SpeakerAngle[1] = F_PI/180.0f * 90.0f;
SetSpeakerArrangement("layout_STEREO", SpeakerAngle, Speaker2Chan, Device->NumChan);
break;
case DevFmtQuad:
Device->NumChan = 4;
Speaker2Chan[0] = BACK_LEFT;
Speaker2Chan[1] = FRONT_LEFT;
Speaker2Chan[2] = FRONT_RIGHT;
Speaker2Chan[3] = BACK_RIGHT;
SpeakerAngle[0] = F_PI/180.0f * -135.0f;
SpeakerAngle[1] = F_PI/180.0f * -45.0f;
SpeakerAngle[2] = F_PI/180.0f * 45.0f;
SpeakerAngle[3] = F_PI/180.0f * 135.0f;
SetSpeakerArrangement("layout_QUAD", SpeakerAngle, Speaker2Chan, Device->NumChan);
break;
case DevFmtX51:
Device->NumChan = 5;
Speaker2Chan[0] = BACK_LEFT;
Speaker2Chan[1] = FRONT_LEFT;
Speaker2Chan[2] = FRONT_CENTER;
Speaker2Chan[3] = FRONT_RIGHT;
Speaker2Chan[4] = BACK_RIGHT;
SpeakerAngle[0] = F_PI/180.0f * -110.0f;
SpeakerAngle[1] = F_PI/180.0f * -30.0f;
SpeakerAngle[2] = F_PI/180.0f * 0.0f;
SpeakerAngle[3] = F_PI/180.0f * 30.0f;
SpeakerAngle[4] = F_PI/180.0f * 110.0f;
SetSpeakerArrangement("layout_51CHN", SpeakerAngle, Speaker2Chan, Device->NumChan);
break;
case DevFmtX51Side:
Device->NumChan = 5;
Speaker2Chan[0] = SIDE_LEFT;
Speaker2Chan[1] = FRONT_LEFT;
Speaker2Chan[2] = FRONT_CENTER;
Speaker2Chan[3] = FRONT_RIGHT;
Speaker2Chan[4] = SIDE_RIGHT;
SpeakerAngle[0] = F_PI/180.0f * -90.0f;
SpeakerAngle[1] = F_PI/180.0f * -30.0f;
SpeakerAngle[2] = F_PI/180.0f * 0.0f;
SpeakerAngle[3] = F_PI/180.0f * 30.0f;
SpeakerAngle[4] = F_PI/180.0f * 90.0f;
SetSpeakerArrangement("layout_51SIDECHN", SpeakerAngle, Speaker2Chan, Device->NumChan);
break;
case DevFmtX61:
Device->NumChan = 6;
Speaker2Chan[0] = SIDE_LEFT;
Speaker2Chan[1] = FRONT_LEFT;
Speaker2Chan[2] = FRONT_CENTER;
Speaker2Chan[3] = FRONT_RIGHT;
Speaker2Chan[4] = SIDE_RIGHT;
Speaker2Chan[5] = BACK_CENTER;
SpeakerAngle[0] = F_PI/180.0f * -90.0f;
SpeakerAngle[1] = F_PI/180.0f * -30.0f;
SpeakerAngle[2] = F_PI/180.0f * 0.0f;
SpeakerAngle[3] = F_PI/180.0f * 30.0f;
SpeakerAngle[4] = F_PI/180.0f * 90.0f;
SpeakerAngle[5] = F_PI/180.0f * 180.0f;
SetSpeakerArrangement("layout_61CHN", SpeakerAngle, Speaker2Chan, Device->NumChan);
break;
case DevFmtX71:
Device->NumChan = 7;
Speaker2Chan[0] = BACK_LEFT;
Speaker2Chan[1] = SIDE_LEFT;
Speaker2Chan[2] = FRONT_LEFT;
Speaker2Chan[3] = FRONT_CENTER;
Speaker2Chan[4] = FRONT_RIGHT;
Speaker2Chan[5] = SIDE_RIGHT;
Speaker2Chan[6] = BACK_RIGHT;
SpeakerAngle[0] = F_PI/180.0f * -150.0f;
SpeakerAngle[1] = F_PI/180.0f * -90.0f;
SpeakerAngle[2] = F_PI/180.0f * -30.0f;
SpeakerAngle[3] = F_PI/180.0f * 0.0f;
SpeakerAngle[4] = F_PI/180.0f * 30.0f;
SpeakerAngle[5] = F_PI/180.0f * 90.0f;
SpeakerAngle[6] = F_PI/180.0f * 150.0f;
SetSpeakerArrangement("layout_71CHN", SpeakerAngle, Speaker2Chan, Device->NumChan);
break;
}
for(pos = 0; pos < LUT_NUM; pos++)
{
ALfloat *PanningLUT = Device->PanningLUT[pos];
/* clear all values */
for(s = 0; s < MAXCHANNELS; s++)
PanningLUT[s] = 0.0f;
if(Device->NumChan == 1)
{
PanningLUT[Speaker2Chan[0]] = 1.0f;
continue;
}
/* source angle */
Theta = aluLUTpos2Angle(pos);
/* set panning values */
for(s = 0; s < Device->NumChan - 1; s++)
{
if(Theta >= SpeakerAngle[s] && Theta < SpeakerAngle[s+1])
{
/* source between speaker s and speaker s+1 */
Alpha = (Theta-SpeakerAngle[s]) /
(SpeakerAngle[s+1]-SpeakerAngle[s]);
PanningLUT[Speaker2Chan[s]] = aluSqrt(1.0f-Alpha);
PanningLUT[Speaker2Chan[s+1]] = aluSqrt( Alpha);
break;
}
}
if(s == Device->NumChan - 1)
{
/* source between last and first speaker */
if(Theta < SpeakerAngle[0])
Theta += F_PI*2.0f;
Alpha = (Theta-SpeakerAngle[s]) /
(F_PI*2.0f + SpeakerAngle[0]-SpeakerAngle[s]);
PanningLUT[Speaker2Chan[s]] = aluSqrt(1.0f-Alpha);
PanningLUT[Speaker2Chan[0]] = aluSqrt( Alpha);
}
}
}
// Update the early and late 3D panning gains.
static ALvoid Update3DPanning( const ALfloat* ReflectionsPan, const ALfloat* LateReverbPan, ALfloat* PanningLUT, ALverbState* State )
{
ALfloat length;
ALfloat earlyPan[3] = { ReflectionsPan[0], ReflectionsPan[1],
ReflectionsPan[2]
};
ALfloat latePan[3] = { LateReverbPan[0], LateReverbPan[1],
LateReverbPan[2]
};
ALint pos;
ALfloat* speakerGain, dirGain, ambientGain;
ALuint index;
// Calculate the 3D-panning gains for the early reflections and late
// reverb.
length = earlyPan[0] * earlyPan[0] + earlyPan[1] * earlyPan[1] + earlyPan[2] * earlyPan[2];
if ( length > 1.0f )
{
length = 1.0f / aluSqrt( length );
earlyPan[0] *= length;
earlyPan[1] *= length;
earlyPan[2] *= length;
}
length = latePan[0] * latePan[0] + latePan[1] * latePan[1] + latePan[2] * latePan[2];
if ( length > 1.0f )
{
length = 1.0f / aluSqrt( length );
latePan[0] *= length;
latePan[1] *= length;
latePan[2] *= length;
}
/* This code applies directional reverb just like the mixer applies
* directional sources. It diffuses the sound toward all speakers as the
* magnitude of the panning vector drops, which is only a rough
* approximation of the expansion of sound across the speakers from the
* panning direction.
*/
pos = aluCart2LUTpos( earlyPan[2], earlyPan[0] );
speakerGain = &PanningLUT[OUTPUTCHANNELS * pos];
dirGain = aluSqrt( ( earlyPan[0] * earlyPan[0] ) + ( earlyPan[2] * earlyPan[2] ) );
ambientGain = ( 1.0 - dirGain );
for ( index = 0; index < OUTPUTCHANNELS; index++ )
{
State->Early.PanGain[index] = dirGain * speakerGain[index] + ambientGain;
}
pos = aluCart2LUTpos( latePan[2], latePan[0] );
speakerGain = &PanningLUT[OUTPUTCHANNELS * pos];
dirGain = aluSqrt( ( latePan[0] * latePan[0] ) + ( latePan[2] * latePan[2] ) );
ambientGain = ( 1.0 - dirGain );
for ( index = 0; index < OUTPUTCHANNELS; index++ )
{
State->Late.PanGain[index] = dirGain * speakerGain[index] + ambientGain;
}
}
// Update the early and late 3D panning gains.
static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALfloat Gain, ALverbState *State)
{
ALfloat earlyPan[3] = { ReflectionsPan[0], ReflectionsPan[1],
ReflectionsPan[2] };
ALfloat latePan[3] = { LateReverbPan[0], LateReverbPan[1],
LateReverbPan[2] };
const ALfloat *speakerGain;
ALfloat ambientGain;
ALfloat dirGain;
ALfloat length;
ALuint index;
ALint pos;
Gain *= ReverbBoost;
// Attenuate non-directional reverb according to the number of channels
ambientGain = aluSqrt(2.0f/Device->NumChan);
// Calculate the 3D-panning gains for the early reflections and late
// reverb.
length = earlyPan[0]*earlyPan[0] + earlyPan[1]*earlyPan[1] + earlyPan[2]*earlyPan[2];
if(length > 1.0f)
{
length = 1.0f / aluSqrt(length);
earlyPan[0] *= length;
earlyPan[1] *= length;
earlyPan[2] *= length;
}
length = latePan[0]*latePan[0] + latePan[1]*latePan[1] + latePan[2]*latePan[2];
if(length > 1.0f)
{
length = 1.0f / aluSqrt(length);
latePan[0] *= length;
latePan[1] *= length;
latePan[2] *= length;
}
/* This code applies directional reverb just like the mixer applies
* directional sources. It diffuses the sound toward all speakers as the
* magnitude of the panning vector drops, which is only a rough
* approximation of the expansion of sound across the speakers from the
* panning direction.
*/
pos = aluCart2LUTpos(earlyPan[2], earlyPan[0]);
speakerGain = Device->PanningLUT[pos];
dirGain = aluSqrt((earlyPan[0] * earlyPan[0]) + (earlyPan[2] * earlyPan[2]));
for(index = 0;index < MAXCHANNELS;index++)
State->Early.PanGain[index] = 0.0f;
for(index = 0;index < Device->NumChan;index++)
{
enum Channel chan = Device->Speaker2Chan[index];
State->Early.PanGain[chan] = lerp(ambientGain, speakerGain[chan], dirGain) * Gain;
}
pos = aluCart2LUTpos(latePan[2], latePan[0]);
speakerGain = Device->PanningLUT[pos];
dirGain = aluSqrt((latePan[0] * latePan[0]) + (latePan[2] * latePan[2]));
for(index = 0;index < MAXCHANNELS;index++)
State->Late.PanGain[index] = 0.0f;
for(index = 0;index < Device->NumChan;index++)
{
enum Channel chan = Device->Speaker2Chan[index];
State->Late.PanGain[chan] = lerp(ambientGain, speakerGain[chan], dirGain) * Gain;
}
}
// This updates the EAX reverb state. This is called any time the EAX reverb
// effect is loaded into a slot.
static ALvoid ReverbUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffectslot *Slot)
{
ALverbState *State = (ALverbState*)effect;
ALuint frequency = Context->Device->Frequency;
ALboolean isEAX = AL_FALSE;
ALfloat cw, x, y, hfRatio;
if(Slot->effect.type == AL_EFFECT_EAXREVERB && !EmulateEAXReverb)
{
State->state.Process = EAXVerbProcess;
isEAX = AL_TRUE;
}
else if(Slot->effect.type == AL_EFFECT_REVERB || EmulateEAXReverb)
{
State->state.Process = VerbProcess;
isEAX = AL_FALSE;
}
// Calculate the master low-pass filter (from the master effect HF gain).
cw = CalcI3DL2HFreq(Slot->effect.Params.Reverb.HFReference, frequency);
// This is done with 2 chained 1-pole filters, so no need to square g.
State->LpFilter.coeff = lpCoeffCalc(Slot->effect.Params.Reverb.GainHF, cw);
if(isEAX)
{
// Update the modulator line.
UpdateModulator(Slot->effect.Params.Reverb.ModulationTime,
Slot->effect.Params.Reverb.ModulationDepth,
frequency, State);
}
// Update the initial effect delay.
UpdateDelayLine(Slot->effect.Params.Reverb.ReflectionsDelay,
Slot->effect.Params.Reverb.LateReverbDelay,
frequency, State);
// Update the early lines.
UpdateEarlyLines(Slot->effect.Params.Reverb.Gain,
Slot->effect.Params.Reverb.ReflectionsGain,
Slot->effect.Params.Reverb.LateReverbDelay, State);
// Update the decorrelator.
UpdateDecorrelator(Slot->effect.Params.Reverb.Density, frequency, State);
// Get the mixing matrix coefficients (x and y).
CalcMatrixCoeffs(Slot->effect.Params.Reverb.Diffusion, &x, &y);
// Then divide x into y to simplify the matrix calculation.
State->Late.MixCoeff = y / x;
// If the HF limit parameter is flagged, calculate an appropriate limit
// based on the air absorption parameter.
hfRatio = Slot->effect.Params.Reverb.DecayHFRatio;
if(Slot->effect.Params.Reverb.DecayHFLimit &&
Slot->effect.Params.Reverb.AirAbsorptionGainHF < 1.0f)
hfRatio = CalcLimitedHfRatio(hfRatio,
Slot->effect.Params.Reverb.AirAbsorptionGainHF,
Slot->effect.Params.Reverb.DecayTime);
// Update the late lines.
UpdateLateLines(Slot->effect.Params.Reverb.Gain, Slot->effect.Params.Reverb.LateReverbGain,
x, Slot->effect.Params.Reverb.Density, Slot->effect.Params.Reverb.DecayTime,
Slot->effect.Params.Reverb.Diffusion, hfRatio, cw, frequency, State);
if(isEAX)
{
// Update the echo line.
UpdateEchoLine(Slot->effect.Params.Reverb.Gain, Slot->effect.Params.Reverb.LateReverbGain,
Slot->effect.Params.Reverb.EchoTime, Slot->effect.Params.Reverb.DecayTime,
Slot->effect.Params.Reverb.Diffusion, Slot->effect.Params.Reverb.EchoDepth,
hfRatio, cw, frequency, State);
// Update early and late 3D panning.
Update3DPanning(Context->Device, Slot->effect.Params.Reverb.ReflectionsPan,
Slot->effect.Params.Reverb.LateReverbPan, Slot->Gain, State);
}
else
{
ALCdevice *Device = Context->Device;
ALfloat gain = Slot->Gain;
ALuint index;
/* Update channel gains */
gain *= aluSqrt(2.0f/Device->NumChan) * ReverbBoost;
for(index = 0;index < MAXCHANNELS;index++)
State->Gain[index] = 0.0f;
for(index = 0;index < Device->NumChan;index++)
{
enum Channel chan = Device->Speaker2Chan[index];
State->Gain[chan] = gain;
}
}
}
ALvoid aluInitPanning(ALCdevice *Device)
{
ALfloat SpeakerAngle[MAXCHANNELS];
ALfloat (*Matrix)[MAXCHANNELS];
Channel *Speaker2Chan;
ALfloat Alpha, Theta;
ALfloat *PanningLUT;
ALint pos, offset;
ALuint s, s2;
for(s = 0;s < MAXCHANNELS;s++)
{
for(s2 = 0;s2 < MAXCHANNELS;s2++)
Device->ChannelMatrix[s][s2] = ((s==s2) ? 1.0f : 0.0f);
}
Speaker2Chan = Device->Speaker2Chan;
Matrix = Device->ChannelMatrix;
switch(Device->FmtChans)
{
case DevFmtMono:
Matrix[FRONT_LEFT][FRONT_CENTER] = aluSqrt(0.5);
Matrix[FRONT_RIGHT][FRONT_CENTER] = aluSqrt(0.5);
Matrix[SIDE_LEFT][FRONT_CENTER] = aluSqrt(0.5);
Matrix[SIDE_RIGHT][FRONT_CENTER] = aluSqrt(0.5);
Matrix[BACK_LEFT][FRONT_CENTER] = aluSqrt(0.5);
Matrix[BACK_RIGHT][FRONT_CENTER] = aluSqrt(0.5);
Matrix[BACK_CENTER][FRONT_CENTER] = 1.0f;
Device->NumChan = 1;
Speaker2Chan[0] = FRONT_CENTER;
SpeakerAngle[0] = 0.0f * M_PI/180.0f;
break;
case DevFmtStereo:
Matrix[FRONT_CENTER][FRONT_LEFT] = aluSqrt(0.5);
Matrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
Matrix[SIDE_LEFT][FRONT_LEFT] = 1.0f;
Matrix[SIDE_RIGHT][FRONT_RIGHT] = 1.0f;
Matrix[BACK_LEFT][FRONT_LEFT] = 1.0f;
Matrix[BACK_RIGHT][FRONT_RIGHT] = 1.0f;
Matrix[BACK_CENTER][FRONT_LEFT] = aluSqrt(0.5);
Matrix[BACK_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
Device->NumChan = 2;
Speaker2Chan[0] = FRONT_LEFT;
Speaker2Chan[1] = FRONT_RIGHT;
SpeakerAngle[0] = -90.0f * M_PI/180.0f;
SpeakerAngle[1] = 90.0f * M_PI/180.0f;
SetSpeakerArrangement("layout_STEREO", SpeakerAngle, Speaker2Chan, Device->NumChan);
break;
case DevFmtQuad:
Matrix[FRONT_CENTER][FRONT_LEFT] = aluSqrt(0.5);
Matrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
Matrix[SIDE_LEFT][FRONT_LEFT] = aluSqrt(0.5);
Matrix[SIDE_LEFT][BACK_LEFT] = aluSqrt(0.5);
Matrix[SIDE_RIGHT][FRONT_RIGHT] = aluSqrt(0.5);
Matrix[SIDE_RIGHT][BACK_RIGHT] = aluSqrt(0.5);
Matrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
Matrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
Device->NumChan = 4;
Speaker2Chan[0] = BACK_LEFT;
Speaker2Chan[1] = FRONT_LEFT;
Speaker2Chan[2] = FRONT_RIGHT;
Speaker2Chan[3] = BACK_RIGHT;
SpeakerAngle[0] = -135.0f * M_PI/180.0f;
SpeakerAngle[1] = -45.0f * M_PI/180.0f;
SpeakerAngle[2] = 45.0f * M_PI/180.0f;
SpeakerAngle[3] = 135.0f * M_PI/180.0f;
SetSpeakerArrangement("layout_QUAD", SpeakerAngle, Speaker2Chan, Device->NumChan);
break;
case DevFmtX51:
Matrix[SIDE_LEFT][FRONT_LEFT] = aluSqrt(0.5);
Matrix[SIDE_LEFT][BACK_LEFT] = aluSqrt(0.5);
Matrix[SIDE_RIGHT][FRONT_RIGHT] = aluSqrt(0.5);
Matrix[SIDE_RIGHT][BACK_RIGHT] = aluSqrt(0.5);
Matrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
Matrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
Device->NumChan = 5;
Speaker2Chan[0] = BACK_LEFT;
Speaker2Chan[1] = FRONT_LEFT;
Speaker2Chan[2] = FRONT_CENTER;
Speaker2Chan[3] = FRONT_RIGHT;
Speaker2Chan[4] = BACK_RIGHT;
SpeakerAngle[0] = -110.0f * M_PI/180.0f;
SpeakerAngle[1] = -30.0f * M_PI/180.0f;
SpeakerAngle[2] = 0.0f * M_PI/180.0f;
SpeakerAngle[3] = 30.0f * M_PI/180.0f;
SpeakerAngle[4] = 110.0f * M_PI/180.0f;
SetSpeakerArrangement("layout_51CHN", SpeakerAngle, Speaker2Chan, Device->NumChan);
break;
case DevFmtX61:
Matrix[BACK_LEFT][BACK_CENTER] = aluSqrt(0.5);
Matrix[BACK_LEFT][SIDE_LEFT] = aluSqrt(0.5);
Matrix[BACK_RIGHT][BACK_CENTER] = aluSqrt(0.5);
Matrix[BACK_RIGHT][SIDE_RIGHT] = aluSqrt(0.5);
Device->NumChan = 6;
Speaker2Chan[0] = SIDE_LEFT;
Speaker2Chan[1] = FRONT_LEFT;
Speaker2Chan[2] = FRONT_CENTER;
Speaker2Chan[3] = FRONT_RIGHT;
Speaker2Chan[4] = SIDE_RIGHT;
Speaker2Chan[5] = BACK_CENTER;
SpeakerAngle[0] = -90.0f * M_PI/180.0f;
SpeakerAngle[1] = -30.0f * M_PI/180.0f;
SpeakerAngle[2] = 0.0f * M_PI/180.0f;
SpeakerAngle[3] = 30.0f * M_PI/180.0f;
SpeakerAngle[4] = 90.0f * M_PI/180.0f;
SpeakerAngle[5] = 180.0f * M_PI/180.0f;
SetSpeakerArrangement("layout_61CHN", SpeakerAngle, Speaker2Chan, Device->NumChan);
break;
case DevFmtX71:
Matrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
Matrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
Device->NumChan = 7;
Speaker2Chan[0] = BACK_LEFT;
Speaker2Chan[1] = SIDE_LEFT;
Speaker2Chan[2] = FRONT_LEFT;
Speaker2Chan[3] = FRONT_CENTER;
Speaker2Chan[4] = FRONT_RIGHT;
Speaker2Chan[5] = SIDE_RIGHT;
Speaker2Chan[6] = BACK_RIGHT;
SpeakerAngle[0] = -150.0f * M_PI/180.0f;
SpeakerAngle[1] = -90.0f * M_PI/180.0f;
SpeakerAngle[2] = -30.0f * M_PI/180.0f;
SpeakerAngle[3] = 0.0f * M_PI/180.0f;
SpeakerAngle[4] = 30.0f * M_PI/180.0f;
SpeakerAngle[5] = 90.0f * M_PI/180.0f;
SpeakerAngle[6] = 150.0f * M_PI/180.0f;
SetSpeakerArrangement("layout_71CHN", SpeakerAngle, Speaker2Chan, Device->NumChan);
break;
}
if(GetConfigValueBool(NULL, "scalemix", 0))
{
ALfloat maxout = 1.0f;
for(s = 0;s < MAXCHANNELS;s++)
{
ALfloat out = 0.0f;
for(s2 = 0;s2 < MAXCHANNELS;s2++)
out += Device->ChannelMatrix[s2][s];
maxout = __max(maxout, out);
}
maxout = 1.0f/maxout;
for(s = 0;s < MAXCHANNELS;s++)
{
for(s2 = 0;s2 < MAXCHANNELS;s2++)
Device->ChannelMatrix[s2][s] *= maxout;
}
}
PanningLUT = Device->PanningLUT;
for(pos = 0; pos < LUT_NUM; pos++)
{
/* clear all values */
offset = MAXCHANNELS * pos;
for(s = 0; s < MAXCHANNELS; s++)
PanningLUT[offset+s] = 0.0f;
if(Device->NumChan == 1)
{
PanningLUT[offset + Speaker2Chan[0]] = 1.0f;
continue;
}
/* source angle */
Theta = aluLUTpos2Angle(pos);
/* set panning values */
for(s = 0; s < Device->NumChan - 1; s++)
{
if(Theta >= SpeakerAngle[s] && Theta < SpeakerAngle[s+1])
{
/* source between speaker s and speaker s+1 */
Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
(SpeakerAngle[s+1]-SpeakerAngle[s]);
PanningLUT[offset + Speaker2Chan[s]] = cos(Alpha);
PanningLUT[offset + Speaker2Chan[s+1]] = sin(Alpha);
break;
}
}
if(s == Device->NumChan - 1)
{
/* source between last and first speaker */
if(Theta < SpeakerAngle[0])
Theta += 2.0f * M_PI;
Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
(2.0f * M_PI + SpeakerAngle[0]-SpeakerAngle[s]);
PanningLUT[offset + Speaker2Chan[s]] = cos(Alpha);
PanningLUT[offset + Speaker2Chan[0]] = sin(Alpha);
}
}
}