wirtz
/
MicroDexed
mirror of https://codeberg.org/dcoredump/MicroDexed.git
1
1
Fork 0
Code Issues Releases Wiki Activity

Using freeverbf for T_3.6 and platereverb fro T_4.x.

Browse Source
pull/37/head
Holger Wirtz 4 years ago
parent 6c4a11c91e
commit 7c8e516caf
5 changed files with 156 additions and 120 deletions
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Show Stats Download Patch File Download Diff File
  1. 57
      MicroDexed.ino
  2. 28
      UI.hpp
  3. 9
      config.h
  4. 122
      effect_platervbstereo.cpp
  5. 60
      effect_platervbstereo.h

57
MicroDexed.ino
Unescape View File

@ -35,8 +35,11 @@
#include "effect_modulated_delay.h" #include "effect_modulated_delay.h"
#include "effect_stereo_mono.h" #include "effect_stereo_mono.h"
#include "effect_mono_stereo.h" #include "effect_mono_stereo.h"
//#include "effect_freeverbf.h" #ifdef USE_PLATEREVERB
#include "effect_platervbstereo.h" #include "effect_platervbstereo.h"
#else
#include "effect_freeverbf.h"
#endif
#include "PluginFx.h" #include "PluginFx.h"
#include "UI.hpp" #include "UI.hpp"
#include "source_microdexed.h" #include "source_microdexed.h"
@ -61,13 +64,12 @@ AudioAnalyzePeak microdexed_peak;
#if defined(USE_FX) #if defined(USE_FX)
AudioMixer4 reverb_mixer_r; AudioMixer4 reverb_mixer_r;
AudioMixer4 reverb_mixer_l; AudioMixer4 reverb_mixer_l;
//AudioEffectFreeverb freeverb_r; #ifdef USE_PLATEREVERB
//AudioEffectFreeverb freeverb_l;
//#if defined(REVERB_ANTIALIAS_FRQ)
//AudioFilterBiquad freeverb_antialias_r;
//AudioFilterBiquad freeverb_antialias_l;
//#endif
AudioEffectPlateReverb reverb; AudioEffectPlateReverb reverb;
#else
AudioEffectFreeverb freeverb_r;
AudioEffectFreeverb freeverb_l;
#endif
#endif #endif
AudioMixer4 master_mixer_r; AudioMixer4 master_mixer_r;
AudioMixer4 master_mixer_l; AudioMixer4 master_mixer_l;
@ -87,21 +89,17 @@ AudioMixer4 audio_thru_mixer_l;
// Audio chain tail // Audio chain tail
#if defined(USE_FX) #if defined(USE_FX)
//AudioConnection patchCord0(reverb_mixer_r, freeverb_r); #ifdef USE_PLATEREVERB
//AudioConnection patchCord1(reverb_mixer_l, freeverb_l);
AudioConnection patchCord0(reverb_mixer_r, 0, reverb, 0); AudioConnection patchCord0(reverb_mixer_r, 0, reverb, 0);
AudioConnection patchCord1(reverb_mixer_l, 0, reverb, 1); AudioConnection patchCord1(reverb_mixer_l, 0, reverb, 1);
AudioConnection patchCord2(reverb, 0, master_mixer_r, 3); AudioConnection patchCord2(reverb, 0, master_mixer_r, 3);
AudioConnection patchCord3(reverb, 0, master_mixer_l, 3); AudioConnection patchCord3(reverb, 0, master_mixer_l, 3);
//#if defined(REVERB_ANTIALIAS_FRQ) #else
//AudioConnection patchCord2(freeverb_r, freeverb_antialias_r); AudioConnection patchCord0(reverb_mixer_r, freeverb_r);
//AudioConnection patchCord3(freeverb_l, freeverb_antialias_l); AudioConnection patchCord1(reverb_mixer_l, freeverb_l);
//AudioConnection patchCord4(freeverb_antialias_r, 0, master_mixer_r, 3); AudioConnection patchCord2(freeverb_r, 0, master_mixer_r, 3);
//AudioConnection patchCord5(freeverb_antialias_l, 0, master_mixer_l, 3); AudioConnection patchCord3(freeverb_l, 0, master_mixer_l, 3);
//#else #endif
//AudioConnection patchCord2(freeverb_r, 0, master_mixer_r, 3);
//AudioConnection patchCord3(freeverb_l, 0, master_mixer_l, 3);
//#endif
#endif #endif
#if defined(ANTIALIAS_FRQ) #if defined(ANTIALIAS_FRQ)
AudioConnection patchCord6(master_mixer_r, antialias_r); AudioConnection patchCord6(master_mixer_r, antialias_r);
@ -409,17 +407,6 @@ void setup()
#endif #endif
#endif #endif
#if defined(USE_FX) && defined(REVERB_ANTIALIAS_FRQ)
freeverb_antialias_r.setLowpass(0, REVERB_ANTIALIAS_FRQ, 0.54);
freeverb_antialias_r.setLowpass(1, REVERB_ANTIALIAS_FRQ, 1.3);
freeverb_antialias_r.setLowpass(2, REVERB_ANTIALIAS_FRQ, 0.54);
freeverb_antialias_r.setLowpass(3, REVERB_ANTIALIAS_FRQ, 1.3);
freeverb_antialias_l.setLowpass(0, REVERB_ANTIALIAS_FRQ, 0.54);
freeverb_antialias_l.setLowpass(1, REVERB_ANTIALIAS_FRQ, 1.3);
freeverb_antialias_l.setLowpass(2, REVERB_ANTIALIAS_FRQ, 0.54);
freeverb_antialias_l.setLowpass(3, REVERB_ANTIALIAS_FRQ, 1.3);
#endif
#if defined(ANTIALIAS_FRQ) #if defined(ANTIALIAS_FRQ)
antialias_r.setLowpass(0, ANTIALIAS_FRQ, 0.54); antialias_r.setLowpass(0, ANTIALIAS_FRQ, 0.54);
antialias_r.setLowpass(1, ANTIALIAS_FRQ, 1.3); antialias_r.setLowpass(1, ANTIALIAS_FRQ, 1.3);
@ -1883,15 +1870,19 @@ void set_fx_params(void)
} }
// REVERB // REVERB
//freeverb_r.roomsize(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0)); #ifdef USE_PLATEREVERB
//freeverb_r.damping(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0));
//freeverb_l.roomsize(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0));
//freeverb_l.damping(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0));
reverb.size(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0)); // max reverb length reverb.size(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0)); // max reverb length
reverb.lowpass(0.3); // sets the reverb master lowpass filter reverb.lowpass(0.3); // sets the reverb master lowpass filter
reverb.lodamp(0.1); // amount of low end loss in the reverb tail reverb.lodamp(0.1); // amount of low end loss in the reverb tail
reverb.hidamp(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0)); // amount of treble loss in the reverb tail reverb.hidamp(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0)); // amount of treble loss in the reverb tail
reverb.diffusion(1.0); // 1.0 is the detault setting, lower it to create more "echoey" reverb reverb.diffusion(1.0); // 1.0 is the detault setting, lower it to create more "echoey" reverb
#else
freeverb_r.roomsize(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0));
freeverb_r.damping(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0));
freeverb_l.roomsize(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0));
freeverb_l.damping(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0));
#endif
master_mixer_r.gain(3, pseudo_log_curve(mapfloat(configuration.fx.reverb_level, REVERB_LEVEL_MIN, REVERB_LEVEL_MAX, 0.0, 1.0))); master_mixer_r.gain(3, pseudo_log_curve(mapfloat(configuration.fx.reverb_level, REVERB_LEVEL_MIN, REVERB_LEVEL_MAX, 0.0, 1.0)));
master_mixer_l.gain(3, pseudo_log_curve(mapfloat(configuration.fx.reverb_level, REVERB_LEVEL_MIN, REVERB_LEVEL_MAX, 0.0, 1.0))); master_mixer_l.gain(3, pseudo_log_curve(mapfloat(configuration.fx.reverb_level, REVERB_LEVEL_MIN, REVERB_LEVEL_MAX, 0.0, 1.0)));
#endif #endif

28
UI.hpp
Unescape View File

@ -25,13 +25,17 @@
#ifdef ENABLE_LCD_UI #ifdef ENABLE_LCD_UI
#ifndef _UI_HPP_ #ifndef _UI_HPP_
#define _UI_HPP_ #define _UI_HPP_
#endif
#include "config.h" #include "config.h"
#include "Disp_Plus.h" #include "Disp_Plus.h"
#include "effect_modulated_delay.h" #include "effect_modulated_delay.h"
#include "effect_stereo_mono.h" #include "effect_stereo_mono.h"
//#include "effect_freeverbf.h" #ifdef USE_PLATEREVERB
#include "effect_platervbstereo.h" #include "effect_platervbstereo.h"
#else
#include "effect_freeverbf.h"
#endif
#include "dexed.h" #include "dexed.h"
#include <LCDMenuLib2.h> #include <LCDMenuLib2.h>
@ -82,9 +86,12 @@ extern AudioAnalyzePeak microdexed_peak;
#if defined(USE_FX) #if defined(USE_FX)
extern AudioMixer4 reverb_mixer_r; extern AudioMixer4 reverb_mixer_r;
extern AudioMixer4 reverb_mixer_l; extern AudioMixer4 reverb_mixer_l;
//extern AudioEffectFreeverb freeverb_r; #ifdef USE_PLATEREVERB
//extern AudioEffectFreeverb freeverb_l;
extern AudioEffectPlateReverb reverb; extern AudioEffectPlateReverb reverb;
#else
extern AudioEffectFreeverb freeverb_r;
extern AudioEffectFreeverb freeverb_l;
#endif
#endif #endif
extern AudioMixer4 master_mixer_r; extern AudioMixer4 master_mixer_r;
extern AudioMixer4 master_mixer_l; extern AudioMixer4 master_mixer_l;
@ -912,9 +919,12 @@ void UI_func_reverb_roomsize(uint8_t param)
} }
lcd_display_bar_int("Reverb Room", configuration.fx.reverb_roomsize, 1.0, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 3, false, false, false); lcd_display_bar_int("Reverb Room", configuration.fx.reverb_roomsize, 1.0, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 3, false, false, false);
//freeverb_r.roomsize(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0)); #ifdef USE_PLATEREVERB
//freeverb_l.roomsize(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0));
reverb.size(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0)); reverb.size(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0));
#else
freeverb_r.roomsize(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0));
freeverb_l.roomsize(mapfloat(configuration.fx.reverb_roomsize, REVERB_ROOMSIZE_MIN, REVERB_ROOMSIZE_MAX, 0.0, 1.0));
#endif
} }
if (LCDML.FUNC_close()) // ****** STABLE END ********* if (LCDML.FUNC_close()) // ****** STABLE END *********
@ -947,9 +957,12 @@ void UI_func_reverb_damping(uint8_t param)
lcd_display_bar_int("Reverb Damp.", configuration.fx.reverb_damping, 1.0, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 3, false, false, false); lcd_display_bar_int("Reverb Damp.", configuration.fx.reverb_damping, 1.0, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 3, false, false, false);
//freeverb_l.damping(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0)); #ifdef USE_PLATEREVERB
//freeverb_r.damping(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0));
reverb.hidamp(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0)); reverb.hidamp(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0));
#else
freeverb_l.damping(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0));
freeverb_r.damping(mapfloat(configuration.fx.reverb_damping, REVERB_DAMPING_MIN, REVERB_DAMPING_MAX, 0.0, 1.0));
#endif
} }
if (LCDML.FUNC_close()) // ****** STABLE END ********* if (LCDML.FUNC_close()) // ****** STABLE END *********
@ -5832,4 +5845,3 @@ void string_trim(char *s)
} }
#endif #endif
#endif

9
config.h
Unescape View File

@ -100,7 +100,7 @@
#define SHOW_CPU_LOAD_MSEC 5000 #define SHOW_CPU_LOAD_MSEC 5000
//************************************************************************************************* //*************************************************************************************************
//* DEXED AND EFECTS SETTINGS //* DEXED AND EFFECTS SETTINGS
//************************************************************************************************* //*************************************************************************************************
#define DEXED_ENGINE DEXED_ENGINE_MODERN // DEXED_ENGINE_MARKI // DEXED_ENGINE_OPL #define DEXED_ENGINE DEXED_ENGINE_MODERN // DEXED_ENGINE_MARKI // DEXED_ENGINE_OPL
@ -113,16 +113,15 @@
#define MOD_WAVEFORM WAVEFORM_TRIANGLE // WAVEFORM_SINE WAVEFORM_TRIANGLE WAVEFORM_SAWTOOTH WAVEFORM_SAWTOOTH_REVERSE #define MOD_WAVEFORM WAVEFORM_TRIANGLE // WAVEFORM_SINE WAVEFORM_TRIANGLE WAVEFORM_SAWTOOTH WAVEFORM_SAWTOOTH_REVERSE
#define MOD_FILTER_OUTPUT MOD_BUTTERWORTH_FILTER_OUTPUT // MOD_LINKWITZ_RILEY_FILTER_OUTPUT MOD_BUTTERWORTH_FILTER_OUTPUT MOD_NO_FILTER_OUTPUT #define MOD_FILTER_OUTPUT MOD_BUTTERWORTH_FILTER_OUTPUT // MOD_LINKWITZ_RILEY_FILTER_OUTPUT MOD_BUTTERWORTH_FILTER_OUTPUT MOD_NO_FILTER_OUTPUT
#define MOD_FILTER_CUTOFF_HZ 2000 #define MOD_FILTER_CUTOFF_HZ 2000
// REVERB parameters
//#define REVERB_ANTIALIAS_FRQ 7500
// ANTIALIAS frequency
//#define ANTIALIAS_FRQ 15000
// SGTL5000 // SGTL5000
#ifdef TEENSY_AUDIO_BOARD #ifdef TEENSY_AUDIO_BOARD
#define SGTL5000_AUDIO_ENHANCE 1 #define SGTL5000_AUDIO_ENHANCE 1
//#define SGTL5000_AUDIO_THRU 1 //#define SGTL5000_AUDIO_THRU 1
#define SGTL5000_HEADPHONE_VOLUME 0.8 #define SGTL5000_HEADPHONE_VOLUME 0.8
#endif #endif
#if defined(TEENSY4)
#define USE_PLATEREVERB 1
#endif
//************************************************************************************************* //*************************************************************************************************
//* AUDIO SOFTWARE SETTINGS //* AUDIO SOFTWARE SETTINGS

122
effect_platervbstereo.cpp
Unescape View File

@ -34,12 +34,12 @@
#include "utility/dspinst.h" #include "utility/dspinst.h"
#include "synth_waveform.h" #include "synth_waveform.h"
#define INP_ALLP_COEFF (0.65) #define INP_ALLP_COEFF (0.65f)
#define LOOP_ALLOP_COEFF (0.65) #define LOOP_ALLOP_COEFF (0.65f)
#define HI_LOSS_FREQ (0.3) #define HI_LOSS_FREQ (0.3f)
#define HI_LOSS_FREQ_MAX (0.08) #define HI_LOSS_FREQ_MAX (0.08f)
#define LO_LOSS_FREQ (0.06) #define LO_LOSS_FREQ (0.06f)
#define LFO_AMPL_BITS (5) // 2^LFO_AMPL_BITS will be the LFO amplitude #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_AMPL ((1<<LFO_AMPL_BITS) + 1) // lfo amplitude
@ -47,10 +47,10 @@
#define LFO_FRAC_BITS (16 - LFO_AMPL_BITS) // fractional part used for linear interpolation #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 LFO_FRAC_MASK ((1<<LFO_FRAC_BITS)-1) // mask for the above
#define LFO1_FREQ_HZ (1.37) // LFO1 frequency in Hz #define LFO1_FREQ_HZ (1.37f) // LFO1 frequency in Hz
#define LFO2_FREQ_HZ (1.52) // LFO2 frequency in Hz #define LFO2_FREQ_HZ (1.52f) // LFO2 frequency in Hz
#define RV_MASTER_LOWPASS_F (0.6) // master lowpass scaled frequency coeff. #define RV_MASTER_LOWPASS_F (0.6f) // master lowpass scaled frequency coeff.
extern "C" { extern "C" {
extern const int16_t AudioWaveformSine[257]; extern const int16_t AudioWaveformSine[257];
@ -71,20 +71,20 @@ float32_t DMAMEM in_allp2_bufR[520];
float32_t DMAMEM in_allp3_bufR[956]; float32_t DMAMEM in_allp3_bufR[956];
float32_t DMAMEM in_allp4_bufR[1289]; float32_t DMAMEM in_allp4_bufR[1289];
float32_t DMAMEM lp_allp1_buf[1303]; // loop allpass buffers float32_t DMAMEM lp_allp1_buf[2303]; // loop allpass buffers
float32_t DMAMEM lp_allp2_buf[905]; float32_t DMAMEM lp_allp2_buf[2905];
float32_t DMAMEM lp_allp3_buf[1175]; float32_t DMAMEM lp_allp3_buf[3175];
float32_t DMAMEM lp_allp4_buf[1398]; float32_t DMAMEM lp_allp4_buf[2398];
float32_t DMAMEM lp_dly1_buf[1423]; float32_t DMAMEM lp_dly1_buf[3423];
float32_t DMAMEM lp_dly2_buf[1589]; float32_t DMAMEM lp_dly2_buf[4589];
float32_t DMAMEM lp_dly3_buf[1365]; float32_t DMAMEM lp_dly3_buf[4365];
float32_t DMAMEM lp_dly4_buf[1698]; float32_t DMAMEM lp_dly4_buf[3698];
#endif #endif
AudioEffectPlateReverb::AudioEffectPlateReverb() : AudioStream(2, inputQueueArray) AudioEffectPlateReverb::AudioEffectPlateReverb() : AudioStream(2, inputQueueArray)
{ {
input_attn = 0.5; input_attn = 0.5f;
in_allp_k = INP_ALLP_COEFF; in_allp_k = INP_ALLP_COEFF;
memset(in_allp1_bufL, 0, sizeof(in_allp1_bufL)); memset(in_allp1_bufL, 0, sizeof(in_allp1_bufL));
@ -105,7 +105,7 @@ AudioEffectPlateReverb::AudioEffectPlateReverb() : AudioStream(2, inputQueueArra
in_allp3_idxR = 0; in_allp3_idxR = 0;
in_allp4_idxR = 0; in_allp4_idxR = 0;
in_allp_out_R = 0; in_allp_out_R = 0.0f;
memset(lp_allp1_buf, 0, sizeof(lp_allp1_buf)); memset(lp_allp1_buf, 0, sizeof(lp_allp1_buf));
memset(lp_allp2_buf, 0, sizeof(lp_allp2_buf)); memset(lp_allp2_buf, 0, sizeof(lp_allp2_buf));
@ -116,7 +116,7 @@ AudioEffectPlateReverb::AudioEffectPlateReverb() : AudioStream(2, inputQueueArra
lp_allp3_idx = 0; lp_allp3_idx = 0;
lp_allp4_idx = 0; lp_allp4_idx = 0;
loop_allp_k = LOOP_ALLOP_COEFF; loop_allp_k = LOOP_ALLOP_COEFF;
lp_allp_out = 0; lp_allp_out = 0.0f;
memset(lp_dly1_buf, 0, sizeof(lp_dly1_buf)); memset(lp_dly1_buf, 0, sizeof(lp_dly1_buf));
memset(lp_dly2_buf, 0, sizeof(lp_dly2_buf)); memset(lp_dly2_buf, 0, sizeof(lp_dly2_buf));
@ -127,25 +127,25 @@ AudioEffectPlateReverb::AudioEffectPlateReverb() : AudioStream(2, inputQueueArra
lp_dly3_idx = 0; lp_dly3_idx = 0;
lp_dly4_idx = 0; lp_dly4_idx = 0;
lp_hidamp_k = 1.0; lp_hidamp_k = 1.0f;
lp_lodamp_k = 0.0; lp_lodamp_k = 0.0f;
lp_lowpass_f = HI_LOSS_FREQ; lp_lowpass_f = HI_LOSS_FREQ;
lp_hipass_f = LO_LOSS_FREQ; lp_hipass_f = LO_LOSS_FREQ;
lpf1 = 0; lpf1 = 0.0f;
lpf2 = 0; lpf2 = 0.0f;
lpf3 = 0; lpf3 = 0.0f;
lpf4 = 0; lpf4 = 0.0f;
hpf1 = 0; hpf1 = 0.0f;
hpf2 = 0; hpf2 = 0.0f;
hpf3 = 0; hpf3 = 0.0f;
hpf4 = 0; hpf4 = 0.0f;
master_lowpass_f = RV_MASTER_LOWPASS_F; master_lowpass_f = RV_MASTER_LOWPASS_F;
master_lowpass_l = 0; master_lowpass_l = 0.0f;
master_lowpass_r = 0; master_lowpass_r = 0.0f;
lfo1_phase_acc = 0; lfo1_phase_acc = 0;
lfo1_adder = (UINT32_MAX + 1)/(AUDIO_SAMPLE_RATE_EXACT * LFO1_FREQ_HZ); lfo1_adder = (UINT32_MAX + 1)/(AUDIO_SAMPLE_RATE_EXACT * LFO1_FREQ_HZ);
@ -153,7 +153,7 @@ AudioEffectPlateReverb::AudioEffectPlateReverb() : AudioStream(2, inputQueueArra
lfo2_adder = (UINT32_MAX + 1)/(AUDIO_SAMPLE_RATE_EXACT * LFO2_FREQ_HZ); lfo2_adder = (UINT32_MAX + 1)/(AUDIO_SAMPLE_RATE_EXACT * LFO2_FREQ_HZ);
} }
#define sat16(n, rshift) signed_saturate_rshift((n), 16, (rshift)) // #define sat16(n, rshift) signed_saturate_rshift((n), 16, (rshift))
// TODO: move this to one of the data files, use in output_adat.cpp, output_tdm.cpp, etc // TODO: move this to one of the data files, use in output_adat.cpp, output_tdm.cpp, etc
static const audio_block_t zeroblock = { static const audio_block_t zeroblock = {
@ -199,6 +199,35 @@ void AudioEffectPlateReverb::update()
int32_t y0, y1; int32_t y0, y1;
int64_t y; int64_t y;
uint32_t idx; 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;
blockL = receiveReadOnly(0); blockL = receiveReadOnly(0);
blockR = receiveReadOnly(1); blockR = receiveReadOnly(1);
@ -214,6 +243,7 @@ void AudioEffectPlateReverb::update()
if (!blockL) blockL = &zeroblock; if (!blockL) blockL = &zeroblock;
if (!blockR) blockR = &zeroblock; if (!blockR) blockR = &zeroblock;
// convert data to float32 // convert data to float32
arm_q15_to_float((q15_t *)blockL->data, input_blockL, AUDIO_BLOCK_SAMPLES); 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); arm_q15_to_float((q15_t *)blockR->data, input_blockR, AUDIO_BLOCK_SAMPLES);
@ -385,10 +415,10 @@ void AudioEffectPlateReverb::update()
if (temp16 >= sizeof(lp_dly1_buf)/sizeof(float32_t)) temp16 = 0; if (temp16 >= sizeof(lp_dly1_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly1_buf[temp16]; // sample next temp2 = lp_dly1_buf[temp16]; // sample next
input = (float32_t)(lfo1_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k 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; acc = (temp1*(1.0f-input) + temp2*input)* 0.8f;
#else #else
temp16 = (lp_dly1_idx + lp_dly1_offset_L) % (sizeof(lp_dly1_buf)/sizeof(float32_t)); temp16 = (lp_dly1_idx + lp_dly1_offset_L) % (sizeof(lp_dly1_buf)/sizeof(float32_t));
acc = lp_dly1_buf[temp16]* 0.8; acc = lp_dly1_buf[temp16]* 0.8f;
#endif #endif
@ -398,10 +428,10 @@ void AudioEffectPlateReverb::update()
if (temp16 >= sizeof(lp_dly2_buf)/sizeof(float32_t)) temp16 = 0; if (temp16 >= sizeof(lp_dly2_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly2_buf[temp16]; temp2 = lp_dly2_buf[temp16];
input = (float32_t)(lfo1_out_sin & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k 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; acc += (temp1*(1.0f-input) + temp2*input)* 0.7f;
#else #else
temp16 = (lp_dly2_idx + lp_dly2_offset_L) % (sizeof(lp_dly2_buf)/sizeof(float32_t)); temp16 = (lp_dly2_idx + lp_dly2_offset_L) % (sizeof(lp_dly2_buf)/sizeof(float32_t));
acc += (temp1*(1.0-input) + temp2*input)* 0.6; acc += (temp1*(1.0f-input) + temp2*input)* 0.6f;
#endif #endif
temp16 = (lp_dly3_idx + lp_dly3_offset_L + (lfo2_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly3_buf)/sizeof(float32_t)); temp16 = (lp_dly3_idx + lp_dly3_offset_L + (lfo2_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly3_buf)/sizeof(float32_t));
@ -409,20 +439,20 @@ void AudioEffectPlateReverb::update()
if (temp16 >= sizeof(lp_dly3_buf)/sizeof(float32_t)) temp16 = 0; if (temp16 >= sizeof(lp_dly3_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly3_buf[temp16]; temp2 = lp_dly3_buf[temp16];
input = (float32_t)(lfo2_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k 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; 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)); 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++]; temp1 = lp_dly4_buf[temp16++];
if (temp16 >= sizeof(lp_dly4_buf)/sizeof(float32_t)) temp16 = 0; if (temp16 >= sizeof(lp_dly4_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly4_buf[temp16]; temp2 = lp_dly4_buf[temp16];
input = (float32_t)(lfo2_out_sin & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k 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; acc += (temp1*(1.0f-input) + temp2*input)* 0.5f;
// Master lowpass filter // Master lowpass filter
temp1 = acc - master_lowpass_l; temp1 = acc - master_lowpass_l;
master_lowpass_l += temp1 * master_lowpass_f; master_lowpass_l += temp1 * master_lowpass_f;
outblockL->data[i] =(int16_t)(master_lowpass_l * 32767.0); //sat16(output * 30, 0); outblockL->data[i] =(int16_t)(master_lowpass_l * 32767.0f); //sat16(output * 30, 0);
// Channel R // Channel R
#ifdef TAP1_MODULATED #ifdef TAP1_MODULATED
@ -432,10 +462,10 @@ void AudioEffectPlateReverb::update()
temp2 = lp_dly1_buf[temp16]; // sample next temp2 = lp_dly1_buf[temp16]; // sample next
input = (float32_t)(lfo2_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k 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; acc = (temp1*(1.0f-input) + temp2*input)* 0.8f;
#else #else
temp16 = (lp_dly1_idx + lp_dly1_offset_R) % (sizeof(lp_dly1_buf)/sizeof(float32_t)); temp16 = (lp_dly1_idx + lp_dly1_offset_R) % (sizeof(lp_dly1_buf)/sizeof(float32_t));
acc = lp_dly1_buf[temp16] * 0.8; acc = lp_dly1_buf[temp16] * 0.8f;
#endif #endif
#ifdef TAP2_MODULATED #ifdef TAP2_MODULATED
temp16 = (lp_dly2_idx + lp_dly2_offset_R + (lfo1_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly2_buf)/sizeof(float32_t)); temp16 = (lp_dly2_idx + lp_dly2_offset_R + (lfo1_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly2_buf)/sizeof(float32_t));
@ -443,29 +473,29 @@ void AudioEffectPlateReverb::update()
if (temp16 >= sizeof(lp_dly2_buf)/sizeof(float32_t)) temp16 = 0; if (temp16 >= sizeof(lp_dly2_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly2_buf[temp16]; temp2 = lp_dly2_buf[temp16];
input = (float32_t)(lfo1_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k 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; acc += (temp1*(1.0f-input) + temp2*input)* 0.7f;
#else #else
temp16 = (lp_dly2_idx + lp_dly2_offset_R) % (sizeof(lp_dly2_buf)/sizeof(float32_t)); temp16 = (lp_dly2_idx + lp_dly2_offset_R) % (sizeof(lp_dly2_buf)/sizeof(float32_t));
acc += (temp1*(1.0-input) + temp2*input)* 0.7; acc += (temp1*(1.0f-input) + temp2*input)* 0.7f;
#endif #endif
temp16 = (lp_dly3_idx + lp_dly3_offset_R + (lfo2_out_sin>>LFO_FRAC_BITS)) % (sizeof(lp_dly3_buf)/sizeof(float32_t)); 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++]; temp1 = lp_dly3_buf[temp16++];
if (temp16 >= sizeof(lp_dly3_buf)/sizeof(float32_t)) temp16 = 0; if (temp16 >= sizeof(lp_dly3_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly3_buf[temp16]; temp2 = lp_dly3_buf[temp16];
input = (float32_t)(lfo2_out_sin & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k 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; 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)); 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++]; temp1 = lp_dly4_buf[temp16++];
if (temp16 >= sizeof(lp_dly4_buf)/sizeof(float32_t)) temp16 = 0; if (temp16 >= sizeof(lp_dly4_buf)/sizeof(float32_t)) temp16 = 0;
temp2 = lp_dly4_buf[temp16]; temp2 = lp_dly4_buf[temp16];
input = (float32_t)(lfo2_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k 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; acc += (temp1*(1.0f-input) + temp2*input)* 0.5f;
// Master lowpass filter // Master lowpass filter
temp1 = acc - master_lowpass_r; temp1 = acc - master_lowpass_r;
master_lowpass_r += temp1 * master_lowpass_f; master_lowpass_r += temp1 * master_lowpass_f;
outblockR->data[i] =(int16_t)(master_lowpass_r * 32767.0); outblockR->data[i] =(int16_t)(master_lowpass_r * 32767.0f);
} }
transmit(outblockL, 0); transmit(outblockL, 0);

60
effect_platervbstereo.h
Unescape View File

@ -69,51 +69,55 @@ public:
void size(float n) void size(float n)
{ {
n = constrain(n, 0.0, 1.0); n = constrain(n, 0.0f, 1.0f);
n = map (n, 0.0, 1.0, 0.2, rv_time_k_max); n = map (n, 0.0f, 1.0f, 0.2f, rv_time_k_max);
float32_t attn = 0.5 * map(n, 0.0, rv_time_k_max, 0.5, 1.0); float32_t attn = 0.5f * map(n, 0.0f, rv_time_k_max, 0.5f, 1.0f);
AudioNoInterrupts(); __disable_irq();
rv_time_k = n; rv_time_k = n;
input_attn = attn; input_attn = attn;
AudioInterrupts(); __enable_irq();
} }
void hidamp(float n) void hidamp(float n)
{ {
n = constrain(n, 0.0, 1.0); n = constrain(n, 0.0f, 1.0f);
AudioNoInterrupts(); __disable_irq();
lp_hidamp_k = 1.0 - n; lp_hidamp_k = 1.0f - n;
AudioInterrupts(); __enable_irq();
} }
void lodamp(float n) void lodamp(float n)
{ {
n = constrain(n, 0.0, 1.0); n = constrain(n, 0.0f, 1.0f);
AudioNoInterrupts(); __disable_irq();
lp_lodamp_k = -n; lp_lodamp_k = -n;
rv_time_scaler = 1.0 - n * 0.12; // limit the max reverb time, otherwise it will clip rv_time_scaler = 1.0f - n * 0.12f; // limit the max reverb time, otherwise it will clip
AudioInterrupts(); __enable_irq();
} }
void lowpass(float n) void lowpass(float n)
{ {
n = constrain(n, 0.0, 1.0); n = constrain(n, 0.0f, 1.0f);
n = map(n*n*n, 0.0, 1.0, 0.05, 1.0); n = map(n*n*n, 0.0f, 1.0f, 0.05f, 1.0f);
master_lowpass_f = n; master_lowpass_f = n;
} }
void diffusion(float n) void diffusion(float n)
{ {
n = constrain(n, 0.0, 1.0); n = constrain(n, 0.0f, 1.0f);
n = map(n, 0.0, 1.0, 0.005, 0.65); n = map(n, 0.0f, 1.0f, 0.005f, 0.65f);
AudioNoInterrupts(); __disable_irq();
in_allp_k = n; in_allp_k = n;
loop_allp_k = n; loop_allp_k = n;
AudioInterrupts(); __enable_irq();
} }
float32_t get_size(void) {return rv_time_k;} float32_t get_size(void) {return rv_time_k;}
bool get_bypass(void) {return bypass;}
void set_bypass(bool state) {bypass = state;};
void tgl_bypass(void) {bypass ^=1;}
private: private:
bool bypass = false;
audio_block_t *inputQueueArray[2]; audio_block_t *inputQueueArray[2];
#ifndef REVERB_USE_DMAMEM #ifndef REVERB_USE_DMAMEM
float32_t input_blockL[AUDIO_BLOCK_SAMPLES]; float32_t input_blockL[AUDIO_BLOCK_SAMPLES];
@ -145,10 +149,10 @@ private:
uint16_t in_allp4_idxR; uint16_t in_allp4_idxR;
float32_t in_allp_out_R; // R allpass chain output float32_t in_allp_out_R; // R allpass chain output
#ifndef REVERB_USE_DMAMEM #ifndef REVERB_USE_DMAMEM
float32_t lp_allp1_buf[1303]; // loop allpass buffers float32_t lp_allp1_buf[2303]; // loop allpass buffers
float32_t lp_allp2_buf[1905]; float32_t lp_allp2_buf[2905];
float32_t lp_allp3_buf[1175]; float32_t lp_allp3_buf[3175];
float32_t lp_allp4_buf[1398]; float32_t lp_allp4_buf[2398];
#endif #endif
uint16_t lp_allp1_idx; uint16_t lp_allp1_idx;
uint16_t lp_allp2_idx; uint16_t lp_allp2_idx;
@ -157,10 +161,10 @@ private:
float32_t loop_allp_k; // loop allpass coeff (default 0.6) float32_t loop_allp_k; // loop allpass coeff (default 0.6)
float32_t lp_allp_out; float32_t lp_allp_out;
#ifndef REVERB_USE_DMAMEM #ifndef REVERB_USE_DMAMEM
float32_t lp_dly1_buf[1423]; float32_t lp_dly1_buf[3423];
float32_t lp_dly2_buf[1589]; float32_t lp_dly2_buf[4589];
float32_t lp_dly3_buf[1365]; float32_t lp_dly3_buf[4365];
float32_t lp_dly4_buf[1698]; float32_t lp_dly4_buf[3698];
#endif #endif
uint16_t lp_dly1_idx; uint16_t lp_dly1_idx;
uint16_t lp_dly2_idx; uint16_t lp_dly2_idx;
@ -197,7 +201,7 @@ private:
float32_t master_lowpass_l; float32_t master_lowpass_l;
float32_t master_lowpass_r; float32_t master_lowpass_r;
const float32_t rv_time_k_max = 0.95; const float32_t rv_time_k_max = 0.95f;
float32_t rv_time_k; // reverb time coeff float32_t rv_time_k; // reverb time coeff
float32_t rv_time_scaler; // with high lodamp settings lower the max reverb time to avoid clipping float32_t rv_time_scaler; // with high lodamp settings lower the max reverb time to avoid clipping

Write Preview
Loading…
Cancel
Save