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MiniDexed/src/effect_platervbstereo.cpp

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/* Stereo plate reverb for Teensy 4
*
* Adapted for MiniDexed (Holger Wirtz <dcoredump@googlemail.com>)
*
* Author: Piotr Zapart
* www.hexefx.com
*
* Copyright (c) 2020 by Piotr Zapart
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice, development funding notice, and this permission
* notice shall be included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <stdio.h>
#include <cstdlib>
#include <assert.h>
#include "effect_platervbstereo.h"
#define INP_ALLP_COEFF (0.65f) // default input allpass coeff
#define LOOP_ALLOP_COEFF (0.65f) // default loop allpass coeff
#define HI_LOSS_FREQ (0.3f) // scaled center freq for the treble loss filter
// #define HI_LOSS_FREQ_MAX (0.08f)
#define LO_LOSS_FREQ (0.06f) // scaled center freq for the bass loss filter
#define LFO_AMPL_BITS (5) // 2^LFO_AMPL_BITS will be the LFO amplitude
#define LFO_AMPL ((1<<LFO_AMPL_BITS) + 1) // lfo amplitude
#define LFO_READ_OFFSET (LFO_AMPL>>1) // read offset = half the amplitude
#define LFO_FRAC_BITS (16 - LFO_AMPL_BITS) // fractional part used for linear interpolation
#define LFO_FRAC_MASK ((1<<LFO_FRAC_BITS)-1) // mask for the above
#define LFO1_FREQ_HZ (1.37f) // LFO1 frequency in Hz
#define LFO2_FREQ_HZ (1.52f) // LFO2 frequency in Hz
#define RV_MASTER_LOWPASS_F (0.6f) // master lowpass scaled frequency coeff.
const int16_t AudioWaveformSine[257] = {
0, 804, 1608, 2410, 3212, 4011, 4808, 5602, 6393, 7179,
7962, 8739, 9512, 10278, 11039, 11793, 12539, 13279, 14010, 14732,
15446, 16151, 16846, 17530, 18204, 18868, 19519, 20159, 20787, 21403,
22005, 22594, 23170, 23731, 24279, 24811, 25329, 25832, 26319, 26790,
27245, 27683, 28105, 28510, 28898, 29268, 29621, 29956, 30273, 30571,
30852, 31113, 31356, 31580, 31785, 31971, 32137, 32285, 32412, 32521,
32609, 32678, 32728, 32757, 32767, 32757, 32728, 32678, 32609, 32521,
32412, 32285, 32137, 31971, 31785, 31580, 31356, 31113, 30852, 30571,
30273, 29956, 29621, 29268, 28898, 28510, 28105, 27683, 27245, 26790,
26319, 25832, 25329, 24811, 24279, 23731, 23170, 22594, 22005, 21403,
20787, 20159, 19519, 18868, 18204, 17530, 16846, 16151, 15446, 14732,
14010, 13279, 12539, 11793, 11039, 10278, 9512, 8739, 7962, 7179,
6393, 5602, 4808, 4011, 3212, 2410, 1608, 804, 0, -804,
-1608, -2410, -3212, -4011, -4808, -5602, -6393, -7179, -7962, -8739,
-9512,-10278,-11039,-11793,-12539,-13279,-14010,-14732,-15446,-16151,
-16846,-17530,-18204,-18868,-19519,-20159,-20787,-21403,-22005,-22594,
-23170,-23731,-24279,-24811,-25329,-25832,-26319,-26790,-27245,-27683,
-28105,-28510,-28898,-29268,-29621,-29956,-30273,-30571,-30852,-31113,
-31356,-31580,-31785,-31971,-32137,-32285,-32412,-32521,-32609,-32678,
-32728,-32757,-32767,-32757,-32728,-32678,-32609,-32521,-32412,-32285,
-32137,-31971,-31785,-31580,-31356,-31113,-30852,-30571,-30273,-29956,
-29621,-29268,-28898,-28510,-28105,-27683,-27245,-26790,-26319,-25832,
-25329,-24811,-24279,-23731,-23170,-22594,-22005,-21403,-20787,-20159,
-19519,-18868,-18204,-17530,-16846,-16151,-15446,-14732,-14010,-13279,
-12539,-11793,-11039,-10278, -9512, -8739, -7962, -7179, -6393, -5602,
-4808, -4011, -3212, -2410, -1608, -804, 0
};
AudioEffectPlateReverb::AudioEffectPlateReverb(float32_t samplerate)
{
input_attn = 0.5f;
in_allp_k = INP_ALLP_COEFF;
memset(in_allp1_bufL, 0, sizeof(in_allp1_bufL));
memset(in_allp2_bufL, 0, sizeof(in_allp2_bufL));
memset(in_allp3_bufL, 0, sizeof(in_allp3_bufL));
memset(in_allp4_bufL, 0, sizeof(in_allp4_bufL));
in_allp1_idxL = 0;
in_allp2_idxL = 0;
in_allp3_idxL = 0;
in_allp4_idxL = 0;
memset(in_allp1_bufR, 0, sizeof(in_allp1_bufR));
memset(in_allp2_bufR, 0, sizeof(in_allp2_bufR));
memset(in_allp3_bufR, 0, sizeof(in_allp3_bufR));
memset(in_allp4_bufR, 0, sizeof(in_allp4_bufR));
in_allp1_idxR = 0;
in_allp2_idxR = 0;
in_allp3_idxR = 0;
in_allp4_idxR = 0;
in_allp_out_R = 0.0f;
memset(lp_allp1_buf, 0, sizeof(lp_allp1_buf));
memset(lp_allp2_buf, 0, sizeof(lp_allp2_buf));
memset(lp_allp3_buf, 0, sizeof(lp_allp3_buf));
memset(lp_allp4_buf, 0, sizeof(lp_allp4_buf));
lp_allp1_idx = 0;
lp_allp2_idx = 0;
lp_allp3_idx = 0;
lp_allp4_idx = 0;
loop_allp_k = LOOP_ALLOP_COEFF;
lp_allp_out = 0.0f;
memset(lp_dly1_buf, 0, sizeof(lp_dly1_buf));
memset(lp_dly2_buf, 0, sizeof(lp_dly2_buf));
memset(lp_dly3_buf, 0, sizeof(lp_dly3_buf));
memset(lp_dly4_buf, 0, sizeof(lp_dly4_buf));
lp_dly1_idx = 0;
lp_dly2_idx = 0;
lp_dly3_idx = 0;
lp_dly4_idx = 0;
lp_hidamp_k = 1.0f;
lp_lodamp_k = 0.0f;
lp_lowpass_f = HI_LOSS_FREQ;
lp_hipass_f = LO_LOSS_FREQ;
lpf1 = 0.0f;
lpf2 = 0.0f;
lpf3 = 0.0f;
lpf4 = 0.0f;
hpf1 = 0.0f;
hpf2 = 0.0f;
hpf3 = 0.0f;
hpf4 = 0.0f;
master_lowpass_f = RV_MASTER_LOWPASS_F;
master_lowpass_l = 0.0f;
master_lowpass_r = 0.0f;
lfo1_phase_acc = 0;
lfo1_adder = (UINT32_MAX + 1)/(samplerate * LFO1_FREQ_HZ);
lfo2_phase_acc = 0;
lfo2_adder = (UINT32_MAX + 1)/(samplerate * LFO2_FREQ_HZ);
send_level = 0.0;
}
// #define sat16(n, rshift) signed_saturate_rshift((n), 16, (rshift))
void AudioEffectPlateReverb::doReverb(uint16_t len, int16_t audioblock[][2])
{
int i;
float32_t input, acc, temp1, temp2;
uint16_t temp16;
float32_t rv_time;
// for LFOs:
int16_t lfo1_out_sin, lfo1_out_cos, lfo2_out_sin, lfo2_out_cos;
int32_t y0, y1;
int64_t y;
uint32_t idx;
static bool cleanup_done = false;
// handle bypass, 1st call will clean the buffers to avoid continuing the previous reverb tail
if (bypass)
{
if (!cleanup_done)
{
memset(in_allp1_bufL, 0, sizeof(in_allp1_bufL));
memset(in_allp2_bufL, 0, sizeof(in_allp2_bufL));
memset(in_allp3_bufL, 0, sizeof(in_allp3_bufL));
memset(in_allp4_bufL, 0, sizeof(in_allp4_bufL));
memset(in_allp1_bufR, 0, sizeof(in_allp1_bufR));
memset(in_allp2_bufR, 0, sizeof(in_allp2_bufR));
memset(in_allp3_bufR, 0, sizeof(in_allp3_bufR));
memset(in_allp4_bufR, 0, sizeof(in_allp4_bufR));
memset(lp_allp1_buf, 0, sizeof(lp_allp1_buf));
memset(lp_allp2_buf, 0, sizeof(lp_allp2_buf));
memset(lp_allp3_buf, 0, sizeof(lp_allp3_buf));
memset(lp_allp4_buf, 0, sizeof(lp_allp4_buf));
memset(lp_dly1_buf, 0, sizeof(lp_dly1_buf));
memset(lp_dly2_buf, 0, sizeof(lp_dly2_buf));
memset(lp_dly3_buf, 0, sizeof(lp_dly3_buf));
memset(lp_dly4_buf, 0, sizeof(lp_dly4_buf));
cleanup_done = true;
}
return;
}
cleanup_done = false;
rv_time = rv_time_k;
for (i=0; i < len; i++)
{
// do the LFOs
lfo1_phase_acc += lfo1_adder;
idx = lfo1_phase_acc >> 24; // 8bit lookup table address
y0 = AudioWaveformSine[idx];
y1 = AudioWaveformSine[idx+1];
idx = lfo1_phase_acc & 0x00FFFFFF; // lower 24 bit = fractional part
y = (int64_t)y0 * (0x00FFFFFF - idx);
y += (int64_t)y1 * idx;
lfo1_out_sin = (int32_t) (y >> (32-8)); // 16bit output
idx = ((lfo1_phase_acc >> 24)+64) & 0xFF;
y0 = AudioWaveformSine[idx];
y1 = AudioWaveformSine[idx + 1];
y = (int64_t)y0 * (0x00FFFFFF - idx);
y += (int64_t)y1 * idx;
lfo1_out_cos = (int32_t) (y >> (32-8)); // 16bit output
lfo2_phase_acc += lfo2_adder;
idx = lfo2_phase_acc >> 24; // 8bit lookup table address
y0 = AudioWaveformSine[idx];
y1 = AudioWaveformSine[idx+1];
idx = lfo2_phase_acc & 0x00FFFFFF; // lower 24 bit = fractional part
y = (int64_t)y0 * (0x00FFFFFF - idx);
y += (int64_t)y1 * idx;
lfo2_out_sin = (int32_t) (y >> (32-8)); //32-8->output 16bit,
idx = ((lfo2_phase_acc >> 24)+64) & 0xFF;
y0 = AudioWaveformSine[idx];
y1 = AudioWaveformSine[idx + 1];
y = (int64_t)y0 * (0x00FFFFFF - idx);
y += (int64_t)y1 * idx;
lfo2_out_cos = (int32_t) (y >> (32-8)); // 16bit output
input = (float32_t(audioblock[i][0])/32767.0f) * input_attn;
// chained input allpasses, channel L
acc = in_allp1_bufL[in_allp1_idxL] + input * in_allp_k;
in_allp1_bufL[in_allp1_idxL] = input - in_allp_k * acc;
input = acc;
if (++in_allp1_idxL >= sizeof(in_allp1_bufL)/sizeof(float32_t)) in_allp1_idxL = 0;
acc = in_allp2_bufL[in_allp2_idxL] + input * in_allp_k;
in_allp2_bufL[in_allp2_idxL] = input - in_allp_k * acc;
input = acc;
if (++in_allp2_idxL >= sizeof(in_allp2_bufL)/sizeof(float32_t)) in_allp2_idxL = 0;
acc = in_allp3_bufL[in_allp3_idxL] + input * in_allp_k;
in_allp3_bufL[in_allp3_idxL] = input - in_allp_k * acc;
input = acc;
if (++in_allp3_idxL >= sizeof(in_allp3_bufL)/sizeof(float32_t)) in_allp3_idxL = 0;
acc = in_allp4_bufL[in_allp4_idxL] + input * in_allp_k;
in_allp4_bufL[in_allp4_idxL] = input - in_allp_k * acc;
in_allp_out_L = acc;
if (++in_allp4_idxL >= sizeof(in_allp4_bufL)/sizeof(float32_t)) in_allp4_idxL = 0;
input = (float32_t(audioblock[i][1])/32767.0f) * input_attn;
// chained input allpasses, channel R
acc = in_allp1_bufR[in_allp1_idxR] + input * in_allp_k;
in_allp1_bufR[in_allp1_idxR] = input - in_allp_k * acc;
input = acc;
if (++in_allp1_idxR >= sizeof(in_allp1_bufR)/sizeof(float32_t)) in_allp1_idxR = 0;
acc = in_allp2_bufR[in_allp2_idxR] + input * in_allp_k;
in_allp2_bufR[in_allp2_idxR] = input - in_allp_k * acc;
input = acc;
if (++in_allp2_idxR >= sizeof(in_allp2_bufR)/sizeof(float32_t)) in_allp2_idxR = 0;
acc = in_allp3_bufR[in_allp3_idxR] + input * in_allp_k;
in_allp3_bufR[in_allp3_idxR] = input - in_allp_k * acc;
input = acc;
if (++in_allp3_idxR >= sizeof(in_allp3_bufR)/sizeof(float32_t)) in_allp3_idxR = 0;
acc = in_allp4_bufR[in_allp4_idxR] + input * in_allp_k;
in_allp4_bufR[in_allp4_idxR] = input - in_allp_k * acc;
in_allp_out_R = acc;
if (++in_allp4_idxR >= sizeof(in_allp4_bufR)/sizeof(float32_t)) in_allp4_idxR = 0;
// input allpases done, start loop allpases
input = lp_allp_out + in_allp_out_R;
acc = lp_allp1_buf[lp_allp1_idx] + input * loop_allp_k; // input is the lp allpass chain output
lp_allp1_buf[lp_allp1_idx] = input - loop_allp_k * acc;
input = acc;
if (++lp_allp1_idx >= sizeof(lp_allp1_buf)/sizeof(float32_t)) lp_allp1_idx = 0;
acc = lp_dly1_buf[lp_dly1_idx]; // read the end of the delay
lp_dly1_buf[lp_dly1_idx] = input; // write new sample
input = acc;
if (++lp_dly1_idx >= sizeof(lp_dly1_buf)/sizeof(float32_t)) lp_dly1_idx = 0; // update index
// hi/lo shelving filter
temp1 = input - lpf1;
lpf1 += temp1 * lp_lowpass_f;
temp2 = input - lpf1;
temp1 = lpf1 - hpf1;
hpf1 += temp1 * lp_hipass_f;
acc = lpf1 + temp2*lp_hidamp_k + hpf1*lp_lodamp_k;
acc = acc * rv_time * rv_time_scaler; // scale by the reveb time
input = acc + in_allp_out_L;
acc = lp_allp2_buf[lp_allp2_idx] + input * loop_allp_k;
lp_allp2_buf[lp_allp2_idx] = input - loop_allp_k * acc;
input = acc;
if (++lp_allp2_idx >= sizeof(lp_allp2_buf)/sizeof(float32_t)) lp_allp2_idx = 0;
acc = lp_dly2_buf[lp_dly2_idx]; // read the end of the delay
lp_dly2_buf[lp_dly2_idx] = input; // write new sample
input = acc;
if (++lp_dly2_idx >= sizeof(lp_dly2_buf)/sizeof(float32_t)) lp_dly2_idx = 0; // update index
// hi/lo shelving filter
temp1 = input - lpf2;
lpf2 += temp1 * lp_lowpass_f;
temp2 = input - lpf2;
temp1 = lpf2 - hpf2;
hpf2 += temp1 * lp_hipass_f;
acc = lpf2 + temp2*lp_hidamp_k + hpf2*lp_lodamp_k;
acc = acc * rv_time * rv_time_scaler;
input = acc + in_allp_out_R;
acc = lp_allp3_buf[lp_allp3_idx] + input * loop_allp_k;
lp_allp3_buf[lp_allp3_idx] = input - loop_allp_k * acc;
input = acc;
if (++lp_allp3_idx >= sizeof(lp_allp3_buf)/sizeof(float32_t)) lp_allp3_idx = 0;
acc = lp_dly3_buf[lp_dly3_idx]; // read the end of the delay
lp_dly3_buf[lp_dly3_idx] = input; // write new sample
input = acc;
if (++lp_dly3_idx >= sizeof(lp_dly3_buf)/sizeof(float32_t)) lp_dly3_idx = 0; // update index
// hi/lo shelving filter
temp1 = input - lpf3;
lpf3 += temp1 * lp_lowpass_f;
temp2 = input - lpf3;
temp1 = lpf3 - hpf3;
hpf3 += temp1 * lp_hipass_f;
acc = lpf3 + temp2*lp_hidamp_k + hpf3*lp_lodamp_k;
acc = acc * rv_time * rv_time_scaler;
input = acc + in_allp_out_L;
acc = lp_allp4_buf[lp_allp4_idx] + input * loop_allp_k;
lp_allp4_buf[lp_allp4_idx] = input - loop_allp_k * acc;
input = acc;
if (++lp_allp4_idx >= sizeof(lp_allp4_buf)/sizeof(float32_t)) lp_allp4_idx = 0;
acc = lp_dly4_buf[lp_dly4_idx]; // read the end of the delay
lp_dly4_buf[lp_dly4_idx] = input; // write new sample
input = acc;
if (++lp_dly4_idx >= sizeof(lp_dly4_buf)/sizeof(float32_t)) lp_dly4_idx= 0; // update index
// hi/lo shelving filter
temp1 = input - lpf4;
lpf4 += temp1 * lp_lowpass_f;
temp2 = input - lpf4;
temp1 = lpf4 - hpf4;
hpf4 += temp1 * lp_hipass_f;
acc = lpf4 + temp2*lp_hidamp_k + hpf4*lp_lodamp_k;
acc = acc * rv_time * rv_time_scaler;
lp_allp_out = acc;
// channel L:
#ifdef TAP1_MODULATED
temp16 = (lp_dly1_idx + lp_dly1_offset_L + (lfo1_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly1_buf)/sizeof(float32_t));
temp1 = lp_dly1_buf[temp16++]; // sample now
if (temp16 >= sizeof(lp_dly1_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly1_buf[temp16]; // sample next
input = (float32_t)(lfo1_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
acc = (temp1*(1.0f-input) + temp2*input)* 0.8f;
#else
temp16 = (lp_dly1_idx + lp_dly1_offset_L) % (sizeof(lp_dly1_buf)/sizeof(float32_t));
acc = lp_dly1_buf[temp16]* 0.8f;
#endif
#ifdef TAP2_MODULATED
temp16 = (lp_dly2_idx + lp_dly2_offset_L + (lfo1_out_sin>>LFO_FRAC_BITS)) % (sizeof(lp_dly2_buf)/sizeof(float32_t));
temp1 = lp_dly2_buf[temp16++];
if (temp16 >= sizeof(lp_dly2_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly2_buf[temp16];
input = (float32_t)(lfo1_out_sin & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
acc += (temp1*(1.0f-input) + temp2*input)* 0.7f;
#else
temp16 = (lp_dly2_idx + lp_dly2_offset_L) % (sizeof(lp_dly2_buf)/sizeof(float32_t));
acc += (temp1*(1.0f-input) + temp2*input)* 0.6f;
#endif
temp16 = (lp_dly3_idx + lp_dly3_offset_L + (lfo2_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly3_buf)/sizeof(float32_t));
temp1 = lp_dly3_buf[temp16++];
if (temp16 >= sizeof(lp_dly3_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly3_buf[temp16];
input = (float32_t)(lfo2_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
acc += (temp1*(1.0f-input) + temp2*input)* 0.6f;
temp16 = (lp_dly4_idx + lp_dly4_offset_L + (lfo2_out_sin>>LFO_FRAC_BITS)) % (sizeof(lp_dly4_buf)/sizeof(float32_t));
temp1 = lp_dly4_buf[temp16++];
if (temp16 >= sizeof(lp_dly4_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly4_buf[temp16];
input = (float32_t)(lfo2_out_sin & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
acc += (temp1*(1.0f-input) + temp2*input)* 0.5f;
// Master lowpass filter
temp1 = acc - master_lowpass_l;
master_lowpass_l += temp1 * master_lowpass_f;
int32_t out = audioblock[i][0] + int16_t(master_lowpass_l * 32767.0f * send_level);
if(out > INT16_MAX)
audioblock[i][0] = INT16_MAX;
else if(out < INT16_MIN)
audioblock[i][0] = INT16_MIN;
else
audioblock[i][0] = out;
// Channel R
#ifdef TAP1_MODULATED
temp16 = (lp_dly1_idx + lp_dly1_offset_R + (lfo2_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly1_buf)/sizeof(float32_t));
temp1 = lp_dly1_buf[temp16++]; // sample now
if (temp16 >= sizeof(lp_dly1_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly1_buf[temp16]; // sample next
input = (float32_t)(lfo2_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
acc = (temp1*(1.0f-input) + temp2*input)* 0.8f;
#else
temp16 = (lp_dly1_idx + lp_dly1_offset_R) % (sizeof(lp_dly1_buf)/sizeof(float32_t));
acc = lp_dly1_buf[temp16] * 0.8f;
#endif
#ifdef TAP2_MODULATED
temp16 = (lp_dly2_idx + lp_dly2_offset_R + (lfo1_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly2_buf)/sizeof(float32_t));
temp1 = lp_dly2_buf[temp16++];
if (temp16 >= sizeof(lp_dly2_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly2_buf[temp16];
input = (float32_t)(lfo1_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
acc += (temp1*(1.0f-input) + temp2*input)* 0.7f;
#else
temp16 = (lp_dly2_idx + lp_dly2_offset_R) % (sizeof(lp_dly2_buf)/sizeof(float32_t));
acc += (temp1*(1.0f-input) + temp2*input)* 0.7f;
#endif
temp16 = (lp_dly3_idx + lp_dly3_offset_R + (lfo2_out_sin>>LFO_FRAC_BITS)) % (sizeof(lp_dly3_buf)/sizeof(float32_t));
temp1 = lp_dly3_buf[temp16++];
if (temp16 >= sizeof(lp_dly3_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly3_buf[temp16];
input = (float32_t)(lfo2_out_sin & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
acc += (temp1*(1.0f-input) + temp2*input)* 0.6f;
temp16 = (lp_dly4_idx + lp_dly4_offset_R + (lfo1_out_sin>>LFO_FRAC_BITS)) % (sizeof(lp_dly4_buf)/sizeof(float32_t));
temp1 = lp_dly4_buf[temp16++];
if (temp16 >= sizeof(lp_dly4_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly4_buf[temp16];
input = (float32_t)(lfo2_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
acc += (temp1*(1.0f-input) + temp2*input)* 0.5f;
// Master lowpass filter
temp1 = acc - master_lowpass_r;
master_lowpass_r += temp1 * master_lowpass_f;
out = audioblock[i][1] + int16_t(master_lowpass_l * 32767.0f * send_level);
if(out > INT16_MAX)
audioblock[i][1] = INT16_MAX;
else if(out < INT16_MIN)
audioblock[i][1] = INT16_MIN;
else
audioblock[i][1] = out;
}
}