/* FluidSynth - A Software Synthesizer * * Copyright (C) 2003 Peter Hanappe and others. * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public License * as published by the Free Software Foundation; either version 2 of * the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the Free * Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA * 02110-1301, USA */ #include "fluid_iir_filter.h" #include "fluid_sys.h" #include "fluid_conv.h" /** * Applies a lowpass filter with variable cutoff frequency and quality factor. * 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, 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"); } void 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->filter_startup = 1; } void fluid_iir_filter_set_fres(fluid_iir_filter_t* iir_filter, fluid_real_t fres) { iir_filter->fres = fres; iir_filter->last_fres = -1.; } void fluid_iir_filter_set_q_dB(fluid_iir_filter_t* iir_filter, fluid_real_t q_dB) { /* The 'sound font' Q is defined in dB. The filter needs a linear q. Convert. */ iir_filter->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. */ iir_filter->filter_gain = (fluid_real_t) (1.0 / sqrt(iir_filter->q_lin)); /* The synthesis loop will have to recalculate the filter coefficients. */ iir_filter->last_fres = -1.; } static 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"); } 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"); }