-CLIP)) { return sc_in; } else if (sc_in > 0.0f) { return MAX_AMP - (CLIP_A / (CLIP_B + sc_in)); } else { return -(MAX_AMP - (CLIP_A / (CLIP_B - sc_in))); } } /* Store data in SVF struct, takes the sampling frequency, cutoff frequency and Q, and fills in the structure passed */ inline void setup_svf(sv_filter *sv, float fs, float fc, float q, int t) { sv->f = 2.0f * sinf(M_PI * fc / (float)(fs * F_R)); sv->q = 2.0f * cosf(powf(q, 0.1f) * M_PI * 0.5f); sv->qnrm = sqrtf(sv->q*0.5f + 0.01f); switch(t) { case F_LP: sv->op = &(sv->l); break; case F_HP: sv->op = &(sv->h); break; case F_BP: sv->op = &(sv->b); break; case F_BR: sv->op = &(sv->n); break; default: sv->op = &(sv->p); } } /* Change the frequency of a running SVF */ inline void setup_f_svf(sv_filter *sv, const float fs, const float fc) { sv->f = 2.0f * sin(M_PI * fc / ((float)(fs * F_R))); } /* Run one sample through the SV filter. Filter is by andy@vellocet */ inline float run_svf(sv_filter *sv, float in) { float out; int i; in = sv->qnrm * in ; for (i=0; i < F_R; i++) { // only needed for pentium chips in = flush_to_zero(in); sv->l = flush_to_zero(sv->l); // very slight waveshape for extra stability sv->b = sv->b - sv->b * sv->b * sv->b * 0.001f; // regular state variable code here // the notch and peaking outputs are optional sv->h = in - sv->l - sv->q * sv->b; sv->b = sv->b + sv->f * sv->h; sv->l = sv->l + sv->f * sv->b; sv->n = sv->l + sv->h; sv->p = sv->l - sv->h; out = *(sv->op); in = out; } return out; } inline int wave_tbl(const float wave) { switch (f_round(wave)) { case 0: return BLO_SINE; break; case 1: return BLO_TRI; break; case 2: return BLO_SAW; break; case 3: return BLO_SQUARE; break; } return NOISE; } ]]> Hermes Filter

This plugin is a simulation of a modern analogue synth called a Pro Tone, with some extra features bolted on, like a crossover. I tried to make it as comprehensive as possible, without requiring ludicrous amounts of CPU juice.

N.B. as far as I know, noone has tried to use this (I certainly haven't), so it may be full of bugs and what not. The parameters are all undocumented, but there is a diagram of the routing on the website. Without a custom interface however it would be very hard to use.

Historical note: the name is a bad pun, it comes from the name Hermes Trimegistus given to the Egyptian god Thoth by the greeks, it means Thrice Blessed, or something similar.

long i; sample_rate = s_rate; count = 0; tables = blo_h_tables_new(1024); osc1_d = blo_h_new(tables, BLO_SINE, (float)s_rate); osc2_d = blo_h_new(tables, BLO_SINE, (float)s_rate); lfo1_d = blo_h_new(tables, BLO_SINE, (float)s_rate); lfo2_d = blo_h_new(tables, BLO_SINE, (float)s_rate); xover_b1_data = calloc(1, sizeof(sv_filter)); xover_b2_data = calloc(1, sizeof(sv_filter)); dela_data = malloc(3 * sizeof(float)); dela_pos = malloc(3 * sizeof(int)); filt_data = malloc(3 * sizeof(sv_filter *)); for (i = 0; i < 3; i++) { dela_data[i] = malloc(sample_rate * 2 * sizeof(float)); dela_pos[i] = 0; filt_data[i] = calloc(1, sizeof(sv_filter)); } lfo1 = 0.0f; lfo2 = 0.0f; lfo1_phase = 0.0f; lfo2_phase = 0.0f; setup_svf(filt_data[0], 0, 0, 0, 0); setup_svf(filt_data[1], 0, 0, 0, 0); setup_svf(filt_data[2], 0, 0, 0, 0); setup_svf(xover_b1_data, sample_rate, 1000.0, 0.0, F_HP); setup_svf(xover_b2_data, sample_rate, 100.0, 0.0, F_LP); memset(dela_data[0], 0, sample_rate * 2 * sizeof(float)); memset(dela_data[1], 0, sample_rate * 2 * sizeof(float)); memset(dela_data[2], 0, sample_rate * 2 * sizeof(float)); dela_pos[0] = 0; dela_pos[1] = 0; dela_pos[2] = 0; /* osc1_d->ph.all = 0; osc2_d->ph.all = 0; lfo1_d->ph.all = 0; lfo2_d->ph.all = 0; */ count = 0; lfo1 = 0.0f; lfo2 = 0.0f; lfo1_phase = 0.0f; lfo2_phase = 0.0f; free(plugin_data->filt_data[0]); free(plugin_data->filt_data[1]); free(plugin_data->filt_data[2]); free(plugin_data->dela_data[0]); free(plugin_data->dela_data[1]); free(plugin_data->dela_data[2]); free(plugin_data->filt_data); free(plugin_data->dela_data); free(plugin_data->dela_pos); free(plugin_data->xover_b1_data); free(plugin_data->xover_b2_data); blo_h_free(plugin_data->osc1_d); blo_h_free(plugin_data->osc2_d); blo_h_free(plugin_data->lfo1_d); blo_h_free(plugin_data->lfo2_d); blo_h_tables_free(plugin_data->tables); sample_rate * 2 || dela_offset[i] < 0) { dela_offset[i] = 0; } dela[i] = 0.0f; filt_t[i] = 0; } // Convert dB gains to coefficients osc1_gain = DB_CO(osc1_gain_db); osc2_gain = DB_CO(osc2_gain_db); in_gain = DB_CO(in_gain_db); rm1_gain = DB_CO(rm1_gain_db); rm2_gain = DB_CO(rm2_gain_db); rm3_gain = DB_CO(rm3_gain_db); band_gain[0] = DB_CO(band1_gain_db); band_gain[1] = DB_CO(band2_gain_db); band_gain[2] = DB_CO(band3_gain_db); osc1_d->wave = wave_tbl(osc1_wave); osc2_d->wave = wave_tbl(osc2_wave); lfo1_d->wave = wave_tbl(lfo1_wave); lfo2_d->wave = wave_tbl(lfo2_wave); blo_hd_set_freq(osc1_d, osc1_freq); blo_hd_set_freq(osc2_d, osc2_freq); blo_hd_set_freq(lfo1_d, lfo1_freq * 16); blo_hd_set_freq(lfo2_d, lfo2_freq * 16); #define SETUP_F(n,f,q,t) setup_svf(filt_data[n], sample_rate, f, q, (int)t) // Set filter stuff SETUP_F(0, filt1_freq, filt1_q, filt1_type); SETUP_F(1, filt2_freq, filt2_q, filt2_type); SETUP_F(2, filt3_freq, filt3_q, filt3_type); filt_freq[0] = filt1_freq; filt_freq[1] = filt2_freq; filt_freq[2] = filt3_freq; filt_res[0] = filt1_res; filt_res[1] = filt2_res; filt_res[2] = filt3_res; filt_lfo1[0] = filt1_lfo1; filt_lfo1[1] = filt2_lfo1; filt_lfo1[2] = filt3_lfo1; filt_lfo2[0] = filt1_lfo2; filt_lfo2[1] = filt2_lfo2; filt_lfo2[2] = filt3_lfo2; // Setup distortions drive[0] = drive1; drive[1] = drive2; drive[2] = drive3; // Setup delays dela_wet[0] = dela1_wet; dela_wet[1] = dela2_wet; dela_wet[2] = dela3_wet; dela_fb[0] = dela1_fb; dela_fb[1] = dela2_fb; dela_fb[2] = dela3_fb; tables = tables; // To shut up gcc for (pos = 0; pos < sample_count; pos++) { count++; // Count of number of samples processed // Calculate oscilator values for this sample if (osc1_d->wave == NOISE) { osc1 = rand() * (0.5f/(float)RAND_MAX) - 1.0f; } else { osc1 = blo_hd_run_lin(osc1_d); } if (osc2_d->wave == NOISE) { osc2 = rand() * (0.5f/(float)RAND_MAX) - 1.0f; } else { osc2 = blo_hd_run_lin(osc2_d); } // Calculate LFO values every 16 samples if ((count & 15) == 1) { // Calculate lfo values if (lfo1_d->wave == NOISE) { lfo1_phase += lfo1_freq; if (lfo1_phase >= sample_rate) { lfo1_phase -= sample_rate; lfo1 = rand() * (0.5f/(float)RAND_MAX) - 1.0f; } } else { lfo1 = blo_hd_run_lin(lfo1_d); } if (lfo2_d->wave == NOISE) { lfo2_phase += lfo1_freq; if (lfo2_phase >= sample_rate) { lfo2_phase -= sample_rate; lfo2 = rand() * (0.5f/(float)RAND_MAX) - 1.0f; } } else { lfo2 = blo_hd_run_lin(lfo2_d); } } in = input[pos]; rm1 = RINGMOD(osc2, osc1, rm1_depth); rm2 = RINGMOD(in, osc2, rm2_depth); rm3 = RINGMOD(osc1, in, rm3_depth); mixer1 = (osc1 * osc1_gain) + (osc2 * osc2_gain) + (in * in_gain) + (rm1 * rm1_gain) + (rm2 * rm2_gain) + (rm3 * rm3_gain); mixer1 = soft_clip(mixer1); // Higpass off the top band xover[0] = run_svf(xover_b1_data, mixer1); // Lowpass off the bottom band xover[2] = run_svf(xover_b2_data, mixer1); // The middle band is whats left xover[1] = mixer1 - xover[0] - xover[2]; mixer2 = 0.0f; for (i = 0; i < 3; i++) { dist[i] = xover[i]*(fabs(xover[i]) + drive1)/(xover[i]*xover[i] + (drive[i]-1)*fabs(xover[i]) + 1.0f); if (filt_t[i] == 0) { filt[i] = dist[i]; } else { if (count % 16 == 1) { setup_f_svf(filt_data[i], sample_rate, filt_freq[i]+LFO(filt_lfo1[i], filt_lfo2[i])); } filt[i] = run_svf(filt_data[i], dist[i] + (filt_res[i] * (filt_data[i])->b)); } dela[i] = (dela_data[i][dela_pos[i]] * dela_wet[i]) + filt[i]; dela_data[i][(dela_pos[i] + dela_offset[i]) % (2 * sample_rate)] = filt[i] + (dela[i] * dela_fb[i]); dela_pos[i] = (dela_pos[i] + 1) % (2 * sample_rate); mixer2 += band_gain[i] * dela[i]; } buffer_write(output[pos], soft_clip(mixer2)); } plugin_data->count = count; plugin_data->lfo1 = lfo1; plugin_data->lfo2 = lfo2; plugin_data->lfo1_phase = lfo1_phase; plugin_data->lfo2_phase = lfo2_phase; ]]> LFO1 freq (Hz) LFO1 wave (0 = sin, 1 = tri, 2 = saw, 3 = squ, 4 = s&h) LFO2 freq (Hz) LFO2 wave (0 = sin, 1 = tri, 2 = saw, 3 = squ, 4 = s&h) Osc1 freq (Hz) Osc1 wave (0 = sin, 1 = tri, 2 = saw, 3 = squ, 4 = noise) Osc2 freq (Hz) Osc2 wave (0 = sin, 1 = tri, 2 = saw, 3 = squ, 4 = noise) Ringmod 1 depth (0=none, 1=AM, 2=RM) Ringmod 2 depth (0=none, 1=AM, 2=RM) Ringmod 3 depth (0=none, 1=AM, 2=RM) Osc1 gain (dB) RM1 gain (dB) Osc2 gain (dB) RM2 gain (dB) Input gain (dB) RM3 gain (dB) Xover lower freq Xover upper freq Dist1 drive Dist2 drive Dist3 drive Filt1 type (0=none, 1=LP, 2=HP, 3=BP, 4=BR, 5=AP) Filt1 freq Filt1 q Filt1 resonance Filt1 LFO1 level Filt1 LFO2 level Filt2 type (0=none, 1=LP, 2=HP, 3=BP, 4=BR, 5=AP) Filt2 freq Filt2 q Filt2 resonance Filt2 LFO1 level Filt2 LFO2 level Filt3 type (0=none, 1=LP, 2=HP, 3=BP, 4=BR, 5=AP) Filt3 freq Filt3 q Filt3 resonance Filt3 LFO1 level Filt3 LFO2 level Delay1 length (s) Delay1 feedback Delay1 wetness Delay2 length (s) Delay2 feedback Delay2 wetness Delay3 length (s) Delay3 feedback Delay3 wetness Band 1 gain (dB) Band 2 gain (dB) Band 3 gain (dB) Input Output