diff options
| -rw-r--r-- | Alc/ALu.c | 36 | ||||
| -rw-r--r-- | Alc/alcReverb.c | 31 | ||||
| -rw-r--r-- | OpenAL32/Include/alFilter.h | 16 |
3 files changed, 29 insertions, 54 deletions
@@ -412,8 +412,8 @@ static ALvoid CalcNonAttnSourceParams(const ALCcontext *ALContext, ALsource *ALS ALfloat WetGain[MAX_SENDS]; ALfloat WetGainHF[MAX_SENDS]; ALint NumSends, Frequency; + ALfloat cw; ALint i; - ALfloat cw, a, g; //Get context properties NumSends = ALContext->Device->NumAuxSends; @@ -479,27 +479,16 @@ static ALvoid CalcNonAttnSourceParams(const ALCcontext *ALContext, ALsource *ALS /* 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. */ - g = __max(DryGainHF, 0.01f); - a = 0.0f; - /* Be careful with gains < 0.0001, as that causes the coefficient - * head towards 1, which will flatten the signal */ - if(g < 0.9999f) /* 1-epsilon */ - a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / - (1 - g); - ALSource->Params.iirFilter.coeff = a; + 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 */ - g = __max(WetGainHF[i], 0.1f); - g *= g; - 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); + ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw); ALSource->Params.Send[i].iirFilter.coeff = a; } } @@ -526,7 +515,7 @@ static ALvoid CalcSourceParams(const ALCcontext *ALContext, ALsource *ALSource) ALuint Frequency; ALint NumSends; ALint pos, s, i; - ALfloat cw, a, g; + ALfloat cw; for(i = 0;i < MAX_SENDS;i++) WetGainHF[i] = 1.0f; @@ -862,26 +851,17 @@ static ALvoid CalcSourceParams(const ALCcontext *ALContext, ALsource *ALSource) /* 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. */ - 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->Params.iirFilter.coeff = a; + 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) */ - g = __max(WetGainHF[i], 0.01f); - 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->Params.Send[i].iirFilter.coeff = a; + ALSource->Params.Send[i].iirFilter.coeff = lpCoeffCalc(WetGainHF[i], cw); } } diff --git a/Alc/alcReverb.c b/Alc/alcReverb.c index fed1d5d2..3d6e2734 100644 --- a/Alc/alcReverb.c +++ b/Alc/alcReverb.c @@ -328,24 +328,6 @@ static __inline ALfloat CalcI3DL2HFreq(ALfloat hfRef, ALuint frequency) return cos(2.0f * M_PI * hfRef / frequency); } -/* Calculate the I3DL2 coefficient given the gain and frequency parameters. - * To allow for optimization when using multiple chained filters, the gain - * is not squared in this function. Callers using a single filter should - * square it to produce the correct coefficient. Those using multiple - * filters should find its N-1 root (where N is the number of chained - * filters). - */ -static __inline ALfloat CalcI3DL2Coeff(ALfloat g, ALfloat cw) -{ - ALfloat coeff; - - coeff = 0.0f; - if(g < 0.9999f) // 1-epsilon - coeff = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g); - - return coeff; -} - // 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) @@ -417,11 +399,10 @@ static __inline ALfloat CalcDampingCoeff(ALfloat hfRatio, ALfloat length, ALfloa // Calculate the low-pass coefficient by dividing the HF decay // coefficient by the full decay coefficient. g = CalcDecayCoeff(length, decayTime * hfRatio) / decayCoeff; - g = __max(g, 0.1f); // Damping is done with a 1-pole filter, so g needs to be squared. g *= g; - coeff = CalcI3DL2Coeff(g, cw); + coeff = lpCoeffCalc(g, cw); // Very low decay times will produce minimal output, so apply an // upper bound to the coefficient. @@ -1067,13 +1048,12 @@ static ALvoid VerbUpdate(ALeffectState *effect, ALCcontext *Context, const ALeff { ALverbState *State = (ALverbState*)effect; ALuint frequency = Context->Device->Frequency; - ALfloat cw, g, x, y, hfRatio; + ALfloat cw, x, y, hfRatio; // Calculate the master low-pass filter (from the master effect HF gain). cw = CalcI3DL2HFreq(Effect->Reverb.HFReference, frequency); - g = __max(Effect->Reverb.GainHF, 0.0001f); // This is done with 2 chained 1-pole filters, so no need to square g. - State->LpFilter.coeff = CalcI3DL2Coeff(g, cw); + State->LpFilter.coeff = lpCoeffCalc(Effect->Reverb.GainHF, cw); // Update the initial effect delay. UpdateDelayLine(Effect->Reverb.ReflectionsDelay, @@ -1110,13 +1090,12 @@ static ALvoid EAXVerbUpdate(ALeffectState *effect, ALCcontext *Context, const AL { ALverbState *State = (ALverbState*)effect; ALuint frequency = Context->Device->Frequency; - ALfloat cw, g, x, y, hfRatio; + ALfloat cw, x, y, hfRatio; // Calculate the master low-pass filter (from the master effect HF gain). cw = CalcI3DL2HFreq(Effect->Reverb.HFReference, frequency); - g = __max(Effect->Reverb.GainHF, 0.0001f); // This is done with 2 chained 1-pole filters, so no need to square g. - State->LpFilter.coeff = CalcI3DL2Coeff(g, cw); + State->LpFilter.coeff = lpCoeffCalc(Effect->Reverb.GainHF, cw); // Update the modulator line. UpdateModulator(Effect->Reverb.ModulationTime, diff --git a/OpenAL32/Include/alFilter.h b/OpenAL32/Include/alFilter.h index 754b0b6f..d5ca965a 100644 --- a/OpenAL32/Include/alFilter.h +++ b/OpenAL32/Include/alFilter.h @@ -61,6 +61,22 @@ static __inline ALfloat lpFilter1P(FILTER *iir, ALuint offset, ALfloat input) return output; } +/* Calculates the low-pass filter coefficient given the pre-scaled gain and + * cos(w) value. Note that g should be pre-scaled (sqr(gain) for one-pole, + * sqrt(gain) for four-pole, etc) */ +static __inline 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 */ + g = __max(g, 0.01f); + if(g < 0.9999f) /* 1-epsilon */ + a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / + (1 - g); + + return a; +} #define AL_FILTER_TYPE 0x8001 |