void fluid_iir_filter_calc(fluid_iir_filter_t* iir_filter,
fluid_real_t output_rate,
fluid_real_t fres_mod)
{
fluid_real_t fres;
/* 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 frequence is above the maximum frequency. Instead, the
* filter frequency is set to a maximum of 0.45 times the sampling
* rate. For a 44100 kHz sampling rate, this amounts to 19845
* Hz. The reason is that there were problems with anti-aliasing when the
* synthesizer was run at lower sampling rates. Thanks to Stephan
* Tassart for pointing me to this bug. By turning the filter on and
* clipping the maximum filter frequency at 0.45*srate, the filter
* is used as an anti-aliasing filter. */
if (fres > 0.45f * output_rate)
fres = 0.45f * output_rate;
else if (fres < 5)
fres = 5;
/* if filter enabled and there is a significant frequency change.. */
if ((abs (fres - iir_filter->last_fres) > 0.01))
{
/* 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. */
iir_filter->last_fres = fres;
fluid_iir_filter_calculate_coefficients(iir_filter, FLUID_BUFSIZE,
output_rate);
}
fluid_check_fpe ("voice_write DSP coefficients");
}
void Voice::dsp_float_config()
{
/* Initialize the coefficients for the interpolation. The math comes
* from a mail, posted by Olli Niemitalo to the music-dsp mailing
* list (I found it in the music-dsp archives
* http://www.smartelectronix.com/musicdsp/). */
for (int i = 0; i < FLUID_INTERP_MAX; i++) {
double x = (double) i / (double) FLUID_INTERP_MAX;
interp_coeff[i][0] = (float)(x * (-0.5 + x * (1 - 0.5 * x)));
interp_coeff[i][1] = (float)(1.0 + x * x * (1.5 * x - 2.5));
interp_coeff[i][2] = (float)(x * (0.5 + x * (2.0 - 1.5 * x)));
interp_coeff[i][3] = (float)(0.5 * x * x * (x - 1.0));
interp_coeff_linear[i][0] = (float)(1.0 - x);
interp_coeff_linear[i][1] = (float)x;
}
/* i: Offset in terms of whole samples */
for (int i = 0; i < SINC_INTERP_ORDER; i++) {
/* i2: Offset in terms of fractional samples ('subsamples') */
for (int i2 = 0; i2 < FLUID_INTERP_MAX; i2++) {
/* center on middle of table */
double i_shifted = (double)i - ((double)SINC_INTERP_ORDER / 2.0)
+ (double)i2 / (double)FLUID_INTERP_MAX;
/* sinc(0) cannot be calculated straightforward (limit needed for 0/0) */
double v;
if (fabs (i_shifted) > 0.000001) {
v = (float)sin (i_shifted * M_PI) / (M_PI * i_shifted);
/* Hamming window */
v *= (float)0.5 * (1.0 + cos (2.0 * M_PI * i_shifted / (float)SINC_INTERP_ORDER));
}
else
v = 1.0;
sinc_table7[FLUID_INTERP_MAX - i2 - 1][i] = v;
}
}
fluid_check_fpe("interpolation table calculation");
}
/**
* @return -1 if voice has finished, 0 if it's currently quiet, 1 otherwise
*/
static inline int
fluid_rvoice_calc_amp(fluid_rvoice_t* voice)
{
fluid_real_t target_amp; /* target amplitude */
if (fluid_adsr_env_get_section(&voice->envlfo.volenv) == FLUID_VOICE_ENVDELAY)
return -1; /* The volume amplitude is in hold phase. No sound is produced. */
if (fluid_adsr_env_get_section(&voice->envlfo.volenv) == FLUID_VOICE_ENVATTACK)
{
/* the envelope is in the attack section: ramp linearly to max value.
* A positive modlfo_to_vol should increase volume (negative attenuation).
*/
target_amp = fluid_atten2amp (voice->dsp.attenuation)
* fluid_cb2amp (fluid_lfo_get_val(&voice->envlfo.modlfo) * -voice->envlfo.modlfo_to_vol)
* fluid_adsr_env_get_val(&voice->envlfo.volenv);
}
else
{
fluid_real_t amplitude_that_reaches_noise_floor;
fluid_real_t amp_max;
target_amp = fluid_atten2amp (voice->dsp.attenuation)
* fluid_cb2amp (960.0f * (1.0f - fluid_adsr_env_get_val(&voice->envlfo.volenv))
+ fluid_lfo_get_val(&voice->envlfo.modlfo) * -voice->envlfo.modlfo_to_vol);
/* We turn off a voice, if the volume has dropped low enough. */
/* A voice can be turned off, when an estimate for the volume
* (upper bound) falls below that volume, that will drop the
* sample below the noise floor.
*/
/* If the loop amplitude is known, we can use it if the voice loop is within
* the sample loop
*/
/* Is the playing pointer already in the loop? */
if (voice->dsp.has_looped)
amplitude_that_reaches_noise_floor = voice->dsp.amplitude_that_reaches_noise_floor_loop;
else
amplitude_that_reaches_noise_floor = voice->dsp.amplitude_that_reaches_noise_floor_nonloop;
/* voice->attenuation_min is a lower boundary for the attenuation
* now and in the future (possibly 0 in the worst case). Now the
* amplitude of sample and volenv cannot exceed amp_max (since
* volenv_val can only drop):
*/
amp_max = fluid_atten2amp (voice->dsp.min_attenuation_cB) *
fluid_adsr_env_get_val(&voice->envlfo.volenv);
/* And if amp_max is already smaller than the known amplitude,
* which will attenuate the sample below the noise floor, then we
* can safely turn off the voice. Duh. */
if (amp_max < amplitude_that_reaches_noise_floor)
{
return 0;
}
}
/* Volume increment to go from voice->amp to target_amp in FLUID_BUFSIZE steps */
voice->dsp.amp_incr = (target_amp - voice->dsp.amp) / FLUID_BUFSIZE;
fluid_check_fpe ("voice_write amplitude calculation");
/* no volume and not changing? - No need to process */
if ((voice->dsp.amp == 0.0f) && (voice->dsp.amp_incr == 0.0f))
return -1;
return 1;
}
/**
* Synthesize a voice to a buffer.
*
* @param voice rvoice to synthesize
* @param dsp_buf Audio buffer to synthesize to (#FLUID_BUFSIZE in length)
* @return Count of samples written to dsp_buf. (-1 means voice is currently
* quiet, 0 .. #FLUID_BUFSIZE-1 means voice finished.)
*
* Panning, reverb and chorus are processed separately. The dsp interpolation
* routine is in (fluid_dsp_float.c).
*/
int
fluid_rvoice_write (fluid_rvoice_t* voice, fluid_real_t *dsp_buf)
{
int ticks = voice->envlfo.ticks;
int count;
/******************* sample sanity check **********/
if (!voice->dsp.sample)
return 0;
if (voice->dsp.check_sample_sanity_flag)
fluid_rvoice_check_sample_sanity(voice);
/******************* noteoff check ****************/
if (voice->envlfo.noteoff_ticks != 0 &&
voice->envlfo.ticks >= voice->envlfo.noteoff_ticks) {
fluid_rvoice_noteoff(voice, 0);
}
voice->envlfo.ticks += FLUID_BUFSIZE;
/******************* vol env **********************/
fluid_adsr_env_calc(&voice->envlfo.volenv, 1);
fluid_check_fpe ("voice_write vol env");
if (fluid_adsr_env_get_section(&voice->envlfo.volenv) == FLUID_VOICE_ENVFINISHED)
return 0;
/******************* mod env **********************/
fluid_adsr_env_calc(&voice->envlfo.modenv, 0);
fluid_check_fpe ("voice_write mod env");
/******************* lfo **********************/
fluid_lfo_calc(&voice->envlfo.modlfo, ticks);
fluid_check_fpe ("voice_write mod LFO");
fluid_lfo_calc(&voice->envlfo.viblfo, ticks);
fluid_check_fpe ("voice_write vib LFO");
/******************* amplitude **********************/
count = fluid_rvoice_calc_amp(voice);
if (count <= 0)
return count;
/******************* phase **********************/
/* Calculate the number of samples, that the DSP loop advances
* through the original waveform with each step in the output
* buffer. It is the ratio between the frequencies of original
* waveform and output waveform.*/
voice->dsp.phase_incr = fluid_ct2hz_real(voice->dsp.pitch +
fluid_lfo_get_val(&voice->envlfo.modlfo) * voice->envlfo.modlfo_to_pitch
+ fluid_lfo_get_val(&voice->envlfo.viblfo) * voice->envlfo.viblfo_to_pitch
+ fluid_adsr_env_get_val(&voice->envlfo.modenv) * voice->envlfo.modenv_to_pitch)
/ voice->dsp.root_pitch_hz;
fluid_check_fpe ("voice_write phase calculation");
/* if phase_incr is not advancing, set it to the minimum fraction value (prevent stuckage) */
if (voice->dsp.phase_incr == 0) voice->dsp.phase_incr = 1;
voice->dsp.is_looping = voice->dsp.samplemode == FLUID_LOOP_DURING_RELEASE
|| (voice->dsp.samplemode == FLUID_LOOP_UNTIL_RELEASE
&& fluid_adsr_env_get_section(&voice->envlfo.volenv) < FLUID_VOICE_ENVRELEASE);
/*********************** run the dsp chain ************************
* The sample is mixed with the output buffer.
* The buffer has to be filled from 0 to FLUID_BUFSIZE-1.
* Depending on the position in the loop and the loop size, this
* may require several runs. */
voice->dsp.dsp_buf = dsp_buf;
switch (voice->dsp.interp_method)
{
case FLUID_INTERP_NONE:
count = fluid_rvoice_dsp_interpolate_none (&voice->dsp);
break;
case FLUID_INTERP_LINEAR:
count = fluid_rvoice_dsp_interpolate_linear (&voice->dsp);
break;
case FLUID_INTERP_4THORDER:
default:
count = fluid_rvoice_dsp_interpolate_4th_order (&voice->dsp);
break;
case FLUID_INTERP_7THORDER:
count = fluid_rvoice_dsp_interpolate_7th_order (&voice->dsp);
break;
}
fluid_check_fpe ("voice_write interpolation");
if (count == 0)
return count;
/*************** resonant filter ******************/
fluid_iir_filter_calc(&voice->resonant_filter, voice->dsp.output_rate,
fluid_lfo_get_val(&voice->envlfo.modlfo) * voice->envlfo.modlfo_to_fc +
fluid_adsr_env_get_val(&voice->envlfo.modenv) * voice->envlfo.modenv_to_fc);
fluid_iir_filter_apply(&voice->resonant_filter, dsp_buf, count);
return count;
}
/* Purpose:
*
* Make sure, that sample start / end point and loop points are in
* proper order. When starting up, calculate the initial phase.
* TODO: Investigate whether this can be moved from rvoice to voice.
*/
static void
fluid_rvoice_check_sample_sanity(fluid_rvoice_t* voice)
{
int min_index_nonloop=(int) voice->dsp.sample->start;
int max_index_nonloop=(int) voice->dsp.sample->end;
/* make sure we have enough samples surrounding the loop */
int min_index_loop=(int) voice->dsp.sample->start + FLUID_MIN_LOOP_PAD;
int max_index_loop=(int) voice->dsp.sample->end - FLUID_MIN_LOOP_PAD + 1; /* 'end' is last valid sample, loopend can be + 1 */
fluid_check_fpe("voice_check_sample_sanity start");
if (!voice->dsp.check_sample_sanity_flag){
return;
}
#if 0
printf("Sample from %i to %i\n",voice->dsp.sample->start, voice->dsp.sample->end);
printf("Sample loop from %i %i\n",voice->dsp.sample->loopstart, voice->dsp.sample->loopend);
printf("Playback from %i to %i\n", voice->dsp.start, voice->dsp.end);
printf("Playback loop from %i to %i\n",voice->dsp.loopstart, voice->dsp.loopend);
#endif
/* Keep the start point within the sample data */
if (voice->dsp.start < min_index_nonloop){
voice->dsp.start = min_index_nonloop;
} else if (voice->dsp.start > max_index_nonloop){
voice->dsp.start = max_index_nonloop;
}
/* Keep the end point within the sample data */
if (voice->dsp.end < min_index_nonloop){
voice->dsp.end = min_index_nonloop;
} else if (voice->dsp.end > max_index_nonloop){
voice->dsp.end = max_index_nonloop;
}
/* Keep start and end point in the right order */
if (voice->dsp.start > voice->dsp.end){
int temp = voice->dsp.start;
voice->dsp.start = voice->dsp.end;
voice->dsp.end = temp;
/*FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Changing order of start / end points!"); */
}
/* Zero length? */
if (voice->dsp.start == voice->dsp.end){
fluid_rvoice_voiceoff(voice);
return;
}
if ((voice->dsp.samplemode == FLUID_LOOP_UNTIL_RELEASE)
|| (voice->dsp.samplemode == FLUID_LOOP_DURING_RELEASE)) {
/* Keep the loop start point within the sample data */
if (voice->dsp.loopstart < min_index_loop){
voice->dsp.loopstart = min_index_loop;
} else if (voice->dsp.loopstart > max_index_loop){
voice->dsp.loopstart = max_index_loop;
}
/* Keep the loop end point within the sample data */
if (voice->dsp.loopend < min_index_loop){
voice->dsp.loopend = min_index_loop;
} else if (voice->dsp.loopend > max_index_loop){
voice->dsp.loopend = max_index_loop;
}
/* Keep loop start and end point in the right order */
if (voice->dsp.loopstart > voice->dsp.loopend){
int temp = voice->dsp.loopstart;
voice->dsp.loopstart = voice->dsp.loopend;
voice->dsp.loopend = temp;
/*FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Changing order of loop points!"); */
}
/* Loop too short? Then don't loop. */
if (voice->dsp.loopend < voice->dsp.loopstart + FLUID_MIN_LOOP_SIZE){
voice->dsp.samplemode = FLUID_UNLOOPED;
}
/* The loop points may have changed. Obtain a new estimate for the loop volume. */
/* Is the voice loop within the sample loop? */
if ((int)voice->dsp.loopstart >= (int)voice->dsp.sample->loopstart
&& (int)voice->dsp.loopend <= (int)voice->dsp.sample->loopend){
/* Is there a valid peak amplitude available for the loop? */
if (voice->dsp.sample->amplitude_that_reaches_noise_floor_is_valid){
voice->dsp.amplitude_that_reaches_noise_floor_loop=voice->dsp.sample->amplitude_that_reaches_noise_floor / voice->dsp.synth_gain;
} else {
/* Worst case */
voice->dsp.amplitude_that_reaches_noise_floor_loop=voice->dsp.amplitude_that_reaches_noise_floor_nonloop;
};
};
} /* if sample mode is looped */
/* Run startup specific code (only once, when the voice is started) */
if (voice->dsp.check_sample_sanity_flag & FLUID_SAMPLESANITY_STARTUP){
if (max_index_loop - min_index_loop < FLUID_MIN_LOOP_SIZE){
if ((voice->dsp.samplemode == FLUID_LOOP_UNTIL_RELEASE)
|| (voice->dsp.samplemode == FLUID_LOOP_DURING_RELEASE)){
voice->dsp.samplemode = FLUID_UNLOOPED;
}
}
/* Set the initial phase of the voice (using the result from the
start offset modulators). */
fluid_phase_set_int(voice->dsp.phase, voice->dsp.start);
} /* if startup */
/* Is this voice run in loop mode, or does it run straight to the
end of the waveform data? */
if (((voice->dsp.samplemode == FLUID_LOOP_UNTIL_RELEASE) &&
(fluid_adsr_env_get_section(&voice->envlfo.volenv) < FLUID_VOICE_ENVRELEASE))
|| (voice->dsp.samplemode == FLUID_LOOP_DURING_RELEASE)) {
/* Yes, it will loop as soon as it reaches the loop point. In
* this case we must prevent, that the playback pointer (phase)
* happens to end up beyond the 2nd loop point, because the
* point has moved. The DSP algorithm is unable to cope with
* that situation. So if the phase is beyond the 2nd loop
* point, set it to the start of the loop. No way to avoid some
* noise here. Note: If the sample pointer ends up -before the
* first loop point- instead, then the DSP loop will just play
* the sample, enter the loop and proceed as expected => no
* actions required.
*/
int index_in_sample = fluid_phase_index(voice->dsp.phase);
if (index_in_sample >= voice->dsp.loopend){
/* FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Phase after 2nd loop point!"); */
fluid_phase_set_int(voice->dsp.phase, voice->dsp.loopstart);
}
}
/* FLUID_LOG(FLUID_DBG, "Loop / sample sanity check: Sample from %i to %i, loop from %i to %i", voice->dsp.start, voice->dsp.end, voice->dsp.loopstart, voice->dsp.loopend); */
/* Sample sanity has been assured. Don't check again, until some
sample parameter is changed by modulation. */
voice->dsp.check_sample_sanity_flag=0;
#if 0
printf("Sane? playback loop from %i to %i\n", voice->dsp.loopstart, voice->dsp.loopend);
#endif
fluid_check_fpe("voice_check_sample_sanity");
}
/*
* Variable description:
* - dsp_a1, dsp_a2, dsp_b0, dsp_b1, dsp_b2: Filter coefficients
*
* A couple of variables are used internally, their results are discarded:
* - dsp_i: Index through the output buffer
* - dsp_phase_fractional: The fractional part of dsp_phase
* - dsp_coeff: A table of four coefficients, depending on the fractional phase.
* Used to interpolate between samples.
* - dsp_process_buffer: Holds the processed signal between stages
* - 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)
{
/* 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 dsp_filter_coeff_incr_count = iir_filter->filter_coeff_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 (fabs (dsp_hist1) < 1e-20) 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.
*/
if (dsp_filter_coeff_incr_count > 0)
{
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 && fabs(dsp_b02) > 0.001) {
fluid_real_t compensate = old_b02 / dsp_b02;
dsp_centernode *= compensate;
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;
}
}
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->filter_coeff_incr_count = dsp_filter_coeff_incr_count;
fluid_check_fpe ("voice_filter");
}
static FLUID_INLINE void
fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t* iir_filter,
int transition_samples,
fluid_real_t output_rate)
{
/*
* 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 / ((float) output_rate)));
fluid_real_t sin_coeff = (fluid_real_t) sin(omega);
fluid_real_t cos_coeff = (fluid_real_t) cos(omega);
fluid_real_t alpha_coeff = sin_coeff / (2.0f * iir_filter->q_lin);
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; */
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 b1_temp = (1.0f - cos_coeff) * a0_inv * iir_filter->filter_gain;
/* both b0 -and- b2 */
fluid_real_t b02_temp = b1_temp * 0.5f;
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 (fabs(iir_filter->b02) > 0.0001) {
fluid_real_t quota = b02_temp / iir_filter->b02;
iir_filter->compensate_incr = quota < 0.5 || quota > 2;
}
/* Have to add the increments filter_coeff_incr_count times. */
iir_filter->filter_coeff_incr_count = transition_samples;
}
fluid_check_fpe ("voice_write filter calculation");
}
