From 0949bdd0a12450ed0fd21539872fa25572af5e7e Mon Sep 17 00:00:00 2001 From: derselbst Date: Sun, 10 Nov 2024 17:06:34 +0100 Subject: [PATCH] Linear interpolation of Q --- src/rvoice/fluid_iir_filter.c | 139 +++++++++++++++++++++------------- src/rvoice/fluid_iir_filter.h | 12 +-- 2 files changed, 93 insertions(+), 58 deletions(-) diff --git a/src/rvoice/fluid_iir_filter.c b/src/rvoice/fluid_iir_filter.c index 6e55c3cb8..20c652deb 100644 --- a/src/rvoice/fluid_iir_filter.c +++ b/src/rvoice/fluid_iir_filter.c @@ -55,7 +55,7 @@ void fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter, fluid_real_t *dsp_buf, int count, fluid_real_t output_rate) { - if(iir_filter->type == FLUID_IIR_DISABLED || iir_filter->q_lin == 0) + if(iir_filter->type == FLUID_IIR_DISABLED || FLUID_FABS(iir_filter->last_q) <= 0.001) { return; } @@ -71,8 +71,8 @@ fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter, fluid_real_t dsp_b02 = iir_filter->b02; fluid_real_t dsp_b1 = iir_filter->b1; - fluid_real_t fres_incr = iir_filter->fres_incr; 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; @@ -103,12 +103,20 @@ fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter, // 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) + if(fres_incr_count > 0 || q_incr_count > 0) { - --fres_incr_count; - iir_filter->last_fres += fres_incr; + 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; + } - FLUID_LOG(FLUID_DBG, "last_fres: %f | target_fres: %f", iir_filter->last_fres, iir_filter->target_fres); + FLUID_LOG(FLUID_DBG, "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); @@ -128,6 +136,7 @@ fluid_iir_filter_apply(fluid_iir_filter_t *iir_filter, 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"); } @@ -154,7 +163,7 @@ 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->q_lin = 0; + iir_filter->last_q = 0; iir_filter->filter_startup = 1; } @@ -225,37 +234,35 @@ DECLARE_FLUID_RVOICE_FUNCTION(fluid_iir_filter_set_q) } FLUID_LOG(FLUID_DBG, "fluid_iir_filter_set_q: Q= %f [linear]",q); - iir_filter->q_lin = q; - iir_filter->filter_gain = 1.0; - - if(!(flags & FLUID_IIR_NO_GAIN_AMP)) + + if(iir_filter->filter_startup) { - /* 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_SQRT(q); + iir_filter->last_q = q; } + else + { + static const int q_incr_count = FLUID_BUFSIZE; + iir_filter->q_incr = (q - iir_filter->last_q) / (q_incr_count); + iir_filter->q_incr_count = q_incr_count; + } + iir_filter->target_q = q; } static FLUID_INLINE void fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t *iir_filter, fluid_real_t output_rate) { - /* FLUID_IIR_Q_LINEAR may switch the filter off by setting Q==0 */ - if(iir_filter->q_lin == 0) + // 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(FLUID_FABS(iir_filter->last_q) <= 0.001) { return; } else { + 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 @@ -269,7 +276,7 @@ fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t *iir_filter, (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->q_lin); + 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 @@ -277,20 +284,35 @@ fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t *iir_filter, * * 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; */ + * 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 * iir_filter->filter_gain; + b1_temp = (1.0f + cos_coeff) * a0_inv * filter_gain; /* both b0 -and- b2 */ b02_temp = b1_temp * 0.5f; @@ -299,7 +321,7 @@ fluid_iir_filter_calculate_coefficients(fluid_iir_filter_t *iir_filter, break; case FLUID_IIR_LOWPASS: - b1_temp = (1.0f - cos_coeff) * a0_inv * iir_filter->filter_gain; + b1_temp = (1.0f - cos_coeff) * a0_inv * filter_gain; /* both b0 -and- b2 */ b02_temp = b1_temp * 0.5f; @@ -324,7 +346,13 @@ void fluid_iir_filter_calc(fluid_iir_filter_t *iir_filter, fluid_real_t output_rate, fluid_real_t fres_mod) { + unsigned int calc_coeff_flag = FALSE; fluid_real_t fres, fres_diff; + + if(iir_filter->type == FLUID_IIR_DISABLED) + { + return; + } /* calculate the frequency of the resonant filter in Hz */ fres = fluid_ct2hz(iir_filter->fres + fres_mod); @@ -348,34 +376,39 @@ void fluid_iir_filter_calc(fluid_iir_filter_t *iir_filter, fres = 5.f; } - FLUID_LOG(FLUID_DBG, "%f + %f = %f cents = %f Hz | Q: %f", iir_filter->fres, fres_mod, iir_filter->fres + fres_mod, fres, iir_filter->q_lin); + FLUID_LOG(FLUID_DBG, "%f + %f = %f cents = %f Hz | Q: %f", iir_filter->fres, fres_mod, iir_filter->fres + fres_mod, fres, iir_filter->last_q); /* if filter enabled and there is a significant frequency change.. */ fres_diff = fres - iir_filter->last_fres; - if(iir_filter->type != FLUID_IIR_DISABLED && FLUID_FABS(fres_diff) > 0.01f) + if(iir_filter->filter_startup) { - /* 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. */ - if(iir_filter->filter_startup) - { - iir_filter->fres_incr_count = 0; - iir_filter->last_fres = fres; - iir_filter->filter_startup = 0; - } - else - { - static const int fres_incr_count = FLUID_BUFSIZE; - iir_filter->fres_incr = fres_diff / (fres_incr_count); - iir_filter->fres_incr_count = fres_incr_count; - } + // The filer was just starting up, make sure to calculate initial coefficients for the initial Q value, even though the fres may not have changed + calc_coeff_flag = TRUE; + + iir_filter->fres_incr_count = 0; + iir_filter->last_fres = fres; + iir_filter->filter_startup = 0; + } + else if(FLUID_FABS(fres_diff) > 0.01f) + { + static const int fres_incr_count = FLUID_BUFSIZE; + iir_filter->fres_incr = fres_diff / (fres_incr_count); + iir_filter->fres_incr_count = fres_incr_count; iir_filter->target_fres = fres; - fluid_iir_filter_calculate_coefficients(iir_filter, output_rate); + + // The filter coefficients have to be recalculated (filter cutoff has changed). + calc_coeff_flag = TRUE; + } + else + { + // We do not account for any change of Q here - if it was changed q_incro_count will be non-zero and recalculating the coeffs + // will be taken care of in fluid_iir_filter_apply(). } + if(calc_coeff_flag) + { + fluid_iir_filter_calculate_coefficients(iir_filter, output_rate); + } fluid_check_fpe("voice_write DSP coefficients"); diff --git a/src/rvoice/fluid_iir_filter.h b/src/rvoice/fluid_iir_filter.h index cf18d178b..c7f321f92 100644 --- a/src/rvoice/fluid_iir_filter.h +++ b/src/rvoice/fluid_iir_filter.h @@ -54,16 +54,18 @@ struct _fluid_iir_filter_t fluid_real_t a2; /* a1 / a0 */ 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. */ + int filter_startup; /* Flag: If set, the filter parameters will be set directly. Else it changes smoothly. */ fluid_real_t fres; /* The desired resonance frequency, in absolute cents, this filter is currently set to */ - fluid_real_t last_fres; /* The filter's currently (smoothed out) resonance frequency in Hz, which will converge towards its target fres once fres_incr_count has become zero */ + fluid_real_t last_fres; /* The filter's current (smoothed out) resonance frequency in Hz, which will converge towards its target fres once fres_incr_count has become zero */ fluid_real_t target_fres; /* The filter's target fres, that last_fres should converge towards - for debugging only */ - fluid_real_t fres_incr; /* The linear increment of fres on each sample */ + fluid_real_t fres_incr; /* The linear increment of fres each sample */ int fres_incr_count; /* The number of samples left for the smoothed last_fres adjustment to complete */ - 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 last_q; /* The filter's current (smoothed) Q-factor (or "bandwidth", or "resonance-friendlyness") on a linear scale. Just like fres, this will converge towards its target Q once q_incr_count has become zero. */ + fluid_real_t target_q; /* The filter's target Q - for debugging only */ + fluid_real_t q_incr; /* The linear increment of q each sample */ + int q_incr_count; /* The number of samples left for the smoothed Q adjustment to complete */ }; #endif