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Remove interpolation of IIR filter coefficients (#1345)
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@derselbst
derselbst authored Jul 27, 2024
1 parent ecb0141 commit 31fa6ea
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Showing 2 changed files with 16 additions and 96 deletions.
106 changes: 16 additions & 90 deletions src/rvoice/fluid_iir_filter.c
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Original file line number Diff line number Diff line change
Expand Up @@ -65,7 +65,6 @@ fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter,
fluid_real_t dsp_a2 = iir_filter->a2;
fluid_real_t dsp_b02 = iir_filter->b02;
fluid_real_t dsp_b1 = iir_filter->b1;
int dsp_filter_coeff_incr_count = iir_filter->filter_coeff_incr_count;

fluid_real_t dsp_centernode;
int dsp_i;
Expand All @@ -83,51 +82,18 @@ fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter,
* doesn't change.
*/

if(dsp_filter_coeff_incr_count > 0)
for(dsp_i = 0; dsp_i < count; dsp_i++)
{
fluid_real_t dsp_a1_incr = iir_filter->a1_incr;
fluid_real_t dsp_a2_incr = iir_filter->a2_incr;
fluid_real_t dsp_b02_incr = iir_filter->b02_incr;
fluid_real_t dsp_b1_incr = iir_filter->b1_incr;


/* Increment is added to each filter coefficient filter_coeff_incr_count times. */
for(dsp_i = 0; dsp_i < count; dsp_i++)
{
/* The filter is implemented in Direct-II form. */
dsp_centernode = dsp_buf[dsp_i] - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2;
dsp_buf[dsp_i] = dsp_b02 * (dsp_centernode + dsp_hist2) + dsp_b1 * dsp_hist1;
dsp_hist2 = dsp_hist1;
dsp_hist1 = dsp_centernode;

if(dsp_filter_coeff_incr_count-- > 0)
{
fluid_real_t old_b02 = dsp_b02;
dsp_a1 += dsp_a1_incr;
dsp_a2 += dsp_a2_incr;
dsp_b02 += dsp_b02_incr;
dsp_b1 += dsp_b1_incr;

/* Compensate history to avoid the filter going havoc with large frequency changes */
if(iir_filter->compensate_incr && FLUID_FABS(dsp_b02) > 0.001f)
{
fluid_real_t compensate = old_b02 / dsp_b02;
dsp_hist1 *= compensate;
dsp_hist2 *= compensate;
}
}
} /* for dsp_i */
}
else /* The filter parameters are constant. This is duplicated to save time. */
{
for(dsp_i = 0; dsp_i < count; dsp_i++)
{
/* The filter is implemented in Direct-II form. */
dsp_centernode = dsp_buf[dsp_i] - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2;
dsp_buf[dsp_i] = dsp_b02 * (dsp_centernode + dsp_hist2) + dsp_b1 * dsp_hist1;
dsp_hist2 = dsp_hist1;
dsp_hist1 = dsp_centernode;
}
/* The filter is implemented in Direct-II form. */
dsp_centernode = dsp_buf[dsp_i] - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2;
dsp_buf[dsp_i] = dsp_b02 * (dsp_centernode + dsp_hist2) + dsp_b1 * dsp_hist1;
dsp_hist2 = dsp_hist1;
dsp_hist1 = dsp_centernode;
/* Alternatively, it could be implemented in Transposed Direct Form II */
// fluid_real_t dsp_input = dsp_buf[dsp_i];
// dsp_buf[dsp_i] = dsp_b02 * dsp_input + dsp_hist1;
// 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];
}

iir_filter->hist1 = dsp_hist1;
Expand All @@ -136,7 +102,6 @@ fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter,
iir_filter->a2 = dsp_a2;
iir_filter->b02 = dsp_b02;
iir_filter->b1 = dsp_b1;
iir_filter->filter_coeff_incr_count = dsp_filter_coeff_incr_count;

fluid_check_fpe("voice_filter");
}
Expand Down Expand Up @@ -320,47 +285,11 @@ fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t *iir_filter,
return;
}

iir_filter->compensate_incr = 0;

if(iir_filter->filter_startup || (transition_samples == 0))
{
/* The filter is calculated, because the voice was started up.
* In this case set the filter coefficients without delay.
*/
iir_filter->a1 = a1_temp;
iir_filter->a2 = a2_temp;
iir_filter->b02 = b02_temp;
iir_filter->b1 = b1_temp;
iir_filter->filter_coeff_incr_count = 0;
iir_filter->filter_startup = 0;
// printf("Setting initial filter coefficients.\n");
}
else
{

/* The filter frequency is changed. Calculate an increment
* factor, so that the new setting is reached after one buffer
* length. x_incr is added to the current value FLUID_BUFSIZE
* times. The length is arbitrarily chosen. Longer than one
* buffer will sacrifice some performance, though. Note: If
* the filter is still too 'grainy', then increase this number
* at will.
*/

iir_filter->a1_incr = (a1_temp - iir_filter->a1) / transition_samples;
iir_filter->a2_incr = (a2_temp - iir_filter->a2) / transition_samples;
iir_filter->b02_incr = (b02_temp - iir_filter->b02) / transition_samples;
iir_filter->b1_incr = (b1_temp - iir_filter->b1) / transition_samples;

if(FLUID_FABS(iir_filter->b02) > 0.0001f)
{
fluid_real_t quota = b02_temp / iir_filter->b02;
iir_filter->compensate_incr = quota < 0.5f || quota > 2.f;
}

/* Have to add the increments filter_coeff_incr_count times. */
iir_filter->filter_coeff_incr_count = transition_samples;
}
iir_filter->a1 = a1_temp;
iir_filter->a2 = a2_temp;
iir_filter->b02 = b02_temp;
iir_filter->b1 = b1_temp;
iir_filter->filter_startup = 0;

fluid_check_fpe("voice_write filter calculation");
}
Expand All @@ -376,9 +305,6 @@ void fluid_iir_filter_calc(fluid_iir_filter_t *iir_filter,
/* calculate the frequency of the resonant filter in Hz */
fres = fluid_ct2hz(iir_filter->fres + fres_mod);

/* FIXME - Still potential for a click during turn on, can we interpolate
between 20khz cutoff and 0 Q? */

/* I removed the optimization of turning the filter off when the
* resonance frequency is above the maximum frequency. Instead, the
* filter frequency is set to a maximum of 0.45 times the sampling
Expand Down
6 changes: 0 additions & 6 deletions src/rvoice/fluid_iir_filter.h
View file
Original file line number Diff line number Diff line change
Expand Up @@ -53,12 +53,6 @@ struct _fluid_iir_filter_t
fluid_real_t a1; /* a0 / a0 */
fluid_real_t a2; /* a1 / a0 */

fluid_real_t b02_incr;
fluid_real_t b1_incr;
fluid_real_t a1_incr;
fluid_real_t a2_incr;
int filter_coeff_incr_count;
int compensate_incr; /* Flag: If set, must compensate history */
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. */
Expand Down

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