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Refactor resonant filter into separate file
This commit is contained in:
David Henningsson
parent
6c0ef45e0e
commit
cb143c8fdf
5 changed files with 369 additions and 255 deletions
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@ -126,6 +126,7 @@ libfluidsynth_la_SOURCES = \
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fluid_tuning.h \
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fluid_voice.c \
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fluid_voice.h \
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fluid_iir_filter.c \
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fluid_filerenderer.c \
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fluid_aufile.c
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284
fluidsynth/src/fluid_iir_filter.c
Normal file
284
fluidsynth/src/fluid_iir_filter.c
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@ -0,0 +1,284 @@
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/* FluidSynth - A Software Synthesizer
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*
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* Copyright (C) 2003 Peter Hanappe and others.
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public License
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* as published by the Free Software Foundation; either version 2 of
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* the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the Free
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA
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* 02111-1307, USA
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*/
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#include "fluid_iir_filter.h"
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#include "fluid_sys.h"
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#include "fluid_conv.h"
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/**
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* Applies a lowpass filter with variable cutoff frequency and quality factor.
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* Also modifies filter state accordingly.
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* @param iir_filter Filter parameter
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* @param dsp_buf Pointer to the synthesized audio data
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* @param count Count of samples in dsp_buf
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*/
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/*
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* Variable description:
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* - dsp_a1, dsp_a2, dsp_b0, dsp_b1, dsp_b2: Filter coefficients
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*
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* A couple of variables are used internally, their results are discarded:
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* - dsp_i: Index through the output buffer
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* - dsp_phase_fractional: The fractional part of dsp_phase
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* - dsp_coeff: A table of four coefficients, depending on the fractional phase.
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* Used to interpolate between samples.
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* - dsp_process_buffer: Holds the processed signal between stages
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* - dsp_centernode: delay line for the IIR filter
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* - dsp_hist1: same
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* - dsp_hist2: same
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*/
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void
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fluid_iir_filter_apply(fluid_iir_filter_t* iir_filter,
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fluid_real_t *dsp_buf, int count)
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{
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/* IIR filter sample history */
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fluid_real_t dsp_hist1 = iir_filter->hist1;
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fluid_real_t dsp_hist2 = iir_filter->hist2;
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/* IIR filter coefficients */
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fluid_real_t dsp_a1 = iir_filter->a1;
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fluid_real_t dsp_a2 = iir_filter->a2;
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fluid_real_t dsp_b02 = iir_filter->b02;
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fluid_real_t dsp_b1 = iir_filter->b1;
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fluid_real_t dsp_a1_incr = iir_filter->a1_incr;
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fluid_real_t dsp_a2_incr = iir_filter->a2_incr;
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fluid_real_t dsp_b02_incr = iir_filter->b02_incr;
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fluid_real_t dsp_b1_incr = iir_filter->b1_incr;
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int dsp_filter_coeff_incr_count = iir_filter->filter_coeff_incr_count;
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fluid_real_t dsp_centernode;
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int dsp_i;
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/* filter (implement the voice filter according to SoundFont standard) */
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/* Check for denormal number (too close to zero). */
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if (fabs (dsp_hist1) < 1e-20) dsp_hist1 = 0.0f; /* FIXME JMG - Is this even needed? */
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/* Two versions of the filter loop. One, while the filter is
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* changing towards its new setting. The other, if the filter
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* doesn't change.
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*/
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if (dsp_filter_coeff_incr_count > 0)
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{
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/* Increment is added to each filter coefficient filter_coeff_incr_count times. */
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for (dsp_i = 0; dsp_i < count; dsp_i++)
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{
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/* The filter is implemented in Direct-II form. */
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dsp_centernode = dsp_buf[dsp_i] - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2;
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dsp_buf[dsp_i] = dsp_b02 * (dsp_centernode + dsp_hist2) + dsp_b1 * dsp_hist1;
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dsp_hist2 = dsp_hist1;
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dsp_hist1 = dsp_centernode;
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if (dsp_filter_coeff_incr_count-- > 0)
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{
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dsp_a1 += dsp_a1_incr;
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dsp_a2 += dsp_a2_incr;
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dsp_b02 += dsp_b02_incr;
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dsp_b1 += dsp_b1_incr;
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}
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} /* for dsp_i */
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}
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else /* The filter parameters are constant. This is duplicated to save time. */
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{
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for (dsp_i = 0; dsp_i < count; dsp_i++)
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{ /* The filter is implemented in Direct-II form. */
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dsp_centernode = dsp_buf[dsp_i] - dsp_a1 * dsp_hist1 - dsp_a2 * dsp_hist2;
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dsp_buf[dsp_i] = dsp_b02 * (dsp_centernode + dsp_hist2) + dsp_b1 * dsp_hist1;
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dsp_hist2 = dsp_hist1;
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dsp_hist1 = dsp_centernode;
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}
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}
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iir_filter->hist1 = dsp_hist1;
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iir_filter->hist2 = dsp_hist2;
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iir_filter->a1 = dsp_a1;
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iir_filter->a2 = dsp_a2;
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iir_filter->b02 = dsp_b02;
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iir_filter->b1 = dsp_b1;
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iir_filter->filter_coeff_incr_count = dsp_filter_coeff_incr_count;
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fluid_check_fpe ("voice_filter");
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}
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void
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fluid_iir_filter_reset(fluid_iir_filter_t* iir_filter)
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{
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iir_filter->hist1 = 0;
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iir_filter->hist2 = 0;
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iir_filter->last_fres = -1.;
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iir_filter->filter_startup = 1;
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}
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void
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fluid_iir_filter_set_fres(fluid_iir_filter_t* iir_filter,
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fluid_real_t fres)
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{
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iir_filter->fres = fres;
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iir_filter->last_fres = -1.;
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}
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void
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fluid_iir_filter_set_q_dB(fluid_iir_filter_t* iir_filter,
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fluid_real_t q_dB)
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{
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/* The 'sound font' Q is defined in dB. The filter needs a linear
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q. Convert. */
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iir_filter->q_lin = (fluid_real_t) (pow(10.0f, q_dB / 20.0f));
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/* SF 2.01 page 59:
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*
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* The SoundFont specs ask for a gain reduction equal to half the
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* height of the resonance peak (Q). For example, for a 10 dB
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* resonance peak, the gain is reduced by 5 dB. This is done by
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* multiplying the total gain with sqrt(1/Q). `Sqrt' divides dB
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* by 2 (100 lin = 40 dB, 10 lin = 20 dB, 3.16 lin = 10 dB etc)
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* The gain is later factored into the 'b' coefficients
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* (numerator of the filter equation). This gain factor depends
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* only on Q, so this is the right place to calculate it.
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*/
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iir_filter->filter_gain = (fluid_real_t) (1.0 / sqrt(iir_filter->q_lin));
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/* The synthesis loop will have to recalculate the filter coefficients. */
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iir_filter->last_fres = -1.;
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}
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static inline void
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fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t* iir_filter,
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int transition_samples,
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fluid_real_t output_rate)
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{
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/*
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* Those equations from Robert Bristow-Johnson's `Cookbook
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* formulae for audio EQ biquad filter coefficients', obtained
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* from Harmony-central.com / Computer / Programming. They are
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* the result of the bilinear transform on an analogue filter
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* prototype. To quote, `BLT frequency warping has been taken
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* into account for both significant frequency relocation and for
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* bandwidth readjustment'. */
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fluid_real_t omega = (fluid_real_t) (2.0 * M_PI *
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(iir_filter->last_fres / ((float) output_rate)));
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fluid_real_t sin_coeff = (fluid_real_t) sin(omega);
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fluid_real_t cos_coeff = (fluid_real_t) cos(omega);
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fluid_real_t alpha_coeff = sin_coeff / (2.0f * iir_filter->q_lin);
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fluid_real_t a0_inv = 1.0f / (1.0f + alpha_coeff);
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/* Calculate the filter coefficients. All coefficients are
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* normalized by a0. Think of `a1' as `a1/a0'.
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*
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* Here a couple of multiplications are saved by reusing common expressions.
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* The original equations should be:
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* iir_filter->b0=(1.-cos_coeff)*a0_inv*0.5*iir_filter->filter_gain;
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* iir_filter->b1=(1.-cos_coeff)*a0_inv*iir_filter->filter_gain;
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* iir_filter->b2=(1.-cos_coeff)*a0_inv*0.5*iir_filter->filter_gain; */
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fluid_real_t a1_temp = -2.0f * cos_coeff * a0_inv;
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fluid_real_t a2_temp = (1.0f - alpha_coeff) * a0_inv;
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fluid_real_t b1_temp = (1.0f - cos_coeff) * a0_inv * iir_filter->filter_gain;
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/* both b0 -and- b2 */
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fluid_real_t b02_temp = b1_temp * 0.5f;
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if (iir_filter->filter_startup || (transition_samples == 0))
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{
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/* The filter is calculated, because the voice was started up.
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* In this case set the filter coefficients without delay.
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*/
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iir_filter->a1 = a1_temp;
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iir_filter->a2 = a2_temp;
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iir_filter->b02 = b02_temp;
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iir_filter->b1 = b1_temp;
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iir_filter->filter_coeff_incr_count = 0;
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iir_filter->filter_startup = 0;
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// printf("Setting initial filter coefficients.\n");
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}
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else
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{
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/* The filter frequency is changed. Calculate an increment
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* factor, so that the new setting is reached after one buffer
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* length. x_incr is added to the current value FLUID_BUFSIZE
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* times. The length is arbitrarily chosen. Longer than one
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* buffer will sacrifice some performance, though. Note: If
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* the filter is still too 'grainy', then increase this number
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* at will.
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*/
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iir_filter->a1_incr = (a1_temp - iir_filter->a1) / transition_samples;
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iir_filter->a2_incr = (a2_temp - iir_filter->a2) / transition_samples;
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iir_filter->b02_incr = (b02_temp - iir_filter->b02) / transition_samples;
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iir_filter->b1_incr = (b1_temp - iir_filter->b1) / transition_samples;
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/* Have to add the increments filter_coeff_incr_count times. */
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iir_filter->filter_coeff_incr_count = transition_samples;
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}
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fluid_check_fpe ("voice_write filter calculation");
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}
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void fluid_iir_filter_calc(fluid_iir_filter_t* iir_filter,
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fluid_real_t output_rate,
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fluid_real_t fres_mod)
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{
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fluid_real_t fres;
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/* calculate the frequency of the resonant filter in Hz */
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fres = fluid_ct2hz(iir_filter->fres + fres_mod);
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/* FIXME - Still potential for a click during turn on, can we interpolate
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between 20khz cutoff and 0 Q? */
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/* I removed the optimization of turning the filter off when the
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* resonance frequence is above the maximum frequency. Instead, the
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* filter frequency is set to a maximum of 0.45 times the sampling
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* rate. For a 44100 kHz sampling rate, this amounts to 19845
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* Hz. The reason is that there were problems with anti-aliasing when the
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* synthesizer was run at lower sampling rates. Thanks to Stephan
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* Tassart for pointing me to this bug. By turning the filter on and
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* clipping the maximum filter frequency at 0.45*srate, the filter
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* is used as an anti-aliasing filter. */
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if (fres > 0.45f * output_rate)
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fres = 0.45f * output_rate;
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else if (fres < 5)
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fres = 5;
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/* if filter enabled and there is a significant frequency change.. */
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if ((abs (fres - iir_filter->last_fres) > 0.01))
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{
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/* The filter coefficients have to be recalculated (filter
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* parameters have changed). Recalculation for various reasons is
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* forced by setting last_fres to -1. The flag filter_startup
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* indicates, that the DSP loop runs for the first time, in this
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* case, the filter is set directly, instead of smoothly fading
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* between old and new settings. */
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iir_filter->last_fres = fres;
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fluid_iir_filter_calculate_coefficients(iir_filter, FLUID_BUFSIZE,
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output_rate);
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}
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fluid_check_fpe ("voice_write DSP coefficients");
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}
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74
fluidsynth/src/fluid_iir_filter.h
Normal file
74
fluidsynth/src/fluid_iir_filter.h
Normal file
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@ -0,0 +1,74 @@
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/* FluidSynth - A Software Synthesizer
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*
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* Copyright (C) 2003 Peter Hanappe and others.
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public License
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* as published by the Free Software Foundation; either version 2 of
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* the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the Free
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA
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* 02111-1307, USA
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*/
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#ifndef _FLUID_IIR_FILTER_H
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#define _FLUID_IIR_FILTER_H
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#include "fluidsynth_priv.h"
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typedef struct _fluid_iir_filter_t fluid_iir_filter_t;
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void fluid_iir_filter_apply(fluid_iir_filter_t* iir_filter,
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fluid_real_t *dsp_buf, int dsp_buf_count);
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void fluid_iir_filter_reset(fluid_iir_filter_t* iir_filter);
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void fluid_iir_filter_set_q_dB(fluid_iir_filter_t* iir_filter,
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fluid_real_t q_dB);
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void fluid_iir_filter_set_fres(fluid_iir_filter_t* iir_filter,
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fluid_real_t fres);
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void fluid_iir_filter_calc(fluid_iir_filter_t* iir_filter,
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fluid_real_t output_rate,
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fluid_real_t fres_mod);
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/* We can't do information hiding here, as fluid_voice_t includes the struct
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without a pointer. */
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struct _fluid_iir_filter_t
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{
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/* filter coefficients */
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/* The coefficients are normalized to a0. */
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/* b0 and b2 are identical => b02 */
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fluid_real_t b02; /* b0 / a0 */
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fluid_real_t b1; /* b1 / a0 */
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fluid_real_t a1; /* a0 / a0 */
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fluid_real_t a2; /* a1 / a0 */
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fluid_real_t b02_incr;
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fluid_real_t b1_incr;
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fluid_real_t a1_incr;
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fluid_real_t a2_incr;
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int filter_coeff_incr_count;
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fluid_real_t hist1, hist2; /* Sample history for the IIR filter */
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int filter_startup; /* Flag: If set, the filter will be set directly.
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Else it changes smoothly. */
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fluid_real_t fres; /* the resonance frequency, in cents (not absolute cents) */
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fluid_real_t last_fres; /* Current resonance frequency of the IIR filter */
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/* Serves as a flag: A deviation between fres and last_fres */
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/* indicates, that the filter has to be recalculated. */
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fluid_real_t q_lin; /* the q-factor on a linear scale */
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fluid_real_t filter_gain; /* Gain correction factor, depends on q */
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};
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#endif
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@ -48,7 +48,6 @@
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static int fluid_voice_calculate_runtime_synthesis_parameters(fluid_voice_t* voice);
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static int calculate_hold_decay_buffers(fluid_voice_t* voice, int gen_base,
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int gen_key2base, int is_decay);
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static inline void fluid_voice_filter (fluid_voice_t *voice);
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static fluid_real_t
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fluid_voice_get_lower_boundary_for_attenuation(fluid_voice_t* voice);
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static void fluid_voice_check_sample_sanity(fluid_voice_t* voice);
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@ -145,8 +144,6 @@ fluid_voice_init(fluid_voice_t* voice, fluid_sample_t* sample,
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voice->noteoff_ticks = 0;
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voice->debug = 0;
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voice->has_looped = 0; /* Will be set during voice_write when the 2nd loop point is reached */
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voice->last_fres = -1; /* The filter coefficients have to be calculated later in the DSP loop. */
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voice->filter_startup = 1; /* Set the filter immediately, don't fade between old and new settings */
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voice->interp_method = fluid_channel_get_interp_method(voice->channel);
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/* vol env initialization */
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@ -171,8 +168,7 @@ fluid_voice_init(fluid_voice_t* voice, fluid_sample_t* sample,
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voice->viblfo_val = 0.0f; /* Fixme: See mod lfo */
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/* Clear sample history in filter */
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voice->hist1 = 0;
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voice->hist2 = 0;
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fluid_iir_filter_reset(&voice->resonant_filter);
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/* Set all the generators to their default value, according to SF
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* 2.01 section 8.1.3 (page 48). The value of NRPN messages are
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|
|
@ -267,7 +263,6 @@ fluid_real_t fluid_voice_gen_value(fluid_voice_t* voice, int num)
|
|||
int
|
||||
fluid_voice_write (fluid_voice_t* voice, fluid_real_t *dsp_buf)
|
||||
{
|
||||
fluid_real_t fres;
|
||||
fluid_real_t target_amp; /* target amplitude */
|
||||
fluid_env_data_t* env_data;
|
||||
fluid_real_t x;
|
||||
|
|
@ -497,109 +492,9 @@ fluid_voice_write (fluid_voice_t* voice, fluid_real_t *dsp_buf)
|
|||
if (voice->phase_incr == 0) voice->phase_incr = 1;
|
||||
|
||||
/*************** resonant filter ******************/
|
||||
|
||||
/* calculate the frequency of the resonant filter in Hz */
|
||||
fres = fluid_ct2hz(voice->fres
|
||||
+ voice->modlfo_val * voice->modlfo_to_fc
|
||||
+ voice->modenv_val * voice->modenv_to_fc);
|
||||
|
||||
/* 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 * voice->output_rate)
|
||||
fres = 0.45f * voice->output_rate;
|
||||
else if (fres < 5)
|
||||
fres = 5;
|
||||
|
||||
/* if filter enabled and there is a significant frequency change.. */
|
||||
if ((abs (fres - voice->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.
|
||||
*
|
||||
* 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 * (fres / ((float) voice->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 * voice->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:
|
||||
* voice->b0=(1.-cos_coeff)*a0_inv*0.5*voice->filter_gain;
|
||||
* voice->b1=(1.-cos_coeff)*a0_inv*voice->filter_gain;
|
||||
* voice->b2=(1.-cos_coeff)*a0_inv*0.5*voice->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 * voice->filter_gain;
|
||||
/* both b0 -and- b2 */
|
||||
fluid_real_t b02_temp = b1_temp * 0.5f;
|
||||
|
||||
if (voice->filter_startup)
|
||||
{
|
||||
/* The filter is calculated, because the voice was started up.
|
||||
* In this case set the filter coefficients without delay.
|
||||
*/
|
||||
voice->a1 = a1_temp;
|
||||
voice->a2 = a2_temp;
|
||||
voice->b02 = b02_temp;
|
||||
voice->b1 = b1_temp;
|
||||
voice->filter_coeff_incr_count = 0;
|
||||
voice->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.
|
||||
*/
|
||||
|
||||
#define FILTER_TRANSITION_SAMPLES (FLUID_BUFSIZE)
|
||||
|
||||
voice->a1_incr = (a1_temp - voice->a1) / FILTER_TRANSITION_SAMPLES;
|
||||
voice->a2_incr = (a2_temp - voice->a2) / FILTER_TRANSITION_SAMPLES;
|
||||
voice->b02_incr = (b02_temp - voice->b02) / FILTER_TRANSITION_SAMPLES;
|
||||
voice->b1_incr = (b1_temp - voice->b1) / FILTER_TRANSITION_SAMPLES;
|
||||
/* Have to add the increments filter_coeff_incr_count times. */
|
||||
voice->filter_coeff_incr_count = FILTER_TRANSITION_SAMPLES;
|
||||
}
|
||||
voice->last_fres = fres;
|
||||
fluid_check_fpe ("voice_write filter calculation");
|
||||
}
|
||||
|
||||
|
||||
fluid_check_fpe ("voice_write DSP coefficients");
|
||||
fluid_iir_filter_calc(&voice->resonant_filter, voice->output_rate,
|
||||
voice->modlfo_val * voice->modlfo_to_fc +
|
||||
voice->modenv_val * voice->modenv_to_fc);
|
||||
|
||||
/*********************** run the dsp chain ************************
|
||||
* The sample is mixed with the output buffer.
|
||||
|
|
@ -629,7 +524,8 @@ fluid_voice_write (fluid_voice_t* voice, fluid_real_t *dsp_buf)
|
|||
voice->dsp_buf_count = count;
|
||||
|
||||
/* Apply filter */
|
||||
if (count > 0) fluid_voice_filter (voice);
|
||||
if (count > 0)
|
||||
fluid_iir_filter_apply(&voice->resonant_filter, voice->dsp_buf, count);
|
||||
|
||||
/* turn off voice if short count (sample ended and not looping) */
|
||||
if (count < FLUID_BUFSIZE)
|
||||
|
|
@ -645,103 +541,6 @@ fluid_voice_write (fluid_voice_t* voice, fluid_real_t *dsp_buf)
|
|||
}
|
||||
|
||||
|
||||
/**
|
||||
* Applies a lowpass filter with variable cutoff frequency and quality factor.
|
||||
* @param voice Voice to apply filter to
|
||||
* @param count Count of samples in voice->dsp_buf
|
||||
*/
|
||||
/*
|
||||
* Variable description:
|
||||
* - dsp_buf: Pointer to the synthesized audio data
|
||||
* - dsp_a1, dsp_a2, dsp_b0, dsp_b1, dsp_b2: Filter coefficients
|
||||
* - voice holds the voice structure
|
||||
*
|
||||
* 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
|
||||
*/
|
||||
static inline void
|
||||
fluid_voice_filter (fluid_voice_t *voice)
|
||||
{
|
||||
/* IIR filter sample history */
|
||||
fluid_real_t dsp_hist1 = voice->hist1;
|
||||
fluid_real_t dsp_hist2 = voice->hist2;
|
||||
|
||||
/* IIR filter coefficients */
|
||||
fluid_real_t dsp_a1 = voice->a1;
|
||||
fluid_real_t dsp_a2 = voice->a2;
|
||||
fluid_real_t dsp_b02 = voice->b02;
|
||||
fluid_real_t dsp_b1 = voice->b1;
|
||||
fluid_real_t dsp_a1_incr = voice->a1_incr;
|
||||
fluid_real_t dsp_a2_incr = voice->a2_incr;
|
||||
fluid_real_t dsp_b02_incr = voice->b02_incr;
|
||||
fluid_real_t dsp_b1_incr = voice->b1_incr;
|
||||
int dsp_filter_coeff_incr_count = voice->filter_coeff_incr_count;
|
||||
|
||||
fluid_real_t *dsp_buf = voice->dsp_buf;
|
||||
|
||||
fluid_real_t dsp_centernode;
|
||||
int count = voice->dsp_buf_count;
|
||||
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)
|
||||
{
|
||||
/* 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)
|
||||
{
|
||||
dsp_a1 += dsp_a1_incr;
|
||||
dsp_a2 += dsp_a2_incr;
|
||||
dsp_b02 += dsp_b02_incr;
|
||||
dsp_b1 += dsp_b1_incr;
|
||||
}
|
||||
} /* 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;
|
||||
}
|
||||
}
|
||||
|
||||
voice->hist1 = dsp_hist1;
|
||||
voice->hist2 = dsp_hist2;
|
||||
voice->a1 = dsp_a1;
|
||||
voice->a2 = dsp_a2;
|
||||
voice->b02 = dsp_b02;
|
||||
voice->b1 = dsp_b1;
|
||||
voice->filter_coeff_incr_count = dsp_filter_coeff_incr_count;
|
||||
|
||||
fluid_check_fpe ("voice_filter");
|
||||
}
|
||||
|
||||
/**
|
||||
* Mix voice data to left/right (panning), reverb and chorus buffers.
|
||||
* @param voice Voice to mix
|
||||
|
|
@ -1146,11 +945,7 @@ fluid_voice_update_param(fluid_voice_t* voice, int gen)
|
|||
* modulation. The allowed range is tested in the 'fluid_ct2hz'
|
||||
* function [PH,20021214]
|
||||
*/
|
||||
voice->fres = _GEN(voice, GEN_FILTERFC);
|
||||
|
||||
/* The synthesis loop will have to recalculate the filter
|
||||
* coefficients. */
|
||||
voice->last_fres = -1.0f;
|
||||
fluid_iir_filter_set_fres(&voice->resonant_filter, _GEN(voice, GEN_FILTERFC));
|
||||
break;
|
||||
|
||||
case GEN_FILTERQ:
|
||||
|
|
@ -1177,26 +972,8 @@ fluid_voice_update_param(fluid_voice_t* voice, int gen)
|
|||
* response of a non-resonant filter. This idea is implemented as
|
||||
* follows: */
|
||||
q_dB -= 3.01f;
|
||||
fluid_iir_filter_set_q_dB(&voice->resonant_filter, q_dB);
|
||||
|
||||
/* The 'sound font' Q is defined in dB. The filter needs a linear
|
||||
q. Convert. */
|
||||
voice->q_lin = (fluid_real_t) (pow(10.0f, q_dB / 20.0f));
|
||||
|
||||
/* 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.
|
||||
*/
|
||||
voice->filter_gain = (fluid_real_t) (1.0 / sqrt(voice->q_lin));
|
||||
|
||||
/* The synthesis loop will have to recalculate the filter coefficients. */
|
||||
voice->last_fres = -1.;
|
||||
break;
|
||||
|
||||
case GEN_MODLFOTOPITCH:
|
||||
|
|
|
|||
|
|
@ -25,6 +25,7 @@
|
|||
#include "fluid_phase.h"
|
||||
#include "fluid_gen.h"
|
||||
#include "fluid_mod.h"
|
||||
#include "fluid_iir_filter.h"
|
||||
|
||||
#define NO_CHANNEL 0xff
|
||||
|
||||
|
|
@ -168,30 +169,7 @@ struct _fluid_voice_t
|
|||
fluid_real_t viblfo_incr; /* the lfo frequency is converted to a per-buffer increment */
|
||||
fluid_real_t viblfo_to_pitch;
|
||||
|
||||
/* resonant filter */
|
||||
fluid_real_t fres; /* the resonance frequency, in cents (not absolute cents) */
|
||||
fluid_real_t last_fres; /* Current resonance frequency of the IIR filter */
|
||||
/* Serves as a flag: A deviation between fres and last_fres */
|
||||
/* indicates, that the filter has to be recalculated. */
|
||||
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 hist1, hist2; /* Sample history for the IIR filter */
|
||||
int filter_startup; /* Flag: If set, the filter will be set directly.
|
||||
Else it changes smoothly. */
|
||||
|
||||
/* filter coefficients */
|
||||
/* The coefficients are normalized to a0. */
|
||||
/* b0 and b2 are identical => b02 */
|
||||
fluid_real_t b02; /* b0 / a0 */
|
||||
fluid_real_t b1; /* b1 / a0 */
|
||||
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;
|
||||
fluid_iir_filter_t resonant_filter; /* IIR resonance dsp filter */
|
||||
|
||||
/* pan */
|
||||
fluid_real_t pan;
|
||||
|
|
|
|||
Loading…
Reference in a new issue