/* Stereo plate reverb for Teensy 4 * * 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 #include "effect_platervbstereo.h" #include "utility/dspinst.h" #include "synth_waveform.h" #define INP_ALLP_COEFF (0.65) #define LOOP_ALLOP_COEFF (0.65) #define HI_LOSS_FREQ (0.3) #define HI_LOSS_FREQ_MAX (0.08) #define LO_LOSS_FREQ (0.06) #define LFO_AMPL_BITS (5) // 2^LFO_AMPL_BITS will be the LFO amplitude #define LFO_AMPL ((1<>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< 16 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, #endif #if AUDIO_BLOCK_SAMPLES > 32 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, #endif #if AUDIO_BLOCK_SAMPLES > 48 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, #endif #if AUDIO_BLOCK_SAMPLES > 64 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, #endif #if AUDIO_BLOCK_SAMPLES > 80 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, #endif #if AUDIO_BLOCK_SAMPLES > 96 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, #endif #if AUDIO_BLOCK_SAMPLES > 112 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, #endif } }; void AudioEffectPlateReverb::update() { const audio_block_t *blockL, *blockR; #if defined(__ARM_ARCH_7EM__) audio_block_t *outblockL; audio_block_t *outblockR; 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; blockL = receiveReadOnly(0); blockR = receiveReadOnly(1); outblockL = allocate(); outblockR = allocate(); if (!outblockL || !outblockR) { if (outblockL) release(outblockL); if (outblockR) release(outblockR); if (blockL) release((audio_block_t *)blockL); if (blockR) release((audio_block_t *)blockR); return; } if (!blockL) blockL = &zeroblock; if (!blockR) blockR = &zeroblock; // convert data to float32 arm_q15_to_float((q15_t *)blockL->data, input_blockL, AUDIO_BLOCK_SAMPLES); arm_q15_to_float((q15_t *)blockR->data, input_blockR, AUDIO_BLOCK_SAMPLES); rv_time = rv_time_k; for (i=0; i < AUDIO_BLOCK_SAMPLES; 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 = input_blockL[i] * 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 = input_blockR[i] * 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*(1.0-input) + temp2*input)* 0.8; #else temp16 = (lp_dly1_idx + lp_dly1_offset_L) % (sizeof(lp_dly1_buf)/sizeof(float32_t)); acc = lp_dly1_buf[temp16]* 0.8; #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*(1.0-input) + temp2*input)* 0.7; #else temp16 = (lp_dly2_idx + lp_dly2_offset_L) % (sizeof(lp_dly2_buf)/sizeof(float32_t)); acc += (temp1*(1.0-input) + temp2*input)* 0.6; #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*(1.0-input) + temp2*input)* 0.6; 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*(1.0-input) + temp2*input)* 0.5; // Master lowpass filter temp1 = acc - master_lowpass_l; master_lowpass_l += temp1 * master_lowpass_f; outblockL->data[i] =(int16_t)(master_lowpass_l * 32767.0); //sat16(output * 30, 0); // 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*(1.0-input) + temp2*input)* 0.8; #else temp16 = (lp_dly1_idx + lp_dly1_offset_R) % (sizeof(lp_dly1_buf)/sizeof(float32_t)); acc = lp_dly1_buf[temp16] * 0.8; #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*(1.0-input) + temp2*input)* 0.7; #else temp16 = (lp_dly2_idx + lp_dly2_offset_R) % (sizeof(lp_dly2_buf)/sizeof(float32_t)); acc += (temp1*(1.0-input) + temp2*input)* 0.7; #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*(1.0-input) + temp2*input)* 0.6; 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*(1.0-input) + temp2*input)* 0.5; // Master lowpass filter temp1 = acc - master_lowpass_r; master_lowpass_r += temp1 * master_lowpass_f; outblockR->data[i] =(int16_t)(master_lowpass_r * 32767.0); } transmit(outblockL, 0); transmit(outblockR, 1); release(outblockL); release(outblockR); if (blockL != &zeroblock) release((audio_block_t *)blockL); if (blockR != &zeroblock) release((audio_block_t *)blockR); #elif defined(KINETISL) blockL = receiveReadOnly(0); if (blockL) release(blockL); blockR = receiveReadOnly(1); if (blockR) release(blockR); #endif }