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-rw-r--r--Alc/effects/reverb.c327
1 files changed, 18 insertions, 309 deletions
diff --git a/Alc/effects/reverb.c b/Alc/effects/reverb.c
index 9fb74b76..9b507b3d 100644
--- a/Alc/effects/reverb.c
+++ b/Alc/effects/reverb.c
@@ -232,12 +232,7 @@ typedef struct T60Filter {
* frequencies, and one to control the high frequencies. The HF filter also
* adjusts the overall output gain, affecting the remaining mid-band.
*/
- ALfloat HFCoeffs[3];
- ALfloat LFCoeffs[3];
-
- /* The HF and LF filters each keep a delay component. */
- ALfloat HFState;
- ALfloat LFState;
+ BiquadFilter HFFilter, LFFilter;
} T60Filter;
typedef struct EarlyReflections {
@@ -400,13 +395,8 @@ static void ALreverbState_Construct(ALreverbState *state)
state->Late.VecAp.Offset[i][0] = 0;
state->Late.VecAp.Offset[i][1] = 0;
- for(j = 0;j < 3;j++)
- {
- state->Late.T60[i].HFCoeffs[j] = 0.0f;
- state->Late.T60[i].LFCoeffs[j] = 0.0f;
- }
- state->Late.T60[i].HFState = 0.0f;
- state->Late.T60[i].LFState = 0.0f;
+ BiquadFilter_clear(&state->Late.T60[i].HFFilter);
+ BiquadFilter_clear(&state->Late.T60[i].LFFilter);
}
for(i = 0;i < NUM_LINES;i++)
@@ -648,284 +638,27 @@ static ALfloat CalcLimitedHfRatio(const ALfloat hfRatio, const ALfloat airAbsorp
return minf(limitRatio, hfRatio);
}
-/* Calculates the first-order high-pass coefficients following the I3DL2
- * reference model. This is the transfer function:
- *
- * 1 - z^-1
- * H(z) = p ------------
- * 1 - p z^-1
- *
- * And this is the I3DL2 coefficient calculation given gain (g) and reference
- * angular frequency (w):
- *
- * g
- * p = ------------------------------------------------------
- * g cos(w) + sqrt((cos(w) - 1) (g^2 cos(w) + g^2 - 2))
- *
- * The coefficient is applied to the partial differential filter equation as:
- *
- * c_0 = p
- * c_1 = -p
- * c_2 = p
- * y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
- *
- */
-static inline void CalcHighpassCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
-{
- ALfloat g, g2, cw, p;
-
- if(gain >= 1.0f)
- {
- coeffs[0] = 1.0f;
- coeffs[1] = 0.0f;
- coeffs[2] = 0.0f;
- return;
- }
-
- g = maxf(0.001f, gain);
- g2 = g * g;
- cw = cosf(w);
- p = g / (g*cw + sqrtf((cw - 1.0f) * (g2*cw + g2 - 2.0f)));
-
- coeffs[0] = p;
- coeffs[1] = -p;
- coeffs[2] = p;
-}
-
-/* Calculates the first-order low-pass coefficients following the I3DL2
- * reference model. This is the transfer function:
- *
- * (1 - a) z^0
- * H(z) = ----------------
- * 1 z^0 - a z^-1
- *
- * And this is the I3DL2 coefficient calculation given gain (g) and reference
- * angular frequency (w):
- *
- * 1 - g^2 cos(w) - sqrt(2 g^2 (1 - cos(w)) - g^4 (1 - cos(w)^2))
- * a = ----------------------------------------------------------------
- * 1 - g^2
- *
- * The coefficient is applied to the partial differential filter equation as:
- *
- * c_0 = 1 - a
- * c_1 = 0
- * c_2 = a
- * y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
- *
- */
-static inline void CalcLowpassCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
-{
- ALfloat g, g2, cw, a;
-
- if(gain >= 1.0f)
- {
- coeffs[0] = 1.0f;
- coeffs[1] = 0.0f;
- coeffs[2] = 0.0f;
- return;
- }
-
- /* Be careful with gains < 0.001, as that causes the coefficient
- * to head towards 1, which will flatten the signal. */
- g = maxf(0.001f, gain);
- g2 = g * g;
- cw = cosf(w);
- a = (1.0f - g2*cw - sqrtf((2.0f*g2*(1.0f - cw)) - g2*g2*(1.0f - cw*cw))) /
- (1.0f - g2);
-
- coeffs[0] = 1.0f - a;
- coeffs[1] = 0.0f;
- coeffs[2] = a;
-}
-
-/* Calculates the first-order low-shelf coefficients. The shelf filters are
- * used in place of low/high-pass filters to preserve the mid-band. This is
- * the transfer function:
- *
- * a_0 + a_1 z^-1
- * H(z) = ----------------
- * 1 + b_1 z^-1
- *
- * And these are the coefficient calculations given cut gain (g) and a center
- * angular frequency (w):
- *
- * sin(0.5 (pi - w) - 0.25 pi)
- * p = -----------------------------
- * sin(0.5 (pi - w) + 0.25 pi)
- *
- * g + 1 g + 1
- * a = ------- + sqrt((-------)^2 - 1)
- * g - 1 g - 1
- *
- * 1 + g + (1 - g) a
- * b_0 = -------------------
- * 2
- *
- * 1 - g + (1 + g) a
- * b_1 = -------------------
- * 2
- *
- * The coefficients are applied to the partial differential filter equation
- * as:
- *
- * b_0 + p b_1
- * c_0 = -------------
- * 1 + p a
- *
- * -(b_1 + p b_0)
- * c_1 = ----------------
- * 1 + p a
- *
- * p + a
- * c_2 = ---------
- * 1 + p a
- *
- * y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
- *
- */
-static inline void CalcLowShelfCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
-{
- ALfloat g, rw, p, n;
- ALfloat alpha, beta0, beta1;
-
- if(gain >= 1.0f)
- {
- coeffs[0] = 1.0f;
- coeffs[1] = 0.0f;
- coeffs[2] = 0.0f;
- return;
- }
-
- g = maxf(0.001f, gain);
- rw = F_PI - w;
- p = sinf(0.5f*rw - 0.25f*F_PI) / sinf(0.5f*rw + 0.25f*F_PI);
- n = (g + 1.0f) / (g - 1.0f);
- alpha = n + sqrtf(n*n - 1.0f);
- beta0 = (1.0f + g + (1.0f - g)*alpha) / 2.0f;
- beta1 = (1.0f - g + (1.0f + g)*alpha) / 2.0f;
-
- coeffs[0] = (beta0 + p*beta1) / (1.0f + p*alpha);
- coeffs[1] = -(beta1 + p*beta0) / (1.0f + p*alpha);
- coeffs[2] = (p + alpha) / (1.0f + p*alpha);
-}
-
-/* Calculates the first-order high-shelf coefficients. The shelf filters are
- * used in place of low/high-pass filters to preserve the mid-band. This is
- * the transfer function:
- *
- * a_0 + a_1 z^-1
- * H(z) = ----------------
- * 1 + b_1 z^-1
- *
- * And these are the coefficient calculations given cut gain (g) and a center
- * angular frequency (w):
- *
- * sin(0.5 w - 0.25 pi)
- * p = ----------------------
- * sin(0.5 w + 0.25 pi)
- *
- * g + 1 g + 1
- * a = ------- + sqrt((-------)^2 - 1)
- * g - 1 g - 1
- *
- * 1 + g + (1 - g) a
- * b_0 = -------------------
- * 2
- *
- * 1 - g + (1 + g) a
- * b_1 = -------------------
- * 2
- *
- * The coefficients are applied to the partial differential filter equation
- * as:
- *
- * b_0 + p b_1
- * c_0 = -------------
- * 1 + p a
- *
- * b_1 + p b_0
- * c_1 = -------------
- * 1 + p a
- *
- * -(p + a)
- * c_2 = ----------
- * 1 + p a
- *
- * y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
- *
- */
-static inline void CalcHighShelfCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
-{
- ALfloat g, p, n;
- ALfloat alpha, beta0, beta1;
-
- if(gain >= 1.0f)
- {
- coeffs[0] = 1.0f;
- coeffs[1] = 0.0f;
- coeffs[2] = 0.0f;
- return;
- }
-
- g = maxf(0.001f, gain);
- p = sinf(0.5f*w - 0.25f*F_PI) / sinf(0.5f*w + 0.25f*F_PI);
- n = (g + 1.0f) / (g - 1.0f);
- alpha = n + sqrtf(n*n - 1.0f);
- beta0 = (1.0f + g + (1.0f - g)*alpha) / 2.0f;
- beta1 = (1.0f - g + (1.0f + g)*alpha) / 2.0f;
-
- coeffs[0] = (beta0 + p*beta1) / (1.0f + p*alpha);
- coeffs[1] = (beta1 + p*beta0) / (1.0f + p*alpha);
- coeffs[2] = -(p + alpha) / (1.0f + p*alpha);
-}
/* Calculates the 3-band T60 damping coefficients for a particular delay line
- * of specified length using a combination of two low/high-pass/shelf or
- * pass-through filter sections (producing 3 coefficients each) given decay
- * times for each band split at two (LF/HF) reference frequencies (w).
+ * of specified length, using a combination of two shelf filter sections given
+ * decay times for each band split at two reference frequencies.
*/
static void CalcT60DampingCoeffs(const ALfloat length, const ALfloat lfDecayTime,
const ALfloat mfDecayTime, const ALfloat hfDecayTime,
- const ALfloat lfW, const ALfloat hfW, ALfloat lfcoeffs[3],
- ALfloat hfcoeffs[3])
+ const ALfloat lf0norm, const ALfloat hf0norm,
+ T60Filter *filter)
{
ALfloat lfGain = CalcDecayCoeff(length, lfDecayTime);
ALfloat mfGain = CalcDecayCoeff(length, mfDecayTime);
ALfloat hfGain = CalcDecayCoeff(length, hfDecayTime);
- if(lfGain <= mfGain)
- {
- CalcHighpassCoeffs(lfGain / mfGain, lfW, lfcoeffs);
- if(mfGain >= hfGain)
- {
- CalcLowpassCoeffs(hfGain / mfGain, hfW, hfcoeffs);
- hfcoeffs[0] *= mfGain; hfcoeffs[1] *= mfGain;
- }
- else
- {
- CalcLowShelfCoeffs(mfGain / hfGain, hfW, hfcoeffs);
- hfcoeffs[0] *= hfGain; hfcoeffs[1] *= hfGain;
- }
- }
- else
- {
- CalcHighShelfCoeffs(mfGain / lfGain, lfW, lfcoeffs);
- if(mfGain >= hfGain)
- {
- CalcLowpassCoeffs(hfGain / mfGain, hfW, hfcoeffs);
- hfcoeffs[0] *= lfGain; hfcoeffs[1] *= lfGain;
- }
- else
- {
- ALfloat hg = mfGain / lfGain;
- ALfloat lg = mfGain / hfGain;
- ALfloat mg = maxf(lfGain, hfGain) / maxf(hg, lg);
-
- CalcLowShelfCoeffs(lg, hfW, hfcoeffs);
- hfcoeffs[0] *= mg; hfcoeffs[1] *= mg;
- }
- }
+ BiquadFilter_setParams(&filter->LFFilter, BiquadType_LowShelf, lfGain/mfGain, lf0norm,
+ calc_rcpQ_from_slope(lfGain/mfGain, 1.0f));
+ BiquadFilter_setParams(&filter->HFFilter, BiquadType_HighShelf, hfGain/mfGain, hf0norm,
+ calc_rcpQ_from_slope(hfGain/mfGain, 1.0f));
+ filter->HFFilter.b0 *= mfGain;
+ filter->HFFilter.b1 *= mfGain;
+ filter->HFFilter.b2 *= mfGain;
}
/* Update the offsets for the main effect delay line. */
@@ -1064,8 +797,7 @@ static ALvoid UpdateLateLines(const ALfloat density, const ALfloat diffusion, co
/* Calculate the T60 damping coefficients for each line. */
CalcT60DampingCoeffs(length, lfDecayTime, mfDecayTime, hfDecayTime,
- lf0norm*F_TAU, hf0norm*F_TAU, Late->T60[i].LFCoeffs,
- Late->T60[i].HFCoeffs);
+ lf0norm, hf0norm, &Late->T60[i]);
}
}
@@ -1583,32 +1315,9 @@ static void EarlyReflection_Faded(ALreverbState *State, ALsizei offset, const AL
/* Applies the two T60 damping filter sections. */
static inline void LateT60Filter(ALfloat *restrict samples, const ALsizei todo, T60Filter *filter)
{
- const ALfloat hfb0 = filter->HFCoeffs[0];
- const ALfloat hfb1 = filter->HFCoeffs[1];
- const ALfloat hfa1 = filter->HFCoeffs[2];
- const ALfloat lfb0 = filter->LFCoeffs[0];
- const ALfloat lfb1 = filter->LFCoeffs[1];
- const ALfloat lfa1 = filter->LFCoeffs[2];
- ALfloat hfz = filter->HFState;
- ALfloat lfz = filter->LFState;
- ALsizei i;
-
- ASSUME(todo > 0);
-
- for(i = 0;i < todo;i++)
- {
- ALfloat in = samples[i];
- ALfloat out = in*hfb0 + hfz;
- hfz = in*hfb1 + out*hfa1;
-
- in = out;
- out = in*lfb0 + lfz;
- lfz = in*lfb1 + out*lfa1;
-
- samples[i] = out;
- }
- filter->HFState = hfz;
- filter->LFState = lfz;
+ ALfloat temp[MAX_UPDATE_SAMPLES];
+ BiquadFilter_process(&filter->HFFilter, temp, samples, todo);
+ BiquadFilter_process(&filter->LFFilter, samples, temp, todo);
}
/* This generates the reverb tail using a modified feed-back delay network