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Inline fluid_iir_filter_apply
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@derselbst
derselbst committed Dec 22, 2024
1 parent 66cc8b2 commit 4b5c8c4
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204 changes: 0 additions & 204 deletions src/rvoice/fluid_iir_filter.c
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#include "fluid_sys.h"
#include "fluid_conv.h"

static const fluid_real_t Q_MIN = 0.001f;

static FLUID_INLINE void
fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t *iir_filter, fluid_real_t output_rate,
fluid_real_t *a1_out, fluid_real_t *a2_out,
fluid_real_t *b02_out, fluid_real_t *b1_out);


/**
* Applies a low- or high-pass filter with variable cutoff frequency and quality factor
* for a given biquad transfer function:
* b0 + b1*z^-1 + b2*z^-2
* H(z) = ------------------------
* a0 + a1*z^-1 + a2*z^-2
*
* Also modifies filter state accordingly.
* @param iir_filter Filter parameter
* @param dsp_buf Pointer to the synthesized audio data
* @param count Count of samples in dsp_buf
*/
/*
* Variable description:
* - dsp_a1, dsp_a2: Filter coefficients for the the previously filtered output signal
* - dsp_b0, dsp_b1, dsp_b2: Filter coefficients for input signal
* - coefficients normalized to a0
*
* A couple of variables are used internally, their results are discarded:
* - dsp_i: Index through the output buffer
* - dsp_centernode: delay line for the IIR filter
* - dsp_hist1: same
* - dsp_hist2: same
*/
void
fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter,
fluid_real_t *dsp_buf, int count, fluid_real_t output_rate)
{
// 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 (iir_filter->type == FLUID_IIR_DISABLED || FLUID_FABS(iir_filter->last_q) < Q_MIN)
{
return;
}
else
{
/* IIR filter sample history */
fluid_real_t dsp_hist1 = iir_filter->hist1;
fluid_real_t dsp_hist2 = iir_filter->hist2;

/* IIR filter coefficients */
fluid_real_t dsp_a1 = iir_filter->a1;
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 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;

/* filter (implement the voice filter according to SoundFont standard) */

/* Check for denormal number (too close to zero). */
if(FLUID_FABS(dsp_hist1) < 1e-20f)
{
dsp_hist1 = 0.0f; /* FIXME JMG - Is this even needed? */
}

/* Two versions of the filter loop. One, while the filter is
* changing towards its new setting. The other, if the filter
* doesn't change.
*/

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;
/* 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];

if(fres_incr_count > 0 || q_incr_count > 0)
{
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;
}

LOG_FILTER("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, &dsp_a1, &dsp_a2, &dsp_b02, &dsp_b1);
}
}

iir_filter->hist1 = dsp_hist1;
iir_filter->hist2 = dsp_hist2;
iir_filter->a1 = dsp_a1;
iir_filter->a2 = dsp_a2;
iir_filter->b02 = dsp_b02;
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");
}
}

DECLARE_FLUID_RVOICE_FUNCTION(fluid_iir_filter_init)
{
Expand Down Expand Up @@ -250,92 +132,6 @@ DECLARE_FLUID_RVOICE_FUNCTION(fluid_iir_filter_set_q)
#endif
}

static FLUID_INLINE void
fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t *iir_filter,
fluid_real_t output_rate,
fluid_real_t *a1_out, fluid_real_t *a2_out,
fluid_real_t *b02_out, fluid_real_t *b1_out)
{
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
* from Harmony-central.com / Computer / Programming. They are
* the result of the bilinear transform on an analogue filter
* prototype. To quote, `BLT frequency warping has been taken
* into account for both significant frequency relocation and for
* bandwidth readjustment'. */

fluid_real_t omega = (fluid_real_t)(2.0 * M_PI) *
(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->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*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 * filter_gain;

/* both b0 -and- b2 */
b02_temp = b1_temp * 0.5f;

b1_temp *= -1.0f;
break;

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

/* both b0 -and- b2 */
b02_temp = b1_temp * 0.5f;
break;

default:
/* filter disabled, should never get here */
return;
}

*a1_out = a1_temp;
*a2_out = a2_temp;
*b02_out = b02_temp;
*b1_out = b1_temp;

fluid_check_fpe("voice_write filter calculation");
}

void fluid_iir_filter_calc(fluid_iir_filter_t *iir_filter,
fluid_real_t output_rate,
fluid_real_t fres_mod)
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