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Linear interpolation of Q
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@derselbst
derselbst committed Nov 16, 2024
1 parent 897dcca commit 0949bdd
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Showing 2 changed files with 93 additions and 58 deletions.
139 changes: 86 additions & 53 deletions src/rvoice/fluid_iir_filter.c
View file
Original file line number Diff line number Diff line change
Expand Up @@ -55,7 +55,7 @@ void
fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter,
fluid_real_t *dsp_buf, int count, fluid_real_t output_rate)
{
if(iir_filter->type == FLUID_IIR_DISABLED || iir_filter->q_lin == 0)
if(iir_filter->type == FLUID_IIR_DISABLED || FLUID_FABS(iir_filter->last_q) <= 0.001)
{
return;
}
Expand All @@ -71,8 +71,8 @@ fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter,
fluid_real_t dsp_b02 = iir_filter->b02;
fluid_real_t dsp_b1 = iir_filter->b1;

fluid_real_t fres_incr = iir_filter->fres_incr;
int fres_incr_count = iir_filter->fres_incr_count;
int q_incr_count = iir_filter->q_incr_count;

fluid_real_t dsp_centernode;
int dsp_i;
Expand Down Expand Up @@ -103,12 +103,20 @@ fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter,
// dsp_hist1 = dsp_b1 * dsp_input - dsp_a1 * dsp_buf[dsp_i] + dsp_hist2;
// dsp_hist2 = dsp_b02 * dsp_input - dsp_a2 * dsp_buf[dsp_i];

if(fres_incr_count > 0)
if(fres_incr_count > 0 || q_incr_count > 0)
{
--fres_incr_count;
iir_filter->last_fres += fres_incr;
if(fres_incr_count > 0)
{
--fres_incr_count;
iir_filter->last_fres += iir_filter->fres_incr;
}
if(q_incr_count > 0)
{
--q_incr_count;
iir_filter->last_q += iir_filter->q_incr;
}

FLUID_LOG(FLUID_DBG, "last_fres: %f | target_fres: %f", iir_filter->last_fres, iir_filter->target_fres);
FLUID_LOG(FLUID_DBG, "last_fres: %.2f Hz | target_fres: %.2f Hz |---| last_q: %.4f | target_q: %.4f", iir_filter->last_fres, iir_filter->target_fres, iir_filter->last_q, iir_filter->target_q);

fluid_iir_filter_calculate_coefficients(iir_filter, output_rate);

Expand All @@ -128,6 +136,7 @@ fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter,
iir_filter->b1 = dsp_b1;

iir_filter->fres_incr_count = fres_incr_count;
iir_filter->q_incr_count = q_incr_count;

fluid_check_fpe("voice_filter");
}
Expand All @@ -154,7 +163,7 @@ fluid_iir_filter_reset(fluid_iir_filter_t *iir_filter)
iir_filter->hist1 = 0;
iir_filter->hist2 = 0;
iir_filter->last_fres = -1.;
iir_filter->q_lin = 0;
iir_filter->last_q = 0;
iir_filter->filter_startup = 1;
}

Expand Down Expand Up @@ -225,37 +234,35 @@ DECLARE_FLUID_RVOICE_FUNCTION(fluid_iir_filter_set_q)
}

FLUID_LOG(FLUID_DBG, "fluid_iir_filter_set_q: Q= %f [linear]",q);
iir_filter->q_lin = q;
iir_filter->filter_gain = 1.0;

if(!(flags & FLUID_IIR_NO_GAIN_AMP))

if(iir_filter->filter_startup)
{
/* SF 2.01 page 59:
*
* The SoundFont specs ask for a gain reduction equal to half the
* height of the resonance peak (Q). For example, for a 10 dB
* resonance peak, the gain is reduced by 5 dB. This is done by
* multiplying the total gain with sqrt(1/Q). `Sqrt' divides dB
* by 2 (100 lin = 40 dB, 10 lin = 20 dB, 3.16 lin = 10 dB etc)
* The gain is later factored into the 'b' coefficients
* (numerator of the filter equation). This gain factor depends
* only on Q, so this is the right place to calculate it.
*/
iir_filter->filter_gain /= FLUID_SQRT(q);
iir_filter->last_q = q;
}
else
{
static const int q_incr_count = FLUID_BUFSIZE;
iir_filter->q_incr = (q - iir_filter->last_q) / (q_incr_count);
iir_filter->q_incr_count = q_incr_count;
}
iir_filter->target_q = q;
}

static FLUID_INLINE void
fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t *iir_filter,
fluid_real_t output_rate)
{
/* FLUID_IIR_Q_LINEAR may switch the filter off by setting Q==0 */
if(iir_filter->q_lin == 0)
// FLUID_IIR_Q_LINEAR may switch the filter off by setting Q==0
// Due to the linear smoothing, last_q may not exactly become zero.
if(FLUID_FABS(iir_filter->last_q) <= 0.001)
{
return;
}
else
{
int flags = iir_filter->flags;
fluid_real_t filter_gain = 1.0f;

/*
* Those equations from Robert Bristow-Johnson's `Cookbook
* formulae for audio EQ biquad filter coefficients', obtained
Expand All @@ -269,28 +276,43 @@ fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t *iir_filter,
(iir_filter->last_fres / output_rate);
fluid_real_t sin_coeff = FLUID_SIN(omega);
fluid_real_t cos_coeff = FLUID_COS(omega);
fluid_real_t alpha_coeff = sin_coeff / (2.0f * iir_filter->q_lin);
fluid_real_t alpha_coeff = sin_coeff / (2.0f * iir_filter->last_q);
fluid_real_t a0_inv = 1.0f / (1.0f + alpha_coeff);

/* Calculate the filter coefficients. All coefficients are
* normalized by a0. Think of `a1' as `a1/a0'.
*
* Here a couple of multiplications are saved by reusing common expressions.
* The original equations should be:
* iir_filter->b0=(1.-cos_coeff)*a0_inv*0.5*iir_filter->filter_gain;
* iir_filter->b1=(1.-cos_coeff)*a0_inv*iir_filter->filter_gain;
* iir_filter->b2=(1.-cos_coeff)*a0_inv*0.5*iir_filter->filter_gain; */
* iir_filter->b0=(1.-cos_coeff)*a0_inv*0.5*filter_gain;
* iir_filter->b1=(1.-cos_coeff)*a0_inv*filter_gain;
* iir_filter->b2=(1.-cos_coeff)*a0_inv*0.5*filter_gain; */

/* "a" coeffs are same for all 3 available filter types */
fluid_real_t a1_temp = -2.0f * cos_coeff * a0_inv;
fluid_real_t a2_temp = (1.0f - alpha_coeff) * a0_inv;

fluid_real_t b02_temp, b1_temp;

if(!(flags & FLUID_IIR_NO_GAIN_AMP))
{
/* SF 2.01 page 59:
*
* The SoundFont specs ask for a gain reduction equal to half the
* height of the resonance peak (Q). For example, for a 10 dB
* resonance peak, the gain is reduced by 5 dB. This is done by
* multiplying the total gain with sqrt(1/Q). `Sqrt' divides dB
* by 2 (100 lin = 40 dB, 10 lin = 20 dB, 3.16 lin = 10 dB etc)
* The gain is later factored into the 'b' coefficients
* (numerator of the filter equation). This gain factor depends
* only on Q, so this is the right place to calculate it.
*/
filter_gain /= FLUID_SQRT(iir_filter->last_q);
}

switch(iir_filter->type)
{
case FLUID_IIR_HIGHPASS:
b1_temp = (1.0f + cos_coeff) * a0_inv * iir_filter->filter_gain;
b1_temp = (1.0f + cos_coeff) * a0_inv * filter_gain;

/* both b0 -and- b2 */
b02_temp = b1_temp * 0.5f;
Expand All @@ -299,7 +321,7 @@ fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t *iir_filter,
break;

case FLUID_IIR_LOWPASS:
b1_temp = (1.0f - cos_coeff) * a0_inv * iir_filter->filter_gain;
b1_temp = (1.0f - cos_coeff) * a0_inv * filter_gain;

/* both b0 -and- b2 */
b02_temp = b1_temp * 0.5f;
Expand All @@ -324,7 +346,13 @@ void fluid_iir_filter_calc(fluid_iir_filter_t *iir_filter,
fluid_real_t output_rate,
fluid_real_t fres_mod)
{
unsigned int calc_coeff_flag = FALSE;
fluid_real_t fres, fres_diff;

if(iir_filter->type == FLUID_IIR_DISABLED)
{
return;
}

/* calculate the frequency of the resonant filter in Hz */
fres = fluid_ct2hz(iir_filter->fres + fres_mod);
Expand All @@ -348,34 +376,39 @@ void fluid_iir_filter_calc(fluid_iir_filter_t *iir_filter,
fres = 5.f;
}

FLUID_LOG(FLUID_DBG, "%f + %f = %f cents = %f Hz | Q: %f", iir_filter->fres, fres_mod, iir_filter->fres + fres_mod, fres, iir_filter->q_lin);
FLUID_LOG(FLUID_DBG, "%f + %f = %f cents = %f Hz | Q: %f", iir_filter->fres, fres_mod, iir_filter->fres + fres_mod, fres, iir_filter->last_q);

/* if filter enabled and there is a significant frequency change.. */
fres_diff = fres - iir_filter->last_fres;
if(iir_filter->type != FLUID_IIR_DISABLED && FLUID_FABS(fres_diff) > 0.01f)
if(iir_filter->filter_startup)
{
/* The filter coefficients have to be recalculated (filter
* parameters have changed). Recalculation for various reasons is
* forced by setting last_fres to -1. The flag filter_startup
* indicates, that the DSP loop runs for the first time, in this
* case, the filter is set directly, instead of smoothly fading
* between old and new settings. */
if(iir_filter->filter_startup)
{
iir_filter->fres_incr_count = 0;
iir_filter->last_fres = fres;
iir_filter->filter_startup = 0;
}
else
{
static const int fres_incr_count = FLUID_BUFSIZE;
iir_filter->fres_incr = fres_diff / (fres_incr_count);
iir_filter->fres_incr_count = fres_incr_count;
}
// The filer was just starting up, make sure to calculate initial coefficients for the initial Q value, even though the fres may not have changed
calc_coeff_flag = TRUE;

iir_filter->fres_incr_count = 0;
iir_filter->last_fres = fres;
iir_filter->filter_startup = 0;
}
else if(FLUID_FABS(fres_diff) > 0.01f)
{
static const int fres_incr_count = FLUID_BUFSIZE;
iir_filter->fres_incr = fres_diff / (fres_incr_count);
iir_filter->fres_incr_count = fres_incr_count;
iir_filter->target_fres = fres;
fluid_iir_filter_calculate_coefficients(iir_filter, output_rate);

// The filter coefficients have to be recalculated (filter cutoff has changed).
calc_coeff_flag = TRUE;
}
else
{
// We do not account for any change of Q here - if it was changed q_incro_count will be non-zero and recalculating the coeffs
// will be taken care of in fluid_iir_filter_apply().
}

if(calc_coeff_flag)
{
fluid_iir_filter_calculate_coefficients(iir_filter, output_rate);
}

fluid_check_fpe("voice_write DSP coefficients");

Expand Down
12 changes: 7 additions & 5 deletions src/rvoice/fluid_iir_filter.h
View file
Original file line number Diff line number Diff line change
Expand Up @@ -54,16 +54,18 @@ struct _fluid_iir_filter_t
fluid_real_t a2; /* a1 / a0 */

fluid_real_t hist1, hist2; /* Sample history for the IIR filter */
int filter_startup; /* Flag: If set, the filter will be set directly. Else it changes smoothly. */
int filter_startup; /* Flag: If set, the filter parameters will be set directly. Else it changes smoothly. */

fluid_real_t fres; /* The desired resonance frequency, in absolute cents, this filter is currently set to */
fluid_real_t last_fres; /* The filter's currently (smoothed out) resonance frequency in Hz, which will converge towards its target fres once fres_incr_count has become zero */
fluid_real_t last_fres; /* The filter's current (smoothed out) resonance frequency in Hz, which will converge towards its target fres once fres_incr_count has become zero */
fluid_real_t target_fres; /* The filter's target fres, that last_fres should converge towards - for debugging only */
fluid_real_t fres_incr; /* The linear increment of fres on each sample */
fluid_real_t fres_incr; /* The linear increment of fres each sample */
int fres_incr_count; /* The number of samples left for the smoothed last_fres adjustment to complete */

fluid_real_t q_lin; /* the q-factor on a linear scale */
fluid_real_t filter_gain; /* Gain correction factor, depends on q */
fluid_real_t last_q; /* The filter's current (smoothed) Q-factor (or "bandwidth", or "resonance-friendlyness") on a linear scale. Just like fres, this will converge towards its target Q once q_incr_count has become zero. */
fluid_real_t target_q; /* The filter's target Q - for debugging only */
fluid_real_t q_incr; /* The linear increment of q each sample */
int q_incr_count; /* The number of samples left for the smoothed Q adjustment to complete */
};

#endif
Expand Down

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