New Teensy 4.1 DIY Synthesizer
- Thread starter Rolfdegen
- Start date
Rolfdegen
Well-known member
Thanks 
The Jeannie housing is a cooperation between a German manufacturer who makes housings for Erica synths and tubeohm.com
Another case manufacturer is http://www.grawart.pl/
Greetings. Rolf
The Jeannie housing is a cooperation between a German manufacturer who makes housings for Erica synths and tubeohm.com
Another case manufacturer is http://www.grawart.pl/
Greetings. Rolf
Rolfdegen
Well-known member
[FONT="]There is a virtual midi controller for the Jeannie. I am still working on it...
[/FONT]The midi controller receives and sends the data via USB. Some functions are too special and can only be programmed or set on the Jeannie.
Now I want to display the prg no. and name when switching. Maybe someone knows how to program the Ctrlr App. For help I would be grateful in any case.
[FONT="]
[/FONT]
[/FONT]The midi controller receives and sends the data via USB. Some functions are too special and can only be programmed or set on the Jeannie.
Now I want to display the prg no. and name when switching. Maybe someone knows how to program the Ctrlr App. For help I would be grateful in any case.
[FONT="]
[/FONT]
Just saw this on Matrix Synth! Sounds good Rolf!
https://www.matrixsynth.com/2023/09/jeannie-vowel-sounds.html
https://www.matrixsynth.com/2023/09/jeannie-vowel-sounds.html
Rolfdegen
Well-known member
Hallo Teensy friends..
I would like to implement the Mutable Braids Vowel sound into my Jeannie DIY synthesizer. It sounds very good without pitch modulation. When I turn the pitch wheel up I get distortions in the sound. There seems to be an overflow of increments in the oscillator. I use the Teensy4.1 Audio lib and the AudioSynthWaveformModulated as an oscillator.
Vowel Sound test without pitch modulation. I change parameter_a for vokal and paramter_b for format_shift
Vowel Sound test with pitch modulation..
Vowel Sound with old shruthi core inbuild Jeannie..
Shruthi has only one formant parameter
Thanks for help..
synth_waveform.h
synth_waveform.cpp
I would like to implement the Mutable Braids Vowel sound into my Jeannie DIY synthesizer. It sounds very good without pitch modulation. When I turn the pitch wheel up I get distortions in the sound. There seems to be an overflow of increments in the oscillator. I use the Teensy4.1 Audio lib and the AudioSynthWaveformModulated as an oscillator.
Vowel Sound test without pitch modulation. I change parameter_a for vokal and paramter_b for format_shift
Vowel Sound test with pitch modulation..
Vowel Sound with old shruthi core inbuild Jeannie..
Shruthi has only one formant parameter
Thanks for help..
synth_waveform.h
Code:
/* Audio Library for Teensy 3.X
* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
*
* 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.
ElectroTechnique 2020
Added WAVEFORM_SILENT, sync()
*/
#ifndef synth_waveform_h_
#define synth_waveform_h_
#include <Arduino.h>
#include "AudioStream.h"
#include "arm_math.h"
// waveforms.c
extern "C" {
extern const int16_t AudioWaveformSine[257];
}
extern uint8_t LFO1mode;
extern uint8_t LFO2mode;
extern uint8_t LFO3mode;
extern boolean lfo1oneShoot;
extern boolean lfo2oneShoot;
extern boolean lfo3oneShoot;
extern uint8_t LFO1phase;
extern uint8_t LFO2phase;
extern uint8_t LFO3phase;
extern uint8_t lfo1ph;
extern uint8_t lfo2ph;
extern uint8_t lfo3ph;
extern int8_t Lfo3Modoutput;
extern boolean LFO1randomFlag;
extern boolean LFO2randomFlag;
extern boolean LFO3randomFlag;
extern uint32_t LFO1delayTime;
extern uint32_t LFO2delayTime;
extern uint32_t LFO3delayTime;
extern uint8_t SupersawSpreadA;
extern uint8_t SupersawSpreadB;
extern float Supersaw_gain1A;
extern float Supersaw_gain2A;
extern float Supersaw_gain1B;
extern float Supersaw_gain2B;
extern uint32_t rng_value32;
extern
// Oscillator
#define WAVEFORM_SINE 0
#define WAVEFORM_TRIANGLE 1
#define WAVEFORM_SAWTOOTH 3
#define WAVEFORM_SQUARE 2
#define WAVEFORM_PULSE 4
#define WAVEFORM_SAWTOOTH_REVERSE 5
#define WAVEFORM_SAMPLE_HOLD 6
#define WAVEFORM_TRIANGLE_VARIABLE 7
#define WAVEFORM_BANDLIMIT_SAWTOOTH_REVERSE 8
#define WAVEFORM_BANDLIMIT_SAWTOOTH 9
#define WAVEFORM_BANDLIMIT_SQUARE 10
#define WAVEFORM_BANDLIMIT_PULSE 11
#define WAVEFORM_ARBITRARY 12
#define WAVEFORM_SAWTOOTH2 30
#define WAVEFORM_SILENT 19
// LFO1 (OSC)
#define WAVEFORM_ARBITRARY1 20
#define WAVEFORM_SAMPLE_HOLD1 21
#define PWM_WAVEFORM_SINE 26
#define PWM_WAVEFORM_TRIANGLE 27
#define PWM_WAVEFORM_SAWTOOTH 28
#define PWM_WAVEFORM_SQUARE 29
const uint8_t wav_res_vowel_data[] = {
27, 40, 89, 15, 13, 1, 0, 18,
51, 62, 13, 12, 6, 0, 15, 69,
93, 14, 12, 7, 0, 10, 84, 110,
13, 10, 8, 0, 23, 44, 87, 15,
12, 1, 0, 13, 29, 80, 13, 8,
0, 0, 6, 46, 81, 12, 3, 0,
0, 9, 51, 95, 15, 3, 0, 3,
6, 73, 99, 7, 3, 14, 9,
};
const int16_t wav_formant_square[] = {
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
};
const int16_t wav_formant_sine[] = {
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, 2, 2, 3, 3, 4, 5, 6,
7, 8, 10, 12, 14, 17, 20, 24,
0, 3, 4, 5, 6, 7, 9, 10,
12, 15, 18, 21, 26, 31, 37, 45,
0, 4, 5, 6, 8, 9, 11, 13,
16, 19, 23, 28, 34, 40, 49, 58,
0, 5, 6, 7, 8, 10, 12, 15,
17, 21, 25, 30, 36, 44, 53, 63,
0, 4, 5, 6, 8, 9, 11, 13,
16, 19, 23, 28, 34, 40, 49, 58,
0, 3, 4, 5, 6, 7, 9, 10,
12, 15, 18, 21, 26, 31, 37, 45,
0, 2, 2, 3, 3, 4, 5, 6,
7, 8, 10, 12, 14, 17, 20, 24,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, -2, -2, -3, -3, -4, -5, -6,
-7, -8, -10, -12, -14, -17, -20, -24,
0, -3, -4, -5, -6, -7, -9, -10,
-12, -15, -18, -21, -26, -31, -37, -45,
0, -4, -5, -6, -8, -9, -11, -13,
-16, -19, -23, -28, -34, -40, -49, -58,
0, -5, -6, -7, -8, -10, -12, -15,
-17, -21, -25, -30, -36, -44, -53, -63,
0, -4, -5, -6, -8, -9, -11, -13,
-16, -19, -23, -28, -34, -40, -49, -58,
0, -3, -4, -5, -6, -7, -9, -10,
-12, -15, -18, -21, -26, -31, -37, -45,
0, -2, -2, -3, -3, -4, -5, -6,
-7, -8, -10, -12, -14, -17, -20, -24,
};
struct PhonemeDefinition {
uint8_t formant_frequency[3];
uint8_t formant_amplitude[3];
};
static const PhonemeDefinition vowels_data[9] = {
{ { 27, 40, 89 }, { 15, 13, 1 } },
{ { 18, 51, 62 }, { 13, 12, 6 } },
{ { 15, 69, 93 }, { 14, 12, 7 } },
{ { 10, 84, 110 }, { 13, 10, 8 } },
{ { 23, 44, 87 }, { 15, 12, 1 } },
{ { 13, 29, 80 }, { 13, 8, 0 } },
{ { 6, 46, 81 }, { 12, 3, 0 } },
{ { 9, 51, 95 }, { 15, 3, 0 } },
{ { 6, 73, 99 }, { 7, 3, 14 } }
};
static const PhonemeDefinition consonant_data[8] = {
{ { 6, 54, 121 }, { 9, 9, 0 } },
{ { 18, 50, 51 }, { 12, 10, 5 } },
{ { 11, 24, 70 }, { 13, 8, 0 } },
{ { 15, 69, 74 }, { 14, 12, 7 } },
{ { 16, 37, 111 }, { 14, 8, 1 } },
{ { 18, 51, 62 }, { 14, 12, 6 } },
{ { 6, 26, 81 }, { 5, 5, 5 } },
{ { 6, 73, 99 }, { 7, 10, 14 } },
};
typedef struct step_state
{
int offset ;
bool positive ;
} step_state ;
class BandLimitedWaveform
{
public:
BandLimitedWaveform (void) ;
int16_t generate_sawtooth (uint32_t new_phase, int i) ;
int16_t generate_square (uint32_t new_phase, int i) ;
int16_t generate_pulse (uint32_t new_phase, uint32_t pulse_width, int i) ;
void init_sawtooth (uint32_t freq_word) ;
void init_square (uint32_t freq_word) ;
void init_pulse (uint32_t freq_word, uint32_t pulse_width) ;
private:
int32_t lookup (int offset) ;
void insert_step (int offset, bool rising, int i) ;
int32_t process_step (int i) ;
int32_t process_active_steps (uint32_t new_phase) ;
int32_t process_active_steps_saw (uint32_t new_phase) ;
int32_t process_active_steps_pulse (uint32_t new_phase, uint32_t pulse_width) ;
void new_step_check_square (uint32_t new_phase, int i) ;
void new_step_check_pulse (uint32_t new_phase, uint32_t pulse_width, int i) ;
void new_step_check_saw (uint32_t new_phase, int i) ;
uint32_t phase_word ;
int32_t dc_offset ;
step_state states [32] ; // circular buffer of active steps
int newptr ; // buffer pointers into states, AND'd with PTRMASK to keep in buffer range.
int delptr ;
int32_t cyclic[16] ; // circular buffer of output samples
bool pulse_state ;
uint32_t sampled_width ; // pulse width is sampled once per waveform
};
class AudioSynthWaveformTS : public AudioStream
{
public:
AudioSynthWaveformTS(void) : AudioStream(0,NULL),
phase_accumulator(0), phase_increment(0), phase_offset(0),
magnitude(0), pulse_width(0x40000000),
arbdata(NULL), sample(0), tone_type(WAVEFORM_SINE),
tone_offset(0),syncFlag(0) {
}
void frequency(float freq) {
if (freq < 0.0) {
freq = 0.0;
} else if (freq > AUDIO_SAMPLE_RATE_EXACT / 2) {
freq = AUDIO_SAMPLE_RATE_EXACT / 2;
}
phase_increment = freq * (4294967296.0 / AUDIO_SAMPLE_RATE_EXACT);
if (phase_increment > 0x7FFE0000u) phase_increment = 0x7FFE0000;
}
void phase(float angle) {
if (angle < 0.0) {
angle = 0.0;
} else if (angle > 360.0) {
angle = angle - 360.0;
if (angle >= 360.0) return;
}
phase_offset = angle * (4294967296.0 / 360.0);
}
void sync() {
syncFlag = 1;
}
void amplitude(float n) { // 0 to 1.0
if (n < 0) {
n = 0;
} else if (n > 1.0) {
n = 1.0;
}
magnitude = n * 65536.0;
}
void offset(float n) {
if (n < -1.0) {
n = -1.0;
} else if (n > 1.0) {
n = 1.0;
}
tone_offset = n * 32767.0;
}
void pulseWidth(float n) { // 0.0 to 1.0
if (n < 0) {
n = 0;
} else if (n > 1.0) {
n = 1.0;
}
pulse_width = n * 4294967296.0;
}
void begin(short t_type) {
phase_offset = 0;
tone_type = t_type;
if (t_type == WAVEFORM_BANDLIMIT_SQUARE)
band_limit_waveform.init_square (phase_increment) ;
else if (t_type == WAVEFORM_BANDLIMIT_PULSE)
band_limit_waveform.init_pulse (phase_increment, pulse_width) ;
else if (t_type == WAVEFORM_BANDLIMIT_SAWTOOTH || t_type == WAVEFORM_BANDLIMIT_SAWTOOTH_REVERSE)
band_limit_waveform.init_sawtooth (phase_increment) ;
}
void begin(float t_amp, float t_freq, short t_type) {
amplitude(t_amp);
frequency(t_freq);
phase_offset = 0;
begin (t_type);
}
void arbitraryWaveform(const int16_t *data, float maxFreq) {
arbdata = data;
}
virtual void update(void);
private:
uint32_t phase_accumulator;
uint32_t phase_increment;
uint32_t phase_offset;
int32_t magnitude;
uint32_t pulse_width;
const int16_t *arbdata;
int16_t sample; // for WAVEFORM_SAMPLE_HOLD
int16_t sample2; // for WAVEFORM_SAMPLE_HOLD and oneShot
short tone_type;
int16_t tone_offset;
int16_t syncFlag;
BandLimitedWaveform band_limit_waveform;
};
class AudioSynthWaveformModulatedTS : public AudioStream
{
public:
AudioSynthWaveformModulatedTS(void) : AudioStream(2, inputQueueArray),
phase_accumulator(0), phase_increment(0), modulation_factor(32768),
magnitude(0), arbdata(NULL), sample(0), tone_offset(0),
tone_type(WAVEFORM_SINE), modulation_type(0), syncFlag(0), osc_par_a(0),
osc_par_b(0)
{
}
void frequency(float freq) {
freq = freq / 2.0f; // only for tone_type Vowel
if (freq < 0.0) {
freq = 0.0;
} else if (freq > AUDIO_SAMPLE_RATE_EXACT / 2) {
freq = AUDIO_SAMPLE_RATE_EXACT / 2;
}
phase_increment = freq * (4294967296.0 / AUDIO_SAMPLE_RATE_EXACT);
//if (phase_increment > 0x7FFE0000u) phase_increment = 0x7FFE0000;
}
void amplitude(float n) { // 0 to 1.0
if (n < 0) {
n = 0;
} else if (n > 1.0) {
n = 1.0;
}
magnitude = n * 65536.0;
}
void sync() {
syncFlag = 1;
}
void offset(float n) {
if (n < -1.0) {
n = -1.0;
} else if (n > 1.0) {
n = 1.0;
}
tone_offset = n * 32767.0;
}
void begin(short t_type) {
tone_type = t_type;
if (t_type == WAVEFORM_BANDLIMIT_SQUARE)
band_limit_waveform.init_square (phase_increment) ;
else if (t_type == WAVEFORM_BANDLIMIT_PULSE)
band_limit_waveform.init_pulse (phase_increment, 0x80000000u) ;
else if (t_type == WAVEFORM_BANDLIMIT_SAWTOOTH || t_type == WAVEFORM_BANDLIMIT_SAWTOOTH_REVERSE)
band_limit_waveform.init_sawtooth (phase_increment) ;
}
void begin(float t_amp, float t_freq, short t_type) {
amplitude(t_amp);
frequency(t_freq);
begin (t_type) ;
}
void arbitraryWaveform(const int16_t *data, float maxFreq) {
arbdata = data;
}
void frequencyModulation(float octaves) {
if (octaves > 12.0) {
octaves = 12.0;
} else if (octaves < 0.1) {
octaves = 0.1;
}
modulation_factor = octaves * 4096.0;
modulation_type = 0;
}
void phaseModulation(float degrees) {
if (degrees > 9000.0) {
degrees = 9000.0;
} else if (degrees < 30.0) {
degrees = 30.0;
}
modulation_factor = degrees * (65536.0 / 180.0);
modulation_type = 1;
}
void parameter_a(uint8_t Osc_par_a) {
osc_par_a = Osc_par_a;
}
void parameter_b(uint8_t Osc_par_b) {
osc_par_b = Osc_par_b;
}
virtual void update(void);
private:
audio_block_t *inputQueueArray[2];
uint32_t phase_accumulator;
uint32_t phase_increment;
uint32_t modulation_factor;
int32_t magnitude;
const int16_t *arbdata;
uint32_t phasedata[AUDIO_BLOCK_SAMPLES];
int16_t sample; // for WAVEFORM_SAMPLE_HOLD
int16_t tone_offset;
uint8_t tone_type;
uint8_t modulation_type;
int16_t syncFlag;
uint32_t data_qs_phase[4];
uint32_t data_qs_phase_2[4];
uint8_t Osc_data_cr_decimate;
uint16_t Osc_data_cr_state;
uint16_t OscData_sec_phase;
BandLimitedWaveform band_limit_waveform;
uint8_t osc_par_a;
uint8_t osc_par_b;
uint8_t Osc_vw_update;
uint32_t state_vow_formant_phase[3];
uint32_t state_vow_formant_increment[3];
uint32_t state_vow_formant_amplitude[3];
boolean strike_;
uint16_t state_vow_consonant_frames;
uint32_t state_saw_phase[6];
uint32_t state_saw_lp;
uint32_t state_saw_bp;
uint16_t Osc_vw1_formant_increment[3];
uint16_t Osc_vw1_formant_amplitude[3];
uint16_t Osc_vw1_formant_phase[3];
};synth_waveform.cpp
Code:
/* Audio Library for Teensy 3.X
* Copyright (c) 2018, Paul Stoffregen, paul@pjrc.com
*
* 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.
ElectroTechnique 2020
Added WAVEFORM_SILENT, syncFlag
*/
#include <Arduino.h>
#include "synth_waveform.h"
#include "arm_math.h"
#include "utility/dspinst.h"
#include "Entropy.h"
// uncomment for more accurate but more computationally expensive frequency modulation
//#define IMPROVE_EXPONENTIAL_ACCURACY
#define BASE_AMPLITUDE 0x6000 // 0x7fff won't work due to Gibb's phenomenon, so use 3/4 of full range.
void AudioSynthWaveformTS::update(void)
{
audio_block_t *block;
int16_t *bp, *end;
int32_t val1, val2;
int16_t val3;
int16_t magnitude15;
uint32_t i, ph, index, index2, scale;
const uint32_t inc = phase_increment;
uint32_t phaseX;
if(syncFlag == 1) {
phase_accumulator = 0;
phaseX = 0;
syncFlag = 0;
LFO1randomFlag = false;
LFO2randomFlag = false;
}
ph = phase_accumulator + phase_offset;
phaseX = phase_accumulator;
if (magnitude == 0) {
phase_accumulator += inc * AUDIO_BLOCK_SAMPLES;
return;
}
block = allocate();
if (!block) {
phase_accumulator += inc * AUDIO_BLOCK_SAMPLES;
return;
}
bp = block->data;
switch(tone_type) {
// PWM SINE
case PWM_WAVEFORM_SINE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
index = ph >> 24;
val1 = AudioWaveformSine[index];
val2 = AudioWaveformSine[index+1];
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0x10000 - scale;
*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
ph += inc;
}
break;
case PWM_WAVEFORM_TRIANGLE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
uint32_t phtop = ph >> 30;
if (phtop == 1 || phtop == 2) {
*bp++ = ((0xFFFF - (ph >> 15)) * magnitude) >> 16;
} else {
*bp++ = (((int32_t)ph >> 15) * magnitude) >> 16;
}
ph += inc;
}
break;
// PWM SAWTOOTH (LFO offset 0.0f)
case PWM_WAVEFORM_SAWTOOTH: // normal sawtooth and inv sawtooth
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
*bp++ = signed_multiply_32x16t(magnitude, ph >> 1);
ph += inc;
}
break;
// PWM SQUARE (LFO offset 1.0)
case PWM_WAVEFORM_SQUARE:
magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
//if (ph & 0x80000000) {
if (ph & 0x80000000) {
*bp++ = -magnitude15;
} else {
*bp++ = magnitude15;
}
ph += inc;
}
break;
case WAVEFORM_ARBITRARY1:
if (!arbdata) {
release(block);
phase_accumulator += inc * AUDIO_BLOCK_SAMPLES;
return;
}
// len = 256
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
index = ph >> 24;
index2 = index + 1;
if (LFO1mode == 0) {
if (index2 >= 256) index2 = 0; // loop
}
else {
if (index2 >= 256) index2 = 255; // one shot
}
val1 = *(arbdata + index);
val2 = *(arbdata + index2);
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0x10000 - scale;
*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
uint32_t ph_old = ph;
uint32_t phaseX_old = phaseX;
if (LFO1phase == 0) { // Shape normal
ph += inc;
phaseX += inc;
}
else {
ph -= inc;
phaseX -= inc; // Shape inverse
}
if (lfo1oneShoot == true && phaseX < inc) {
ph = ph_old;
phaseX = phaseX_old;
}
}
break;
case WAVEFORM_SAMPLE_HOLD1:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
uint32_t newph = ph + inc;
if (lfo1oneShoot == true && LFO1randomFlag == false) { // one shot
sample = random(magnitude) - (magnitude >> 1);
sample2 = sample;
LFO1randomFlag = true;
}
else if (newph < ph && lfo1oneShoot == false) {
sample = random(magnitude) - (magnitude >> 1);
}
else if (lfo1oneShoot == true && LFO1randomFlag == true) {
sample = sample2;
}
*bp++ = sample;
ph = newph;
}
break;
phase_accumulator = ph - phase_offset;
if (tone_offset) {
bp = block->data;
end = bp + AUDIO_BLOCK_SAMPLES;
do {
val1 = *bp;
*bp++ = signed_saturate_rshift(val1 + tone_offset, 16, 0);
} while (bp < end);
}
transmit(block, 0);
release(block);
}
//--------------------------------------------------------------------------------
void AudioSynthWaveformModulatedTS::update(void)
{
audio_block_t *block, *moddata, *shapedata;
int16_t *bp, *end;
int32_t val1, val2;
int16_t magnitude15;
uint32_t i, ph, index, index2, scale, priorphase;
const uint32_t inc = phase_increment;
moddata = receiveReadOnly(0);
shapedata = receiveReadOnly(1);
if(syncFlag==1){
phase_accumulator = 0;
syncFlag = 0;
}
// Pre-compute the phase angle for every output sample of this update
ph = phase_accumulator;
priorphase = phasedata[AUDIO_BLOCK_SAMPLES-1];
if (moddata && modulation_type == 0) {
// Frequency Modulation
bp = moddata->data;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
int32_t n = (*bp++) * modulation_factor; // n is # of octaves to mod
int32_t ipart = n >> 27; // 4 integer bits
n &= 0x7FFFFFF; // 27 fractional bits
#ifdef IMPROVE_EXPONENTIAL_ACCURACY
// exp2 polynomial suggested by Stefan Stenzel on "music-dsp"
// mail list, Wed, 3 Sep 2014 10:08:55 +0200
int32_t x = n << 3;
n = multiply_accumulate_32x32_rshift32_rounded(5368709 12, x, 1494202713);
int32_t sq = multiply_32x32_rshift32_rounded(x, x);
n = multiply_accumulate_32x32_rshift32_rounded(n, sq, 1934101615);
n = n + (multiply_32x32_rshift32_rounded(sq,
multiply_32x32_rshift32_rounded(x, 1358044250)) << 1);
n = n << 1;
#else
// exp2 algorithm by Laurent de Soras
// https://www.musicdsp.org/en/latest/Other/106-fast-exp2-approximation.html
n = (n + 134217728) << 3;
n = multiply_32x32_rshift32_rounded(n, n);
n = multiply_32x32_rshift32_rounded(n, 715827883) << 3;
n = n + 715827882;
#endif
uint32_t scale = n >> (14 - ipart);
uint64_t phstep = (uint64_t)inc * scale;
uint32_t phstep_msw = phstep >> 32;
if (phstep_msw < 0x7FFE) {
ph += phstep >> 16;
} else {
ph += 0x7FFE0000;
}
phasedata[i] = ph;
}
release(moddata);
} else if (moddata) {
// Phase Modulation
bp = moddata->data;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
// more than +/- 180 deg shift by 32 bit overflow of "n"
uint32_t n = (uint16_t)(*bp++) * modulation_factor;
phasedata[i] = ph + n;
ph += inc;
}
release(moddata);
} else {
// No Modulation Input
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
phasedata[i] = ph;
ph += inc;
}
}
phase_accumulator = ph;
//Amplitude is always 1 on TSynth when oscillator is sounding
//magnitude must be set to zero, otherwise digital noise comes through
if(tone_type == WAVEFORM_SILENT){
magnitude = 0;
}else{
magnitude = 65536.0;
}
// If the amplitude is zero, no output, but phase still increments properly
if (magnitude == 0) {
if (shapedata) release(shapedata);
return;
}
block = allocate();
if (!block) {
if (shapedata) release(shapedata);
return;
}
bp = block->data;
// Now generate the output samples using the pre-computed phase angles
switch(tone_type) {
case WAVEFORM_SINE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[i];
index = ph >> 24;
val1 = AudioWaveformSine[index];
val2 = AudioWaveformSine[index+1];
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0x10000 - scale;
*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
}
break;
case WAVEFORM_ARBITRARY:
if (!arbdata) {
release(block);
if (shapedata) release(shapedata);
return;
}
// len = 256
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[i];
index = ph >> 24;
index2 = index + 1;
if (index2 >= 256) index2 = 0;
val1 = *(arbdata + index);
val2 = *(arbdata + index2);
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0x10000 - scale;
*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
}
break;
case WAVEFORM_PULSE:
if (shapedata) {
magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
uint32_t width = ((shapedata->data[i] + 0x8000) & 0xFFFF) << 16;
if (phasedata[i] < width) {
*bp++ = magnitude15;
} else {
*bp++ = -magnitude15;
}
}
break;
} // else fall through to orginary square without shape modulation
case WAVEFORM_SQUARE:
magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
if (phasedata[i] & 0x80000000) {
*bp++ = -magnitude15;
} else {
*bp++ = magnitude15;
}
}
break;
case WAVEFORM_BANDLIMIT_PULSE:
if (shapedata)
{
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++)
{
uint32_t width = ((shapedata->data[i] + 0x8000) & 0xFFFF) << 16;
int32_t val = band_limit_waveform.generate_pulse (phasedata[i], width, i) ;
*bp++ = (int16_t) ((val * magnitude) >> 16) ;
}
break;
} // else fall through to orginary square without shape modulation
case WAVEFORM_BANDLIMIT_SQUARE:
for (i = 0 ; i < AUDIO_BLOCK_SAMPLES ; i++)
{
int32_t val = band_limit_waveform.generate_square (phasedata[i], i) ;
*bp++ = (int16_t) ((val * magnitude) >> 16);
}
break;
// Vowel ------------------------------------------------
case WAVEFORM_Vowel
int16_t parameter_[2];
uint16_t balance;
uint8_t vowel_index;
uint16_t formant_shift;
parameter_[0] = osc_par_a << 8; // convert parameter into int16 value
parameter_[1] = osc_par_b << 8; // convert parameter into int16 value
vowel_index = parameter_[0] >> 12;
balance = parameter_[0] & 0x0fff;
formant_shift = (400 + (parameter_[1] >> 5));
if (strike_) {
strike_ = false;
state_vow_consonant_frames = 160;
for (size_t i = 0; i < 3; i++) {
state_vow_formant_increment[i] = static_cast<uint32_t>(consonant_data[index].formant_frequency[i]) *
0x1000 * formant_shift;
state_vow_formant_amplitude[i] = consonant_data[index].formant_amplitude[i];
}
}
if (state_vow_consonant_frames) {
--state_vow_consonant_frames;
} else {
for (size_t i = 0; i < 3; ++i) {
state_vow_formant_increment[i] = (vowels_data[vowel_index].formant_frequency[i] *
(0x1000 - balance) + vowels_data[vowel_index + 1].formant_frequency[i] * balance) * formant_shift;
state_vow_formant_amplitude[i] = (vowels_data[vowel_index].formant_amplitude[i] *
(0x1000 - balance) + vowels_data[vowel_index + 1].formant_amplitude[i] * balance) >> 12;
}
}
uint32_t phase_;
for (uint8_t i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
phase_ = phasedata[i];
size_t phaselet;
int16_t sample = 0;
state_vow_formant_phase[0] += state_vow_formant_increment[0];
phaselet = (state_vow_formant_phase[0] >> 24) & 0xf0;
sample += wav_formant_sine[phaselet | state_vow_formant_amplitude[0]];
state_vow_formant_phase[1] += state_vow_formant_increment[1];
phaselet = (state_vow_formant_phase[1] >> 24) & 0xf0;
sample += wav_formant_sine[phaselet | state_vow_formant_amplitude[1]];
state_vow_formant_phase[2] += state_vow_formant_increment[2];
phaselet = (state_vow_formant_phase[2] >> 24) & 0xf0;
sample += wav_formant_square[phaselet | state_vow_formant_amplitude[2]];
sample *= 255 - (phase_ >> 24);
if (phase_ < phase_increment_) {
state_vow_formant_phase[0] = 0;
state_vow_formant_phase[1] = 0;
state_vow_formant_phase[2] = 0;
sample = 0;
}
// amplify and soft clip Sound
int32_t x = (sample * 1.35f);
int32_t Threshold = 32768;
float sample_x = (1.0f / Threshold) * x; // convert input into float
if (sample_x > 1.0f) {
sample_x = 1.0f;
}
if (sample_x < - 1.0f) {
sample_x = -1.0f;
}
sample_x = 1.49f * sample_x - 0.5f * sample_x * sample_x * sample_x;
x = sample_x * Threshold; // convert float into int
*bp++ = x;
}
break;
case WAVEFORM_SAWTOOTH_REVERSE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
*bp++ = signed_multiply_32x16t(0xFFFFFFFFu - magnitude, phasedata[i]);
}
break;
case WAVEFORM_BANDLIMIT_SAWTOOTH:
case WAVEFORM_BANDLIMIT_SAWTOOTH_REVERSE:
for (i = 0 ; i < AUDIO_BLOCK_SAMPLES ; i++)
{
int16_t val = band_limit_waveform.generate_sawtooth (phasedata[i], i) ;
val = (int16_t) ((val * magnitude) >> 16) ;
*bp++ = tone_type == WAVEFORM_BANDLIMIT_SAWTOOTH_REVERSE ? (int16_t) -val : (int16_t) +val ;
}
break;
case WAVEFORM_TRIANGLE_VARIABLE:
if (shapedata) {
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
uint32_t width = (shapedata->data[i] + 0x8000) & 0xFFFF;
uint32_t rise = 0xFFFFFFFF / width;
uint32_t fall = 0xFFFFFFFF / (0xFFFF - width);
uint32_t halfwidth = width << 15;
uint32_t n;
ph = phasedata[i];
if (ph < halfwidth) {
n = (ph >> 16) * rise;
*bp++ = ((n >> 16) * magnitude) >> 16;
} else if (ph < 0xFFFFFFFF - halfwidth) {
n = 0x7FFFFFFF - (((ph - halfwidth) >> 16) * fall);
*bp++ = (((int32_t)n >> 16) * magnitude) >> 16;
} else {
n = ((ph + halfwidth) >> 16) * rise + 0x80000000;
*bp++ = (((int32_t)n >> 16) * magnitude) >> 16;
}
ph += inc;
}
break;
} // else fall through to orginary triangle without shape modulation
case WAVEFORM_TRIANGLE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[i];
uint32_t phtop = ph >> 30;
if (phtop == 1 || phtop == 2) {
*bp++ = ((0xFFFF - (ph >> 15)) * magnitude) >> 16;
} else {
*bp++ = (((int32_t)ph >> 15) * magnitude) >> 16;
}
}
break;
case WAVEFORM_SAMPLE_HOLD:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[i];
if (ph < priorphase) { // does not work for phase modulation
sample = random(magnitude) - (magnitude >> 1);
}
priorphase = ph;
*bp++ = sample;
}
break;
}
if (tone_offset) {
bp = block->data;
end = bp + AUDIO_BLOCK_SAMPLES;
do {
val1 = *bp;
*bp++ = signed_saturate_rshift(val1 + tone_offset, 16, 0);
} while (bp < end);
}
if (shapedata) release(shapedata);
transmit(block, 0);
release(block);
}
// BandLimitedWaveform
#define SUPPORT_SHIFT 4
#define SUPPORT (1 << SUPPORT_SHIFT)
#define PTRMASK ((2 << SUPPORT_SHIFT) - 1)
#define SCALE 16
#define SCALE_MASK (SCALE-1)
#define N (SCALE * SUPPORT * 2)
#define GUARD_BITS 8
#define GUARD (1 << GUARD_BITS)
#define HALF_GUARD (1 << (GUARD_BITS-1))
#define DEG180 0x80000000u
#define PHASE_SCALE (0x100000000L / (2 * BASE_AMPLITUDE))
extern "C"
{
extern const int16_t bandlimit_step_table [258] ;
}
int32_t BandLimitedWaveform::lookup (int offset)
{
int off = offset >> GUARD_BITS ;
int frac = offset & (GUARD-1) ;
int32_t a, b ;
if (off < N/2) // handle odd symmetry by reflecting table
{
a = bandlimit_step_table [off+1] ;
b = bandlimit_step_table [off+2] ;
}
else
{
a = - bandlimit_step_table [N-off] ;
b = - bandlimit_step_table [N-off-1] ;
}
return BASE_AMPLITUDE + ((frac * b + (GUARD - frac) * a + HALF_GUARD) >> GUARD_BITS) ; // interpolated
}
// create a new step, apply its past waveform into the cyclic sample buffer
// and add a step_state object into active list so it can be added for the future samples
void BandLimitedWaveform::insert_step (int offset, bool rising, int i)
{
while (offset <= (N/2-SCALE)<<GUARD_BITS)
{
if (offset >= 0)
cyclic [i & 15] += rising ? lookup (offset) : -lookup (offset) ;
offset += SCALE<<GUARD_BITS ;
i ++ ;
}
states[newptr].offset = offset ;
states[newptr].positive = rising ;
newptr = (newptr+1) & PTRMASK ;
}
// generate value for current sample from one active step, checking for the
// dc_offset adjustment at the end of the table.
int32_t BandLimitedWaveform::process_step (int i)
{
int off = states[i].offset ;
bool positive = states[i].positive ;
int32_t entry = lookup (off) ;
off += SCALE<<GUARD_BITS ;
states[i].offset = off ; // update offset in table for next sample
if (off >= N<<GUARD_BITS) // at end of step table we alter dc_offset to extend the step into future
dc_offset += positive ? 2*BASE_AMPLITUDE : -2*BASE_AMPLITUDE ;
return positive ? entry : -entry ;
}
// process all active steps for current sample, basically generating the waveform portion
// due only to steps
// square waves use this directly.
int32_t BandLimitedWaveform::process_active_steps (uint32_t new_phase)
{
int32_t sample = dc_offset ;
int step_count = (newptr - delptr) & PTRMASK ;
if (step_count > 0) // for any steps in-flight we sum in table entry and update its state
{
int i = newptr ;
do
{
i = (i-1) & PTRMASK ;
sample += process_step (i) ;
} while (i != delptr) ;
if (states[delptr].offset >= N<<GUARD_BITS) // remove any finished entries from the buffer.
{
delptr = (delptr+1) & PTRMASK ;
// can be upto two steps per sample now for pulses
if (newptr != delptr && states[delptr].offset >= N<<GUARD_BITS)
delptr = (delptr+1) & PTRMASK ;
}
}
return sample ;
}
// for sawtooth need to add in the slope and compensate for all the steps being one way
int32_t BandLimitedWaveform::process_active_steps_saw (uint32_t new_phase)
{
int32_t sample = process_active_steps (new_phase) ;
sample += (int16_t) ((((uint64_t)phase_word * (2*BASE_AMPLITUDE)) >> 32) - BASE_AMPLITUDE) ; // generate the sloped part of the wave
if (new_phase < DEG180 && phase_word >= DEG180) // detect wrap around, correct dc offset
dc_offset += 2*BASE_AMPLITUDE ;
return sample ;
}
// for pulse need to adjust the baseline according to the pulse width to cancel the DC component.
int32_t BandLimitedWaveform::process_active_steps_pulse (uint32_t new_phase, uint32_t pulse_width)
{
int32_t sample = process_active_steps (new_phase) ;
return sample + BASE_AMPLITUDE/2 - pulse_width / (0x80000000u / BASE_AMPLITUDE) ; // correct DC offset for duty cycle
}
// Check for new steps using the phase update for the current sample for a square wave
void BandLimitedWaveform::new_step_check_square (uint32_t new_phase, int i)
{
if (new_phase >= DEG180 && phase_word < DEG180) // detect falling step
{
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (sampled_width - phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
if (pulse_state) // guard against two falling steps in a row (if pulse width changing for instance)
{
insert_step (- offset, false, i) ;
pulse_state = false ;
}
}
else if (new_phase < DEG180 && phase_word >= DEG180) // detect wrap around, rising step
{
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (- phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
if (!pulse_state) // guard against two rising steps in a row (if pulse width changing for instance)
{
insert_step (- offset, true, i) ;
pulse_state = true ;
}
}
}
// Checking for new steps for pulse waveform has to deal with changing frequency and pulse width and
// not letting a pulse glitch out of existence as these change across a single period of the waveform
// now we detect the rising edge just like for a square wave and use that to sample the pulse width
// parameter, which then has to be checked against the instantaneous frequency every sample.
void BandLimitedWaveform::new_step_check_pulse (uint32_t new_phase, uint32_t pulse_width, int i)
{
if (pulse_state && phase_word < sampled_width && (new_phase >= sampled_width || new_phase < phase_word)) // falling edge
{
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (sampled_width - phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
insert_step (- offset, false, i) ;
pulse_state = false ;
}
if ((!pulse_state) && phase_word >= DEG180 && new_phase < DEG180) // detect wrap around, rising step
{
// sample the pulse width value so its not changing under our feet later in cycle due to modulation
sampled_width = pulse_width ;
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (- phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
insert_step (- offset, true, i) ;
pulse_state = true ;
if (pulse_state && new_phase >= sampled_width) // detect falling step directly after a rising edge
//if (new_phase - sampled_width < DEG180) // detect falling step directly after a rising edge
{
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (sampled_width - phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
insert_step (- offset, false, i) ;
pulse_state = false ;
}
}
}
// new steps for sawtooth are at 180 degree point, always falling.
void BandLimitedWaveform::new_step_check_saw (uint32_t new_phase, int i)
{
if (new_phase >= DEG180 && phase_word < DEG180) // detect falling step
{
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (DEG180 - phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
insert_step (- offset, false, i) ;
}
}
// the generation function pushd new sample into cyclic buffer, having taken out the oldest entry
// to return. The output is thus 16 samples behind, which allows the non-casual step function to
// work in real time.
int16_t BandLimitedWaveform::generate_sawtooth (uint32_t new_phase, int i)
{
new_step_check_saw (new_phase, i) ;
int32_t val = process_active_steps_saw (new_phase) ;
int16_t sample = (int16_t) cyclic [i&15] ;
cyclic [i&15] = val ;
phase_word = new_phase ;
return sample ;
}
int16_t BandLimitedWaveform::generate_square (uint32_t new_phase, int i)
{
new_step_check_square (new_phase, i) ;
int32_t val = process_active_steps (new_phase) ;
int16_t sample = (int16_t) cyclic [i&15] ;
cyclic [i&15] = val ;
phase_word = new_phase ;
return sample ;
}
int16_t BandLimitedWaveform::generate_pulse (uint32_t new_phase, uint32_t pulse_width, int i)
{
new_step_check_pulse (new_phase, pulse_width, i) ;
int32_t val = process_active_steps_pulse (new_phase, pulse_width) ;
int32_t sample = cyclic [i&15] ;
cyclic [i&15] = val ;
phase_word = new_phase ;
return (int16_t) ((sample >> 1) - (sample >> 5)) ; // scale down to avoid overflow on narrow pulses, where the DC shift is big
}
void BandLimitedWaveform::init_sawtooth (uint32_t freq_word)
{
phase_word = 0 ;
newptr = 0 ;
delptr = 0 ;
for (int i = 0 ; i < 2*SUPPORT ; i++)
phase_word -= freq_word ;
dc_offset = phase_word < DEG180 ? BASE_AMPLITUDE : -BASE_AMPLITUDE ;
for (int i = 0 ; i < 2*SUPPORT ; i++)
{
uint32_t new_phase = phase_word + freq_word ;
new_step_check_saw (new_phase, i) ;
cyclic [i & 15] = (int16_t) process_active_steps_saw (new_phase) ;
phase_word = new_phase ;
}
}
void BandLimitedWaveform::init_square (uint32_t freq_word)
{
init_pulse (freq_word, DEG180) ;
}
void BandLimitedWaveform::init_pulse (uint32_t freq_word, uint32_t pulse_width)
{
phase_word = 0 ;
sampled_width = pulse_width ;
newptr = 0 ;
delptr = 0 ;
for (int i = 0 ; i < 2*SUPPORT ; i++)
phase_word -= freq_word ;
if (phase_word < pulse_width)
{
dc_offset = BASE_AMPLITUDE ;
pulse_state = true ;
}
else
{
dc_offset = -BASE_AMPLITUDE ;
pulse_state = false ;
}
for (int i = 0 ; i < 2*SUPPORT ; i++)
{
uint32_t new_phase = phase_word + freq_word ;
new_step_check_pulse (new_phase, pulse_width, i) ;
cyclic [i & 15] = (int16_t) process_active_steps_pulse (new_phase, pulse_width) ;
phase_word = new_phase ;
}
}
BandLimitedWaveform::BandLimitedWaveform()
{
newptr = 0 ;
delptr = 0 ;
dc_offset = BASE_AMPLITUDE ;
phase_word = 0 ;
}
Last edited:
Rolfdegen
Well-known member
Result into vowel code..
synth_waveform.h
synth_waveform.cpp
Code:
if (phase_ < phaseOld_) {
state_vow_formant_phase[0] = 0;
state_vow_formant_phase[1] = 0;
state_vow_formant_phase[2] = 0;
sample = 0;
}
phaseOld_ = phase_;synth_waveform.h
Code:
/* Audio Library for Teensy 3.X
* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
*
* 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.
ElectroTechnique 2020
Added WAVEFORM_SILENT, sync()
*/
#ifndef synth_waveform_h_
#define synth_waveform_h_
#include <Arduino.h>
#include "AudioStream.h"
#include "arm_math.h"
// waveforms.c
extern "C" {
extern const int16_t AudioWaveformSine[257];
}
//extern int8_t Lfo3Modoutput;
// Oscillator
#define WAVEFORM_SINE 0
#define WAVEFORM_TRIANGLE 1
#define WAVEFORM_SAWTOOTH 3
#define WAVEFORM_SQUARE 2
#define WAVEFORM_PULSE 4
#define WAVEFORM_SAWTOOTH_REVERSE 5
#define WAVEFORM_SAMPLE_HOLD 6
#define WAVEFORM_TRIANGLE_VARIABLE 7
#define WAVEFORM_BANDLIMIT_SAWTOOTH_REVERSE 8
#define WAVEFORM_BANDLIMIT_SAWTOOTH 9
#define WAVEFORM_BANDLIMIT_SQUARE 10
#define WAVEFORM_BANDLIMIT_PULSE 11
#define WAVEFORM_ARBITRARY 12
#define WAVEFORM_SAWTOOTH2 30
#define WAVEFORM_SILENT 19
// LFO and PWM waveforms
#define LFO_WAVEFORM_ARBITRARY 20
#define LFO_WAVEFORM_SAMPLE_HOLD 21
#define PWM_WAVEFORM_SINE 26
#define PWM_WAVEFORM_TRIANGLE 27
#define PWM_WAVEFORM_SAWTOOTH 28
#define PWM_WAVEFORM_SQUARE 29
/*
const uint8_t wav_res_sine[] {
2, 2, 2, 3, 2, 3, 3, 4,
5, 4, 7, 5, 9, 7, 10, 11,
11, 13, 13, 17, 16, 18, 21, 21,
23, 25, 27, 28, 32, 31, 36, 36,
39, 41, 43, 46, 48, 51, 53, 55,
57, 62, 63, 65, 70, 70, 75, 76,
81, 82, 85, 89, 92, 94, 97, 100,
104, 107, 109, 112, 116, 119, 122, 124,
129, 130, 135, 137, 140, 144, 147, 148,
154, 155, 158, 163, 163, 169, 169, 174,
177, 178, 182, 185, 187, 191, 192, 195,
199, 200, 203, 205, 209, 210, 212, 216,
216, 220, 221, 223, 226, 227, 230, 230,
233, 235, 235, 239, 238, 241, 242, 243,
245, 245, 246, 249, 247, 251, 249, 252,
251, 252, 253, 253, 253, 254, 254, 254,
253, 255, 254, 253, 253, 254, 252, 253,
251, 251, 250, 250, 248, 248, 246, 246,
245, 242, 243, 240, 239, 238, 236, 234,
233, 231, 230, 226, 226, 224, 221, 219,
217, 215, 212, 211, 208, 206, 202, 201,
198, 196, 192, 190, 188, 184, 182, 179,
177, 173, 170, 168, 164, 162, 158, 156,
153, 149, 147, 143, 141, 136, 135, 131,
128, 125, 122, 118, 116, 113, 109, 106,
104, 101, 96, 95, 92, 88, 85, 83,
80, 77, 74, 71, 69, 66, 63, 61,
57, 56, 53, 50, 49, 45, 43, 42,
38, 37, 35, 32, 31, 29, 26, 26,
22, 22, 20, 19, 16, 16, 13, 14,
11, 10, 10, 8, 8, 6, 6, 5,
4, 5, 2, 4, 2, 2, 2, 3,
1
};
*/
const int8_t wav_res_formant_square[256] = {
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16
};
const int8_t wav_res_formant_sine[256] = {
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, 2, 2, 3, 3, 4, 5, 6,
7, 8, 10, 12, 14, 17, 20, 24,
0, 3, 4, 5, 6, 7, 9, 10,
12, 15, 18, 21, 26, 31, 37, 45,
0, 4, 5, 6, 8, 9, 11, 13,
16, 19, 23, 28, 34, 40, 49, 58,
0, 5, 6, 7, 8, 10, 12, 15,
17, 21, 25, 30, 36, 44, 53, 63,
0, 4, 5, 6, 8, 9, 11, 13,
16, 19, 23, 28, 34, 40, 49, 58,
0, 3, 4, 5, 6, 7, 9, 10,
12, 15, 18, 21, 26, 31, 37, 45,
0, 2, 2, 3, 3, 4, 5, 6,
7, 8, 10, 12, 14, 17, 20, 24,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, -2, -2, -3, -3, -4, -5, -6,
-7, -8, -10, -12, -14, -17, -20, -24,
0, -3, -4, -5, -6, -7, -9, -10,
-12, -15, -18, -21, -26, -31, -37, -45,
0, -4, -5, -6, -8, -9, -11, -13,
-16, -19, -23, -28, -34, -40, -49, -58,
0, -5, -6, -7, -8, -10, -12, -15,
-17, -21, -25, -30, -36, -44, -53, -63,
0, -4, -5, -6, -8, -9, -11, -13,
-16, -19, -23, -28, -34, -40, -49, -58,
0, -3, -4, -5, -6, -7, -9, -10,
-12, -15, -18, -21, -26, -31, -37, -45,
0, -2, -2, -3, -3, -4, -5, -6,
-7, -8, -10, -12, -14, -17, -20, -24
};
const uint16_t wav_res_sine16[] = {
0, 10, 39, 88, 157, 245, 352, 479,
625, 790, 974, 1178, 1400, 1641, 1901, 2179,
2475, 2790, 3122, 3472, 3840, 4225, 4627, 5045,
5481, 5932, 6400, 6884, 7382, 7897, 8426, 8969,
9527, 10098, 10684, 11282, 11893, 12517, 13153, 13800,
14459, 15129, 15809, 16500, 17200, 17909, 18628, 19355,
20090, 20832, 21582, 22338, 23100, 23869, 24642, 25421,
26203, 26990, 27780, 28574, 29369, 30167, 30966, 31767,
32568, 33369, 34170, 34969, 35768, 36565, 37359, 38151,
38940, 39724, 40505, 41281, 42052, 42818, 43577, 44330,
45076, 45815, 46546, 47268, 47982, 48687, 49383, 50068,
50743, 51408, 52061, 52703, 53332, 53950, 54555, 55147,
55725, 56290, 56840, 57377, 57898, 58405, 58896, 59372,
59831, 60275, 60702, 61112, 61506, 61882, 62241, 62582,
62906, 63211, 63499, 63767, 64018, 64249, 64462, 64656,
64831, 64987, 65123, 65240, 65338, 65416, 65475, 65514,
65534, 65534, 65514, 65475, 65416, 65338, 65240, 65123,
64987, 64831, 64656, 64462, 64249, 64018, 63767, 63499,
63211, 62906, 62582, 62241, 61882, 61506, 61112, 60702,
60275, 59831, 59372, 58896, 58405, 57898, 57377, 56840,
56290, 55725, 55147, 54555, 53950, 53332, 52703, 52061,
51408, 50743, 50068, 49383, 48687, 47982, 47268, 46546,
45815, 45076, 44330, 43577, 42818, 42052, 41281, 40505,
39724, 38940, 38151, 37359, 36565, 35768, 34969, 34170,
33369, 32568, 31767, 30966, 30167, 29369, 28574, 27780,
26990, 26203, 25421, 24642, 23869, 23100, 22338, 21582,
20832, 20090, 19355, 18628, 17909, 17200, 16500, 15809,
15129, 14459, 13800, 13153, 12517, 11893, 11282, 10684,
10098, 9527, 8969, 8426, 7897, 7382, 6884, 6400,
5932, 5481, 5045, 4627, 4225, 3840, 3472, 3122,
2790, 2475, 2179, 1901, 1641, 1400, 1178, 974,
790, 625, 479, 352, 245, 157, 88, 39,
10
};
const uint8_t wav_res_vowel_data[] = {
27, 40, 89, 15, 13, 1, 0, 18,
51, 62, 13, 12, 6, 0, 15, 69,
93, 14, 12, 7, 0, 10, 84, 110,
13, 10, 8, 0, 23, 44, 87, 15,
12, 1, 0, 13, 29, 80, 13, 8,
0, 0, 6, 46, 81, 12, 3, 0,
0, 9, 51, 95, 15, 3, 0, 3,
6, 73, 99, 7, 3, 14, 9,
};
const int16_t wav_formant_square[] = {
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, 1, 1, 2, 2, 3, 3, 4,
4, 5, 6, 8, 9, 11, 13, 16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
0, -1, -1, -2, -2, -3, -3, -4,
-4, -5, -6, -8, -9, -11, -13, -16,
};
const int16_t wav_formant_sine[] = {
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, 2, 2, 3, 3, 4, 5, 6,
7, 8, 10, 12, 14, 17, 20, 24,
0, 3, 4, 5, 6, 7, 9, 10,
12, 15, 18, 21, 26, 31, 37, 45,
0, 4, 5, 6, 8, 9, 11, 13,
16, 19, 23, 28, 34, 40, 49, 58,
0, 5, 6, 7, 8, 10, 12, 15,
17, 21, 25, 30, 36, 44, 53, 63,
0, 4, 5, 6, 8, 9, 11, 13,
16, 19, 23, 28, 34, 40, 49, 58,
0, 3, 4, 5, 6, 7, 9, 10,
12, 15, 18, 21, 26, 31, 37, 45,
0, 2, 2, 3, 3, 4, 5, 6,
7, 8, 10, 12, 14, 17, 20, 24,
0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0,
0, -2, -2, -3, -3, -4, -5, -6,
-7, -8, -10, -12, -14, -17, -20, -24,
0, -3, -4, -5, -6, -7, -9, -10,
-12, -15, -18, -21, -26, -31, -37, -45,
0, -4, -5, -6, -8, -9, -11, -13,
-16, -19, -23, -28, -34, -40, -49, -58,
0, -5, -6, -7, -8, -10, -12, -15,
-17, -21, -25, -30, -36, -44, -53, -63,
0, -4, -5, -6, -8, -9, -11, -13,
-16, -19, -23, -28, -34, -40, -49, -58,
0, -3, -4, -5, -6, -7, -9, -10,
-12, -15, -18, -21, -26, -31, -37, -45,
0, -2, -2, -3, -3, -4, -5, -6,
-7, -8, -10, -12, -14, -17, -20, -24,
};
struct PhonemeDefinition {
uint8_t formant_frequency[3];
uint8_t formant_amplitude[3];
};
static const PhonemeDefinition vowels_data[9] = {
{ { 27, 40, 89 }, { 15, 13, 1 } },
{ { 18, 51, 62 }, { 13, 12, 6 } },
{ { 15, 69, 93 }, { 14, 12, 7 } },
{ { 10, 84, 110 }, { 13, 10, 8 } },
{ { 23, 44, 87 }, { 15, 12, 1 } },
{ { 13, 29, 80 }, { 13, 8, 0 } },
{ { 6, 46, 81 }, { 12, 3, 0 } },
{ { 9, 51, 95 }, { 15, 3, 0 } },
{ { 6, 73, 99 }, { 7, 3, 14 } }
};
static const PhonemeDefinition consonant_data[8] = {
{ { 6, 54, 121 }, { 9, 9, 0 } },
{ { 18, 50, 51 }, { 12, 10, 5 } },
{ { 11, 24, 70 }, { 13, 8, 0 } },
{ { 15, 69, 74 }, { 14, 12, 7 } },
{ { 16, 37, 111 }, { 14, 8, 1 } },
{ { 18, 51, 62 }, { 14, 12, 6 } },
{ { 6, 26, 81 }, { 5, 5, 5 } },
{ { 6, 73, 99 }, { 7, 10, 14 } },
};
static const size_t kNumFormants = 5;
typedef struct step_state
{
int offset ;
bool positive ;
} step_state ;
class BandLimitedWaveform
{
public:
BandLimitedWaveform (void) ;
int16_t generate_sawtooth (uint32_t new_phase, int i) ;
int16_t generate_square (uint32_t new_phase, int i) ;
int16_t generate_pulse (uint32_t new_phase, uint32_t pulse_width, int i) ;
void init_sawtooth (uint32_t freq_word) ;
void init_square (uint32_t freq_word) ;
void init_pulse (uint32_t freq_word, uint32_t pulse_width) ;
private:
int32_t lookup (int offset) ;
void insert_step (int offset, bool rising, int i) ;
int32_t process_step (int i) ;
int32_t process_active_steps (uint32_t new_phase) ;
int32_t process_active_steps_saw (uint32_t new_phase) ;
int32_t process_active_steps_pulse (uint32_t new_phase, uint32_t pulse_width) ;
void new_step_check_square (uint32_t new_phase, int i) ;
void new_step_check_pulse (uint32_t new_phase, uint32_t pulse_width, int i) ;
void new_step_check_saw (uint32_t new_phase, int i) ;
uint32_t phase_word ;
int32_t dc_offset ;
step_state states [32] ; // circular buffer of active steps
int newptr ; // buffer pointers into states, AND'd with PTRMASK to keep in buffer range.
int delptr ;
int32_t cyclic[16] ; // circular buffer of output samples
bool pulse_state ;
uint32_t sampled_width ; // pulse width is sampled once per waveform
};
class AudioSynthWaveformTS : public AudioStream
{
public:
AudioSynthWaveformTS(void) : AudioStream(0,NULL),
phase_accumulator(0), phase_increment(0), phase_offset(0),
magnitude(0), pulse_width(0x40000000),
arbdata(NULL), sample(0), tone_type(WAVEFORM_SINE),
tone_offset(0),syncFlag(0) {
}
void frequency(float freq) {
if (freq < 0.0) {
freq = 0.0;
} else if (freq > AUDIO_SAMPLE_RATE_EXACT / 2) {
freq = AUDIO_SAMPLE_RATE_EXACT / 2;
}
phase_increment = freq * (4294967296.0 / AUDIO_SAMPLE_RATE_EXACT);
if (phase_increment > 0x7FFE0000u) phase_increment = 0x7FFE0000;
}
void phase(float angle) {
if (angle < 0.0) {
angle = 0.0;
} else if (angle > 360.0) {
angle = angle - 360.0;
if (angle >= 360.0) return;
}
phase_offset = angle * (4294967296.0 / 360.0);
}
void sync() {
syncFlag = 1;
}
void amplitude(float n) { // 0 to 1.0
if (n < 0) {
n = 0;
} else if (n > 1.0) {
n = 1.0;
}
magnitude = n * 65536.0;
}
void offset(float n) {
if (n < -1.0) {
n = -1.0;
} else if (n > 1.0) {
n = 1.0;
}
tone_offset = n * 32767.0;
}
void pulseWidth(float n) { // 0.0 to 1.0
if (n < 0) {
n = 0;
} else if (n > 1.0) {
n = 1.0;
}
pulse_width = n * 4294967296.0;
}
void begin(short t_type) {
phase_offset = 0;
tone_type = t_type;
if (t_type == WAVEFORM_BANDLIMIT_SQUARE)
band_limit_waveform.init_square (phase_increment) ;
else if (t_type == WAVEFORM_BANDLIMIT_PULSE)
band_limit_waveform.init_pulse (phase_increment, pulse_width) ;
else if (t_type == WAVEFORM_BANDLIMIT_SAWTOOTH || t_type == WAVEFORM_BANDLIMIT_SAWTOOTH_REVERSE)
band_limit_waveform.init_sawtooth (phase_increment) ;
}
void begin(float t_amp, float t_freq, short t_type) {
amplitude(t_amp);
frequency(t_freq);
phase_offset = 0;
begin (t_type);
}
void arbitraryWaveform(const int16_t *data, float maxFreq) {
arbdata = data;
}
void LFO_mode (uint8_t lfo_Mode) {
lfo_mode = lfo_Mode;
}
void LFO_phase (uint8_t lfo_Phase) { // LFO parameter: SYN
lfo_phase = lfo_Phase;
}
void LFO_oneShoot (boolean lfo_OneShoot) {
lfo_oneShoot = lfo_OneShoot;
}
virtual void update(void);
private:
uint32_t phase_accumulator;
uint32_t phase_increment;
uint32_t phase_offset;
int32_t magnitude;
uint32_t pulse_width;
const int16_t *arbdata;
int16_t sample; // for WAVEFORM_SAMPLE_HOLD
int16_t sample2; // for WAVEFORM_SAMPLE_HOLD and oneShot
short tone_type;
int16_t tone_offset;
int16_t syncFlag;
BandLimitedWaveform band_limit_waveform;
uint8_t lfo_mode;
uint8_t lfo_phase;
boolean lfo_oneShoot;
uint8_t lfo1ph;
boolean lfo_randomFlag;
};
class AudioSynthWaveformModulatedTS : public AudioStream
{
public:
AudioSynthWaveformModulatedTS(void) : AudioStream(2, inputQueueArray),
phase_accumulator(0), phase_increment(0), modulation_factor(32768),
magnitude(0), arbdata(NULL), sample(0), tone_offset(0),
tone_type(WAVEFORM_SINE), modulation_type(0), syncFlag(0), osc_par_a(0),
osc_par_b(0)
{
}
void frequency(float freq) {
//freq = freq / 2.0f; // only for tone_type Vowel
if (freq < 0.0) {
freq = 0.0;
} else if (freq > AUDIO_SAMPLE_RATE_EXACT / 2) {
freq = AUDIO_SAMPLE_RATE_EXACT / 2;
}
phase_increment = freq * (4294967296.0 / AUDIO_SAMPLE_RATE_EXACT);
if (phase_increment > 0x7FFE0000u) phase_increment = 0x7FFE0000;
}
void amplitude(float n) { // 0 to 1.0
if (n < 0) {
n = 0;
} else if (n > 1.0) {
n = 1.0;
}
magnitude = n * 65536.0;
}
void sync() {
syncFlag = 1;
}
void offset(float n) {
if (n < -1.0) {
n = -1.0;
} else if (n > 1.0) {
n = 1.0;
}
tone_offset = n * 32767.0;
}
void begin(short t_type) {
tone_type = t_type;
if (t_type == WAVEFORM_BANDLIMIT_SQUARE)
band_limit_waveform.init_square (phase_increment) ;
else if (t_type == WAVEFORM_BANDLIMIT_PULSE)
band_limit_waveform.init_pulse (phase_increment, 0x80000000u) ;
else if (t_type == WAVEFORM_BANDLIMIT_SAWTOOTH || t_type == WAVEFORM_BANDLIMIT_SAWTOOTH_REVERSE)
band_limit_waveform.init_sawtooth (phase_increment) ;
}
void begin(float t_amp, float t_freq, short t_type) {
amplitude(t_amp);
frequency(t_freq);
begin (t_type) ;
}
void arbitraryWaveform(const int16_t *data, float maxFreq) {
arbdata = data;
}
void frequencyModulation(float octaves) {
if (octaves > 12.0) {
octaves = 12.0;
} else if (octaves < 0.1) {
octaves = 0.1;
}
modulation_factor = octaves * 4096.0;
modulation_type = 0;
}
void phaseModulation(float degrees) {
if (degrees > 9000.0) {
degrees = 9000.0;
} else if (degrees < 30.0) {
degrees = 30.0;
}
modulation_factor = degrees * (65536.0 / 180.0);
modulation_type = 1;
}
void parameter_a(uint8_t Osc_par_a) {
osc_par_a = Osc_par_a;
}
void parameter_b(uint8_t Osc_par_b) {
osc_par_b = Osc_par_b;
}
virtual void update(void);
private:
audio_block_t *inputQueueArray[2];
uint32_t phase_accumulator;
uint32_t phase_increment;
uint32_t modulation_factor;
int32_t magnitude;
const int16_t *arbdata;
uint32_t phasedata[AUDIO_BLOCK_SAMPLES];
int16_t sample; // for WAVEFORM_SAMPLE_HOLD
int16_t tone_offset;
uint8_t tone_type;
uint8_t modulation_type;
int16_t syncFlag;
uint32_t data_qs_phase[4];
uint32_t data_qs_phase_2[4];
uint8_t Osc_data_cr_decimate;
uint16_t Osc_data_cr_state;
uint16_t OscData_sec_phase;
BandLimitedWaveform band_limit_waveform;
uint8_t osc_par_a;
uint8_t osc_par_b;
uint8_t Osc_vw_update;
uint32_t state_vow_formant_phase[3];
uint32_t state_vow_formant_increment[3];
uint32_t state_vow_formant_amplitude[3];
boolean strike_;
uint16_t state_vow_consonant_frames;
uint32_t state_saw_phase[6];
uint32_t state_saw_lp;
uint32_t state_saw_bp;
uint16_t Osc_vw1_formant_increment[3];
uint16_t Osc_vw1_formant_amplitude[3];
uint16_t Osc_vw1_formant_phase[3];
uint32_t phaseOld_= 0;
};
#endifsynth_waveform.cpp
Code:
/* Audio Library for Teensy 3.X
* Copyright (c) 2018, Paul Stoffregen, paul@pjrc.com
*
* 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.
ElectroTechnique 2020
Added WAVEFORM_SILENT, syncFlag
R.Degen 2023
Added Supersaw and Mutable instruments Braids digital oscillator
*/
#include <Arduino.h>
#include "synth_waveform.h"
#include "arm_math.h"
#include "utility/dspinst.h"
#include "Entropy.h"
// uncomment for more accurate but more computationally expensive frequency modulation
//#define IMPROVE_EXPONENTIAL_ACCURACY
#define BASE_AMPLITUDE 0x6000 // 0x7fff won't work due to Gibb's phenomenon, so use 3/4 of full range.
void AudioSynthWaveformTS::update(void)
{
audio_block_t *block;
int16_t *bp, *end;
int32_t val1, val2;
int16_t val3;
int16_t magnitude15;
uint32_t i, ph, index, index2, scale;
const uint32_t inc = phase_increment;
uint32_t phaseX;
if(syncFlag == 1) {
phase_accumulator = 0;
phaseX = 0;
syncFlag = 0;
lfo_randomFlag = false;
}
ph = phase_accumulator + phase_offset;
phaseX = phase_accumulator;
if (magnitude == 0) {
phase_accumulator += inc * AUDIO_BLOCK_SAMPLES;
return;
}
block = allocate();
if (!block) {
phase_accumulator += inc * AUDIO_BLOCK_SAMPLES;
return;
}
bp = block->data;
switch(tone_type) {
// PWM SINE -------------------------------------------------------
case PWM_WAVEFORM_SINE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
index = ph >> 24;
val1 = AudioWaveformSine[index];
val2 = AudioWaveformSine[index+1];
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0x10000 - scale;
*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
ph += inc;
}
break;
// PWM Triangle ---------------------------------------------------
case PWM_WAVEFORM_TRIANGLE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
uint32_t phtop = ph >> 30;
if (phtop == 1 || phtop == 2) {
*bp++ = ((0xFFFF - (ph >> 15)) * magnitude) >> 16;
} else {
*bp++ = (((int32_t)ph >> 15) * magnitude) >> 16;
}
ph += inc;
}
break;
// PWM SAWTOOTH ---------------------------------------------------
case PWM_WAVEFORM_SAWTOOTH: // normal sawtooth and inv sawtooth
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
*bp++ = signed_multiply_32x16t(magnitude, ph >> 1);
ph += inc;
}
break;
// PWM SQUARE -----------------------------------------------------
case PWM_WAVEFORM_SQUARE:
magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
//if (ph & 0x80000000) {
if (ph & 0x80000000) {
*bp++ = -magnitude15;
} else {
*bp++ = magnitude15;
}
ph += inc;
}
break;
// LFO Arbitrary waveform -----------------------------------------
case LFO_WAVEFORM_ARBITRARY:
uint8_t lfo_mode_;
uint8_t lfo_phase_;
boolean lfo_oneShoot_;
lfo_mode_ = lfo_mode;
lfo_phase_ = lfo_phase; // LFO parameter: SYN
lfo_oneShoot_ = lfo_oneShoot;
if (!arbdata) {
release(block);
phase_accumulator += inc * AUDIO_BLOCK_SAMPLES;
return;
}
// len = 256
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
index = ph >> 24;
index2 = index + 1;
if (lfo_mode_ == 0) {
if (index2 >= 256) index2 = 0; // loop
}
else {
if (index2 >= 256) index2 = 255; // OneShoot
}
val1 = *(arbdata + index);
val2 = *(arbdata + index2);
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0x10000 - scale;
*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
uint32_t ph_old = ph;
uint32_t phaseX_old = phaseX;
if (lfo_phase_ <= 1) { // Shape normal
ph += inc;
phaseX += inc;
}
else {
ph -= inc;
phaseX -= inc; // Shape inverse
}
if (lfo_oneShoot_ == true && phaseX < inc) {
ph = ph_old;
phaseX = phaseX_old;
}
}
break;
// LFO S&H waveform -----------------------------------------------
case LFO_WAVEFORM_SAMPLE_HOLD:
boolean randomFlag_;
boolean oneShoot_;
randomFlag_ = lfo_randomFlag;
oneShoot_ = lfo_oneShoot;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
uint32_t newph = ph + inc;
if (oneShoot_ == true && randomFlag_ == false) { // one shot
sample = random(magnitude) - (magnitude >> 1);
sample2 = sample;
randomFlag_ = true;
}
else if (newph < ph && oneShoot_ == false) {
sample = random(magnitude) - (magnitude >> 1);
}
else if (oneShoot_ == true && randomFlag_ == true) {
sample = sample2;
}
*bp++ = sample;
ph = newph;
}
lfo_randomFlag = randomFlag_;
break;
}
phase_accumulator = ph - phase_offset;
if (tone_offset) {
bp = block->data;
end = bp + AUDIO_BLOCK_SAMPLES;
do {
val1 = *bp;
*bp++ = signed_saturate_rshift(val1 + tone_offset, 16, 0);
} while (bp < end);
}
transmit(block, 0);
release(block);
}
//--------------------------------------------------------------------------------
void AudioSynthWaveformModulatedTS::update(void)
{
audio_block_t *block, *moddata, *shapedata;
int16_t *bp, *end;
int32_t val1, val2;
int16_t magnitude15;
uint32_t i, ph, index, index2, scale, priorphase;
const uint32_t inc = phase_increment;
uint32_t phase_spread;
uint32_t saw_phase_increment;
uint32_t increments[4];
uint32_t ph_1;
uint32_t ph_2;
uint32_t ph_3;
uint32_t ph_4;
uint32_t ph_5;
int32_t Ssaw_value;
moddata = receiveReadOnly(0);
shapedata = receiveReadOnly(1);
if(syncFlag==1){
phase_accumulator = 0;
syncFlag = 0;
}
// Pre-compute the phase angle for every output sample of this update
ph = phase_accumulator;
priorphase = phasedata[AUDIO_BLOCK_SAMPLES-1];
if (moddata && modulation_type == 0) {
// Frequency Modulation
bp = moddata->data;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
int32_t n = (*bp++) * modulation_factor; // n is # of octaves to mod
int32_t ipart = n >> 27; // 4 integer bits
n &= 0x7FFFFFF; // 27 fractional bits
#ifdef IMPROVE_EXPONENTIAL_ACCURACY
// exp2 polynomial suggested by Stefan Stenzel on "music-dsp"
// mail list, Wed, 3 Sep 2014 10:08:55 +0200
int32_t x = n << 3;
n = multiply_accumulate_32x32_rshift32_rounded(536870912, x, 1494202713);
int32_t sq = multiply_32x32_rshift32_rounded(x, x);
n = multiply_accumulate_32x32_rshift32_rounded(n, sq, 1934101615);
n = n + (multiply_32x32_rshift32_rounded(sq,
multiply_32x32_rshift32_rounded(x, 1358044250)) << 1);
n = n << 1;
#else
// exp2 algorithm by Laurent de Soras
// https://www.musicdsp.org/en/latest/Other/106-fast-exp2-approximation.html
n = (n + 134217728) << 3;
n = multiply_32x32_rshift32_rounded(n, n);
n = multiply_32x32_rshift32_rounded(n, 715827883) << 3;
n = n + 715827882;
#endif
uint32_t scale = n >> (14 - ipart);
uint64_t phstep = (uint64_t)inc * scale;
uint32_t phstep_msw = phstep >> 32;
if (phstep_msw < 0x7FFE) {
ph += phstep >> 16;
} else {
ph += 0x7FFE0000;
}
phasedata[i] = ph;
}
release(moddata);
} else if (moddata) {
// Phase Modulation
bp = moddata->data;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
// more than +/- 180 deg shift by 32 bit overflow of "n"
uint32_t n = (uint16_t)(*bp++) * modulation_factor;
phasedata[i] = ph + n;
ph += inc;
}
release(moddata);
} else {
// No Modulation Input
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
phasedata[i] = ph;
ph += inc;
}
}
phase_accumulator = ph;
//Amplitude is always 1 on TSynth when oscillator is sounding
//magnitude must be set to zero, otherwise digital noise comes through
if(tone_type == WAVEFORM_SILENT){
magnitude = 0;
}else{
magnitude = 65536.0;
}
// If the amplitude is zero, no output, but phase still increments properly
if (magnitude == 0) {
if (shapedata) release(shapedata);
return;
}
block = allocate();
if (!block) {
if (shapedata) release(shapedata);
return;
}
bp = block->data;
// Now generate the output samples using the pre-computed phase angles
switch(tone_type) {
// WaveformModulated Sine -----------------------------------------
case WAVEFORM_SINE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[i];
index = ph >> 24;
val1 = AudioWaveformSine[index];
val2 = AudioWaveformSine[index+1];
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0x10000 - scale;
*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
}
break;
// WaveformModulated Arbitrary ------------------------------------
case WAVEFORM_ARBITRARY:
if (!arbdata) {
release(block);
if (shapedata) release(shapedata);
return;
}
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[i];
index = ph >> 24;
index2 = index + 1;
if (index2 >= 256) index2 = 0;
val1 = *(arbdata + index);
val2 = *(arbdata + index2);
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0x10000 - scale;
*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
}
break;
// WaveformModulated Pulse ----------------------------------------
case WAVEFORM_PULSE:
if (shapedata) {
magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
uint32_t width = ((shapedata->data[i] + 0x8000) & 0xFFFF) << 16;
if (phasedata[i] < width) {
*bp++ = magnitude15;
} else {
*bp++ = -magnitude15;
}
}
break;
} // else fall through to orginary square without shape modulation
// WaveformModulated Square ---------------------------------------
case WAVEFORM_SQUARE:
magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
if (phasedata[i] & 0x80000000) {
*bp++ = -magnitude15;
} else {
*bp++ = magnitude15;
}
}
break;
// WaveformModulated Bandlimit Puls -------------------------------
case WAVEFORM_BANDLIMIT_PULSE:
if (shapedata)
{
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++)
{
uint32_t width = ((shapedata->data[i] + 0x8000) & 0xFFFF) << 16;
int32_t val = band_limit_waveform.generate_pulse (phasedata[i], width, i) ;
*bp++ = (int16_t) ((val * magnitude) >> 16) ;
}
break;
} // else fall through to orginary square without shape modulation
// WaveformModulated Bandlimit Square -----------------------------
case WAVEFORM_BANDLIMIT_SQUARE:
for (i = 0 ; i < AUDIO_BLOCK_SAMPLES ; i++)
{
int32_t val = band_limit_waveform.generate_square (phasedata[i], i) ;
*bp++ = (int16_t) ((val * magnitude) >> 16);
}
break;
// WaveformModulated Supersaw -------------------------------------
case WAVEFORM_SAWTOOTH:
// Braids vowel ---------------------------------------------------
int16_t parameter_[2];
uint16_t balance;
uint8_t vowel_index;
uint16_t formant_shift;
parameter_[0] = osc_par_a << 8; // convert parameter into int16 value
parameter_[1] = osc_par_b << 8; // convert parameter into int16 value
vowel_index = parameter_[0] >> 12;
balance = parameter_[0] & 0x0fff;
formant_shift = (440 + (parameter_[1] >> 5));
if (strike_) {
strike_ = false;
state_vow_consonant_frames = 160;
for (size_t i = 0; i < 3; i++) {
state_vow_formant_increment[i] = static_cast<uint32_t>(consonant_data[index].formant_frequency[i]) *
0x1000 * formant_shift;
state_vow_formant_amplitude[i] = consonant_data[index].formant_amplitude[i];
}
}
if (state_vow_consonant_frames) {
--state_vow_consonant_frames;
} else {
for (size_t i = 0; i < 3; ++i) {
state_vow_formant_increment[i] = (vowels_data[vowel_index].formant_frequency[i] *
(0x1000 - balance) + vowels_data[vowel_index + 1].formant_frequency[i] * balance) * formant_shift;
state_vow_formant_amplitude[i] = (vowels_data[vowel_index].formant_amplitude[i] *
(0x1000 - balance) + vowels_data[vowel_index + 1].formant_amplitude[i] * balance) >> 12;
}
}
uint32_t phase_;
for (uint8_t i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
phase_ = phasedata[i];
size_t phaselet;
int16_t sample = 0;
state_vow_formant_phase[0] += state_vow_formant_increment[0];
phaselet = (state_vow_formant_phase[0] >> 24) & 0xf0;
sample += wav_formant_sine[phaselet | state_vow_formant_amplitude[0]];
state_vow_formant_phase[1] += state_vow_formant_increment[1];
phaselet = (state_vow_formant_phase[1] >> 24) & 0xf0;
sample += wav_formant_sine[phaselet | state_vow_formant_amplitude[1]];
state_vow_formant_phase[2] += state_vow_formant_increment[2];
phaselet = (state_vow_formant_phase[2] >> 24) & 0xf0;
sample += wav_formant_square[phaselet | state_vow_formant_amplitude[2]];
sample *= 255 - (phase_ >> 24);
if (phase_ < phaseOld_) {
state_vow_formant_phase[0] = 0;
state_vow_formant_phase[1] = 0;
state_vow_formant_phase[2] = 0;
sample = 0;
}
phaseOld_ = phase_;
float sample_f;
sample_f = sample * 1.35f / 32768;
// a = 1.5*a - 0.5 * a³ //should be converted to fixed point
sample_f = 1.50f * sample_f - 0.5f * sample_f * sample_f * sample_f;
sample = sample_f * 32768;
*bp++ = (int16_t) ((sample * magnitude) >> 16);
}
break;
/*
// Braids VowelFof ------------------------------------------------
int16_t parameter_[2];
const uint8_t* sync,
int16_t* buffer,
size_t size) {
// The original implementation used FOF but we live in the future and it's
// less computationally expensive to render a proper bank of 5 SVF.
int16_t amplitudes[kNumFormants];
int32_t svf_lp[kNumFormants];
int32_t svf_bp[kNumFormants];
int16_t svf_f[kNumFormants];
for (size_t i = 0; i < kNumFormants; ++i) {
int32_t frequency = InterpolateFormantParameter(
formant_f_data,
parameter_[1],
parameter_[0],
i) + (12 << 7);
svf_f[i] = Interpolate824(lut_svf_cutoff, frequency << 17);
amplitudes[i] = InterpolateFormantParameter(
formant_a_data,
parameter_[1],
parameter_[0],
i);
if (init_) {
svf_lp[i] = 0;
svf_bp[i] = 0;
} else {
svf_lp[i] = state_.fof.svf_lp[i];
svf_bp[i] = state_.fof.svf_bp[i];
}
}
if (init_) {
init_ = false;
}
uint32_t phase = phase_;
int32_t previous_sample = state_.fof.previous_sample;
int32_t next_saw_sample = state_.fof.next_saw_sample;
uint32_t increment = phase_increment_ << 1;
while (size) {
int32_t this_saw_sample = next_saw_sample;
next_saw_sample = 0;
phase += increment;
if (phase < increment) {
uint32_t t = phase / (increment >> 16);
if (t > 65535) {
t = 65535;
}
this_saw_sample -= static_cast<int32_t>(t * t >> 18);
t = 65535 - t;
next_saw_sample -= -static_cast<int32_t>(t * t >> 18);
}
next_saw_sample += phase >> 17;
int32_t in = this_saw_sample;
int32_t out = 0;
for (int32_t i = 0; i < 5; ++i) {
int32_t notch = in - (svf_bp[i] >> 6);
svf_lp[i] += svf_f[i] * svf_bp[i] >> 15;
CLIP(svf_lp[i])
int32_t hp = notch - svf_lp[i];
svf_bp[i] += svf_f[i] * hp >> 15;
CLIP(svf_bp[i])
out += svf_bp[i] * amplitudes[0] >> 17;
}
CLIP(out);
*buffer++ = (out + previous_sample) >> 1;
*buffer++ = out;
previous_sample = out;
size -= 2;
}
phase_ = phase;
state_.fof.next_saw_sample = next_saw_sample;
state_.fof.previous_sample = previous_sample;
for (size_t i = 0; i < kNumFormants; ++i) {
state_.fof.svf_lp[i] = svf_lp[i];
state_.fof.svf_bp[i] = svf_bp[i];
}
}
break;
*/
// WaveformModulated Sawtooth reverse -----------------------------
case WAVEFORM_SAWTOOTH_REVERSE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
*bp++ = signed_multiply_32x16t(0xFFFFFFFFu - magnitude, phasedata[i]);
}
break;
// WaveformModulated Bandlimit Sawtooth and Sawtooth reverse ------
case WAVEFORM_BANDLIMIT_SAWTOOTH:
case WAVEFORM_BANDLIMIT_SAWTOOTH_REVERSE:
for (i = 0 ; i < AUDIO_BLOCK_SAMPLES ; i++)
{
int16_t val = band_limit_waveform.generate_sawtooth (phasedata[i], i) ;
val = (int16_t) ((val * magnitude) >> 16) ;
*bp++ = tone_type == WAVEFORM_BANDLIMIT_SAWTOOTH_REVERSE ? (int16_t) -val : (int16_t) +val ;
}
break;
// WaveformModulated Variable Triangle (Triangle -> Saw) ----------
case WAVEFORM_TRIANGLE_VARIABLE:
if (shapedata) {
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
uint32_t width = (shapedata->data[i] + 0x8000) & 0xFFFF;
uint32_t rise = 0xFFFFFFFF / width;
uint32_t fall = 0xFFFFFFFF / (0xFFFF - width);
uint32_t halfwidth = width << 15;
uint32_t n;
ph = phasedata[i];
if (ph < halfwidth) {
n = (ph >> 16) * rise;
*bp++ = ((n >> 16) * magnitude) >> 16;
} else if (ph < 0xFFFFFFFF - halfwidth) {
n = 0x7FFFFFFF - (((ph - halfwidth) >> 16) * fall);
*bp++ = (((int32_t)n >> 16) * magnitude) >> 16;
} else {
n = ((ph + halfwidth) >> 16) * rise + 0x80000000;
*bp++ = (((int32_t)n >> 16) * magnitude) >> 16;
}
ph += inc;
}
break;
} // else fall through to orginary triangle without shape modulation
// WaveformModulated Triangle -------------------------------------
case WAVEFORM_TRIANGLE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[i];
uint32_t phtop = ph >> 30;
if (phtop == 1 || phtop == 2) {
*bp++ = ((0xFFFF - (ph >> 15)) * magnitude) >> 16;
} else {
*bp++ = (((int32_t)ph >> 15) * magnitude) >> 16;
}
}
break;
// WaveformModulated Sample & Hold --------------------------------
case WAVEFORM_SAMPLE_HOLD:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[i];
if (ph < priorphase) { // does not work for phase modulation
sample = random(magnitude) - (magnitude >> 1);
}
priorphase = ph;
*bp++ = sample;
}
break;
}
if (tone_offset) {
bp = block->data;
end = bp + AUDIO_BLOCK_SAMPLES;
do {
val1 = *bp;
*bp++ = signed_saturate_rshift(val1 + tone_offset, 16, 0);
} while (bp < end);
}
if (shapedata) release(shapedata);
transmit(block, 0);
release(block);
}
// BandLimitedWaveform
#define SUPPORT_SHIFT 4
#define SUPPORT (1 << SUPPORT_SHIFT)
#define PTRMASK ((2 << SUPPORT_SHIFT) - 1)
#define SCALE 16
#define SCALE_MASK (SCALE-1)
#define N (SCALE * SUPPORT * 2)
#define GUARD_BITS 8
#define GUARD (1 << GUARD_BITS)
#define HALF_GUARD (1 << (GUARD_BITS-1))
#define DEG180 0x80000000u
#define PHASE_SCALE (0x100000000L / (2 * BASE_AMPLITUDE))
extern "C"
{
extern const int16_t bandlimit_step_table [258] ;
}
int32_t BandLimitedWaveform::lookup (int offset)
{
int off = offset >> GUARD_BITS ;
int frac = offset & (GUARD-1) ;
int32_t a, b ;
if (off < N/2) // handle odd symmetry by reflecting table
{
a = bandlimit_step_table [off+1] ;
b = bandlimit_step_table [off+2] ;
}
else
{
a = - bandlimit_step_table [N-off] ;
b = - bandlimit_step_table [N-off-1] ;
}
return BASE_AMPLITUDE + ((frac * b + (GUARD - frac) * a + HALF_GUARD) >> GUARD_BITS) ; // interpolated
}
// create a new step, apply its past waveform into the cyclic sample buffer
// and add a step_state object into active list so it can be added for the future samples
void BandLimitedWaveform::insert_step (int offset, bool rising, int i)
{
while (offset <= (N/2-SCALE)<<GUARD_BITS)
{
if (offset >= 0)
cyclic [i & 15] += rising ? lookup (offset) : -lookup (offset) ;
offset += SCALE<<GUARD_BITS ;
i ++ ;
}
states[newptr].offset = offset ;
states[newptr].positive = rising ;
newptr = (newptr+1) & PTRMASK ;
}
// generate value for current sample from one active step, checking for the
// dc_offset adjustment at the end of the table.
int32_t BandLimitedWaveform::process_step (int i)
{
int off = states[i].offset ;
bool positive = states[i].positive ;
int32_t entry = lookup (off) ;
off += SCALE<<GUARD_BITS ;
states[i].offset = off ; // update offset in table for next sample
if (off >= N<<GUARD_BITS) // at end of step table we alter dc_offset to extend the step into future
dc_offset += positive ? 2*BASE_AMPLITUDE : -2*BASE_AMPLITUDE ;
return positive ? entry : -entry ;
}
// process all active steps for current sample, basically generating the waveform portion
// due only to steps
// square waves use this directly.
int32_t BandLimitedWaveform::process_active_steps (uint32_t new_phase)
{
int32_t sample = dc_offset ;
int step_count = (newptr - delptr) & PTRMASK ;
if (step_count > 0) // for any steps in-flight we sum in table entry and update its state
{
int i = newptr ;
do
{
i = (i-1) & PTRMASK ;
sample += process_step (i) ;
} while (i != delptr) ;
if (states[delptr].offset >= N<<GUARD_BITS) // remove any finished entries from the buffer.
{
delptr = (delptr+1) & PTRMASK ;
// can be upto two steps per sample now for pulses
if (newptr != delptr && states[delptr].offset >= N<<GUARD_BITS)
delptr = (delptr+1) & PTRMASK ;
}
}
return sample ;
}
// for sawtooth need to add in the slope and compensate for all the steps being one way
int32_t BandLimitedWaveform::process_active_steps_saw (uint32_t new_phase)
{
int32_t sample = process_active_steps (new_phase) ;
sample += (int16_t) ((((uint64_t)phase_word * (2*BASE_AMPLITUDE)) >> 32) - BASE_AMPLITUDE) ; // generate the sloped part of the wave
if (new_phase < DEG180 && phase_word >= DEG180) // detect wrap around, correct dc offset
dc_offset += 2*BASE_AMPLITUDE ;
return sample ;
}
// for pulse need to adjust the baseline according to the pulse width to cancel the DC component.
int32_t BandLimitedWaveform::process_active_steps_pulse (uint32_t new_phase, uint32_t pulse_width)
{
int32_t sample = process_active_steps (new_phase) ;
return sample + BASE_AMPLITUDE/2 - pulse_width / (0x80000000u / BASE_AMPLITUDE) ; // correct DC offset for duty cycle
}
// Check for new steps using the phase update for the current sample for a square wave
void BandLimitedWaveform::new_step_check_square (uint32_t new_phase, int i)
{
if (new_phase >= DEG180 && phase_word < DEG180) // detect falling step
{
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (sampled_width - phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
if (pulse_state) // guard against two falling steps in a row (if pulse width changing for instance)
{
insert_step (- offset, false, i) ;
pulse_state = false ;
}
}
else if (new_phase < DEG180 && phase_word >= DEG180) // detect wrap around, rising step
{
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (- phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
if (!pulse_state) // guard against two rising steps in a row (if pulse width changing for instance)
{
insert_step (- offset, true, i) ;
pulse_state = true ;
}
}
}
// Checking for new steps for pulse waveform has to deal with changing frequency and pulse width and
// not letting a pulse glitch out of existence as these change across a single period of the waveform
// now we detect the rising edge just like for a square wave and use that to sample the pulse width
// parameter, which then has to be checked against the instantaneous frequency every sample.
void BandLimitedWaveform::new_step_check_pulse (uint32_t new_phase, uint32_t pulse_width, int i)
{
if (pulse_state && phase_word < sampled_width && (new_phase >= sampled_width || new_phase < phase_word)) // falling edge
{
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (sampled_width - phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
insert_step (- offset, false, i) ;
pulse_state = false ;
}
if ((!pulse_state) && phase_word >= DEG180 && new_phase < DEG180) // detect wrap around, rising step
{
// sample the pulse width value so its not changing under our feet later in cycle due to modulation
sampled_width = pulse_width ;
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (- phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
insert_step (- offset, true, i) ;
pulse_state = true ;
if (pulse_state && new_phase >= sampled_width) // detect falling step directly after a rising edge
//if (new_phase - sampled_width < DEG180) // detect falling step directly after a rising edge
{
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (sampled_width - phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
insert_step (- offset, false, i) ;
pulse_state = false ;
}
}
}
// new steps for sawtooth are at 180 degree point, always falling.
void BandLimitedWaveform::new_step_check_saw (uint32_t new_phase, int i)
{
if (new_phase >= DEG180 && phase_word < DEG180) // detect falling step
{
int32_t offset = (int32_t) ((uint64_t) (SCALE<<GUARD_BITS) * (DEG180 - phase_word) / (new_phase - phase_word)) ;
if (offset == SCALE<<GUARD_BITS)
offset -- ;
insert_step (- offset, false, i) ;
}
}
// the generation function pushd new sample into cyclic buffer, having taken out the oldest entry
// to return. The output is thus 16 samples behind, which allows the non-casual step function to
// work in real time.
int16_t BandLimitedWaveform::generate_sawtooth (uint32_t new_phase, int i)
{
new_step_check_saw (new_phase, i) ;
int32_t val = process_active_steps_saw (new_phase) ;
int16_t sample = (int16_t) cyclic [i&15] ;
cyclic [i&15] = val ;
phase_word = new_phase ;
return sample ;
}
int16_t BandLimitedWaveform::generate_square (uint32_t new_phase, int i)
{
new_step_check_square (new_phase, i) ;
int32_t val = process_active_steps (new_phase) ;
int16_t sample = (int16_t) cyclic [i&15] ;
cyclic [i&15] = val ;
phase_word = new_phase ;
return sample ;
}
int16_t BandLimitedWaveform::generate_pulse (uint32_t new_phase, uint32_t pulse_width, int i)
{
new_step_check_pulse (new_phase, pulse_width, i) ;
int32_t val = process_active_steps_pulse (new_phase, pulse_width) ;
int32_t sample = cyclic [i&15] ;
cyclic [i&15] = val ;
phase_word = new_phase ;
return (int16_t) ((sample >> 1) - (sample >> 5)) ; // scale down to avoid overflow on narrow pulses, where the DC shift is big
}
void BandLimitedWaveform::init_sawtooth (uint32_t freq_word)
{
phase_word = 0 ;
newptr = 0 ;
delptr = 0 ;
for (int i = 0 ; i < 2*SUPPORT ; i++)
phase_word -= freq_word ;
dc_offset = phase_word < DEG180 ? BASE_AMPLITUDE : -BASE_AMPLITUDE ;
for (int i = 0 ; i < 2*SUPPORT ; i++)
{
uint32_t new_phase = phase_word + freq_word ;
new_step_check_saw (new_phase, i) ;
cyclic [i & 15] = (int16_t) process_active_steps_saw (new_phase) ;
phase_word = new_phase ;
}
}
void BandLimitedWaveform::init_square (uint32_t freq_word)
{
init_pulse (freq_word, DEG180) ;
}
void BandLimitedWaveform::init_pulse (uint32_t freq_word, uint32_t pulse_width)
{
phase_word = 0 ;
sampled_width = pulse_width ;
newptr = 0 ;
delptr = 0 ;
for (int i = 0 ; i < 2*SUPPORT ; i++)
phase_word -= freq_word ;
if (phase_word < pulse_width)
{
dc_offset = BASE_AMPLITUDE ;
pulse_state = true ;
}
else
{
dc_offset = -BASE_AMPLITUDE ;
pulse_state = false ;
}
for (int i = 0 ; i < 2*SUPPORT ; i++)
{
uint32_t new_phase = phase_word + freq_word ;
new_step_check_pulse (new_phase, pulse_width, i) ;
cyclic [i & 15] = (int16_t) process_active_steps_pulse (new_phase, pulse_width) ;
phase_word = new_phase ;
}
}
BandLimitedWaveform::BandLimitedWaveform()
{
newptr = 0 ;
delptr = 0 ;
dc_offset = BASE_AMPLITUDE ;
phase_word = 0 ;
}
// Braids Vowel waveform
int16_t InterpolateFormantParameter(
const int16_t table[][kNumFormants][kNumFormants],
int16_t x,
int16_t y,
uint8_t formant) {
uint16_t x_index = x >> 13;
uint16_t x_mix = x << 3;
uint16_t y_index = y >> 13;
uint16_t y_mix = y << 3;
int16_t a = table[x_index][y_index][formant];
int16_t b = table[x_index + 1][y_index][formant];
int16_t c = table[x_index][y_index + 1][formant];
int16_t d = table[x_index + 1][y_index + 1][formant];
a = a + ((b - a) * x_mix >> 16);
c = c + ((d - c) * x_mix >> 16);
return a + ((c - a) * y_mix >> 16);
}Rolfdegen
Well-known member
I programmed AudioSynthWaveformModulated with two additional modulation inputs..
The modulation inputs can be connected to any modulation source (envelope, LFO and more..) in the AudioPatch.h
waveformMod into synth_waveform.cpp
Synth_waveform.h
The modulation inputs can be connected to any modulation source (envelope, LFO and more..) in the AudioPatch.h
waveformMod into synth_waveform.cpp
Code:
void AudioSynthWaveformModulatedTS::update(void)
{
audio_block_t *block, *moddata, *shapedata, *par_A, *par_B;
//audio_block_t *block, *moddata, *shapedata;
int16_t *bp, *end, *par_A_mod, *par_B_mod;
int32_t val1, val2;
int16_t magnitude15;
uint32_t i, ph, index, index2, scale, priorphase;
const uint32_t inc = phase_increment;
uint32_t phase_spread;
uint32_t saw_phase_increment;
uint32_t increments[4];
uint32_t ph_1;
uint32_t ph_2;
uint32_t ph_3;
uint32_t ph_4;
uint32_t ph_5;
int32_t Ssaw_value;
moddata = receiveReadOnly(0);
shapedata = receiveReadOnly(1);
par_A = receiveReadOnly(2); // new parameter_A input
par_B = receiveReadOnly(3); // new parameter_B input
if(syncFlag==1){
phase_accumulator = 0;
syncFlag = 0;
}
// Pre-compute the phase angle for every output sample of this update
ph = phase_accumulator;
priorphase = phasedata[AUDIO_BLOCK_SAMPLES-1];
if (moddata && modulation_type == 0) {
// Frequency Modulation
bp = moddata->data;
// read two new Modulation parameter from AudioConnection into AudioPatching.h
par_A_mod = par_A->data;
par_B_mod = par_B->data;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
int32_t n = (*bp++) * modulation_factor; // n is # of octaves to mod
int32_t ipart = n >> 27; // 4 integer bits
n &= 0x7FFFFFF; // 27 fractional bits
#ifdef IMPROVE_EXPONENTIAL_ACCURACY
// exp2 polynomial suggested by Stefan Stenzel on "music-dsp"
// mail list, Wed, 3 Sep 2014 10:08:55 +0200
int32_t x = n << 3;
n = multiply_accumulate_32x32_rshift32_rounded(536870912, x, 1494202713);
int32_t sq = multiply_32x32_rshift32_rounded(x, x);
n = multiply_accumulate_32x32_rshift32_rounded(n, sq, 1934101615);
n = n + (multiply_32x32_rshift32_rounded(sq,
multiply_32x32_rshift32_rounded(x, 1358044250)) << 1);
n = n << 1;
#else
// exp2 algorithm by Laurent de Soras
// https://www.musicdsp.org/en/latest/Other/106-fast-exp2-approximation.html
n = (n + 134217728) << 3;
n = multiply_32x32_rshift32_rounded(n, n);
n = multiply_32x32_rshift32_rounded(n, 715827883) << 3;
n = n + 715827882;
#endif
uint32_t scale = n >> (14 - ipart);
uint64_t phstep = (uint64_t)inc * scale;
uint32_t phstep_msw = phstep >> 32;
if (phstep_msw < 0x7FFE) {
ph += phstep >> 16;
} else {
ph += 0x7FFE0000;
}
phasedata[i] = ph;
}
release(moddata);
release(par_A);
release(par_B);
par_a_mod_ = *par_A_mod; // save modulation data
par_b_mod_ = *par_B_mod; // save modulation data
} else if (moddata) {
// Phase Modulation
bp = moddata->data;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
// more than +/- 180 deg shift by 32 bit overflow of "n"
uint32_t n = (uint16_t)(*bp++) * modulation_factor;
phasedata[i] = ph + n;
ph += inc;
}
release(moddata);
} else {
// No Modulation Input
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
phasedata[i] = ph;
ph += inc;
}
}
phase_accumulator = ph;
//Amplitude is always 1 on TSynth when oscillator is sounding
//magnitude must be set to zero, otherwise digital noise comes through
if(tone_type == WAVEFORM_SILENT){
magnitude = 0;
}else{
magnitude = 65536.0;
}
// If the amplitude is zero, no output, but phase still increments properly
if (magnitude == 0) {
if (shapedata) release(shapedata);
return;
}
block = allocate();
if (!block) {
if (shapedata) release(shapedata);
return;
}
bp = block->data;
// Now generate the output samples using the pre-computed phase anglesSynth_waveform.h
Code:
class AudioSynthWaveformModulatedTS : public AudioStream
{
public:
AudioSynthWaveformModulatedTS(void) : AudioStream(4, inputQueueArray),
phase_accumulator(0), phase_increment(0), modulation_factor(32768),
magnitude(0), arbdata(NULL), sample(0), tone_offset(0),
tone_type(WAVEFORM_SINE), modulation_type(0), syncFlag(0), osc_par_a(0),
osc_par_b(0), par_a_mod_(0), par_b_mod_(0)
{
}
void frequency(float freq) {
//freq = freq / 2.0f; // only for tone_type Vowel
if (freq < 0.0) {
freq = 0.0;
} else if (freq > AUDIO_SAMPLE_RATE_EXACT / 2) {
freq = AUDIO_SAMPLE_RATE_EXACT / 2;
}
phase_increment = freq * (4294967296.0 / AUDIO_SAMPLE_RATE_EXACT);
if (phase_increment > 0x7FFE0000u) phase_increment = 0x7FFE0000;
}
void amplitude(float n) { // 0 to 1.0
if (n < 0) {
n = 0;
} else if (n > 1.0) {
n = 1.0;
}
magnitude = n * 65536.0;
}
void sync() {
syncFlag = 1;
}
void offset(float n) {
if (n < -1.0) {
n = -1.0;
} else if (n > 1.0) {
n = 1.0;
}
tone_offset = n * 32767.0;
}
void begin(short t_type) {
tone_type = t_type;
if (t_type == WAVEFORM_BANDLIMIT_SQUARE)
band_limit_waveform.init_square (phase_increment) ;
else if (t_type == WAVEFORM_BANDLIMIT_PULSE)
band_limit_waveform.init_pulse (phase_increment, 0x80000000u) ;
else if (t_type == WAVEFORM_BANDLIMIT_SAWTOOTH || t_type == WAVEFORM_BANDLIMIT_SAWTOOTH_REVERSE)
band_limit_waveform.init_sawtooth (phase_increment) ;
}
void begin(float t_amp, float t_freq, short t_type) {
amplitude(t_amp);
frequency(t_freq);
begin (t_type) ;
}
void arbitraryWaveform(const int16_t *data, float maxFreq) {
arbdata = data;
}
void frequencyModulation(float octaves) {
if (octaves > 12.0) {
octaves = 12.0;
} else if (octaves < 0.1) {
octaves = 0.1;
}
modulation_factor = octaves * 4096.0;
modulation_type = 0;
}
void phaseModulation(float degrees) {
if (degrees > 9000.0) {
degrees = 9000.0;
} else if (degrees < 30.0) {
degrees = 30.0;
}
modulation_factor = degrees * (65536.0 / 180.0);
modulation_type = 1;
}
void parameter_a(uint8_t Osc_par_a) { // parameter_a im Osc Menu
osc_par_a = Osc_par_a;
}
void parameter_b(uint8_t Osc_par_b) { // parameter_b im Osc Menu
osc_par_b = Osc_par_b;
}
virtual void update(void);
private:
audio_block_t *inputQueueArray[4];
uint32_t phase_accumulator;
uint32_t phase_increment;
uint32_t modulation_factor;
int32_t magnitude;
const int16_t *arbdata;
uint32_t phasedata[AUDIO_BLOCK_SAMPLES];
int16_t sample; // for WAVEFORM_SAMPLE_HOLD
int16_t tone_offset;
uint8_t tone_type;
uint8_t modulation_type;
int16_t syncFlag;
uint32_t data_qs_phase[4];
uint32_t data_qs_phase_2[4];
uint8_t Osc_data_cr_decimate;
uint16_t Osc_data_cr_state;
uint16_t OscData_sec_phase;
BandLimitedWaveform band_limit_waveform;
uint8_t osc_par_a;
uint8_t osc_par_b;
uint8_t Osc_vw_update;
uint32_t state_vow_formant_phase[3];
uint32_t state_vow_formant_increment[3];
uint32_t state_vow_formant_amplitude[3];
boolean strike_;
uint16_t state_vow_consonant_frames;
uint32_t state_saw_phase[6];
uint32_t state_saw_lp;
uint32_t state_saw_bp;
uint16_t Osc_vw1_formant_increment[3];
uint16_t Osc_vw1_formant_amplitude[3];
uint16_t Osc_vw1_formant_phase[3];
uint32_t phaseOld_= 0;
uint16_t par_a_mod_; // vowel parameter_A
uint16_t par_b_mod_; // vowel parameter_B
};
Last edited:
Rolfdegen
Well-known member
Hi Teensy friends
Jeannie version 2.74 is now available as PlatformIO code on github..
https://github.com/rolfdegen/Jeannie-Open-source-Synthesizer
Jeannie version 2.74 is now available as PlatformIO code on github..
https://github.com/rolfdegen/Jeannie-Open-source-Synthesizer
Rolfdegen
Well-known member
Next Update coming soon.. and more Braids synthesis
Video
https://youtu.be/rIK02G-8oj4
Code on github..
Link: https://github.com/rolfdegen/Jeannie-Open-source-Synthesizer
Greetings. Rolf
| 22.10.23 | V2.79 | Added Braids and Shruthi Synthesis |
| | | Braids VOWL : low-fi Vowel synthesis. PRM_A : formant, PRM_B : formant_shift |
| | | Shruthi ZSAW : Phase-distortion sawtooth with filter sweep |
| | | Shruthi ZSYNC : |
| | | Shruthi ZTRI : |
| | | Shruthi ZRESO : |
| | | Shruthi ZPULS : |
| | | Shruthi Chrushed_Sine : |
| | | Braids VOSM : Sawtooth with 2 formants. PRM_A : formant1 frequency, PRM_B : formant2 frequency |
| | | Braids TOY : Low-fi Circuit-bent sounds. PRM_A : sample reduction, PRM_B : bit toggling |
| | | Braids SuperSaw : Swarm of 7 sawtooths, PRM_A: Detune, PRM_B High-pass filter |
| | | Braids ZLPF : Direct synthesis of LP/Peaking/BP/HP filtered waveform |
| | | Braids ZPKF : PRM_A: Cutoff frequency, PRM_B: Waveshape |
| | | Braids ZBPF : |
| | | Braids ZHPF : |
Video
https://youtu.be/rIK02G-8oj4
Code on github..
Link: https://github.com/rolfdegen/Jeannie-Open-source-Synthesizer
Greetings. Rolf