diff options
Diffstat (limited to 'Alc/effects')
-rw-r--r-- | Alc/effects/reverb.c | 327 |
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 |