#envelope-generator — Public Fediverse posts

Microrack – First Impressions

When I first heard about the Microrack concept, I was still playing around with my own Educational DIY Synth Thing so naturally I was really interested in seeing what it would be about. That was back-end of 2024. It has taken until now to realise that from the original Kickstarter.

Now that I have one in my possession, are are a few notes about it as I start to play.

What Arrived

I ordered the Synth Starter Kit and an additional Subtractive Extra Kit. This has given me the following modules:

• 1x USB-C power module
• 1x 3.5mm speaker/output module
• 2x VCO/LFO
• 2x High/Low pass VCF
• 2x AD EG
• 1x Noise + S/H
• 1x Clock Counter
• 1x Lo-Fi Delay
• 1x Stylus Keyboard
• 1x 830 point solderless breadboard
• 1x pack of “premium” jumper leads

One immediate surprise was that there was only one breadboard and now additional means to power it. The description of the Subtractive Extra Kit in the actual pledge was as follows:

So I was expecting two breadboards and an additional set of jumper wires and whatever a “power extender” was. I’ve pinged them a message just to check I’m not missing anything.

But in reality, as these are standard breadboards and wires, the extender is as simple as connecting the power rails together between two breadboards. In fact that isn’t even necessary if the power module is placed in the middle across both boards, as shown in my photo above. So I just added another 830 point breadboard myself and got started.

First Use

Before the first use, I took their advice and “primed” the solderless breadboard. I soldered up a quick “jumper pin pusher” as shown below to push into each of the power rail pin holes to ensure the modules can be inserted easily.

My first attempt used a USB power hub and USB-C lead. The first visual impressions are great – the LEDs lighting up through the PCBs look really good and make for very pleasing visual feedback on what is going on.

Everything worked fine to start with, when there was just an oscillator and output. But when adding the filter and second oscillator as a LFO, I soon found that certain combinations would cause a power-out. It turns out it is really easy to overload a typical USB power pack once there are a few modules running.

I switched to a Raspberry Pi 4 USB-C power supply, which is rated at 5.1V / 3.0A. This works really well and I’ve not had any power issues since even with all modules powered up.

It is curious that the system only uses the four power rails of the breadboard. This introduces a few limitations:

• Modules can’t be placed absolutely anywhere – the breadboards have gaps where there are no pin holes so you have to place modules accordingly.
• Modules have to be placed to span the middle gap, as that has no continuity in the tracks, otherwise some jumper wires are required.
• The rails are not symmetrical – they provide +5V, -5V, +12V and GND so it is important to know which is which!

The idea, as I understand it, is to allow prototyping of additional circuits on the breadboard itself. But for most cases, the centre blocks of pin holes are ignored.

A basic oscillator – filter – output synth chain is easily put together and can be continuous driven or driven via CV from the stylus keyboard.

And this was where I hit the first hints of the limitations of the choice of provided modules.

Module Selection

The more I think about it, the more the choice of modules for the Synth Starter Kit seems a little odd.

When adding in the stylus keyboard, it has a GATE and CV output. CV to the oscillator is obvious. And naturally the GATE would probably be expected to go to an envelope generator.

But there is no VCA! I have a Lo-Fi delay module and an audio output module, but no other means, as far as I can see, of applying an envelope to an amplitude.

The clock module is also a bit of a mystery. First of all, it isn’t actually a clock but takes a clock input and uses to drive 8 GATE outputs, so to my mind that makes it a clock-driven gate sequencer. But there is nothing to generate a clock directly and I have nothing to trigger off 8 GATE outputs.

Similarly I’m struggling at this stage to know what to do with the noise + S/H module. With no EG-driven VCA the noise outputs make little sense, and I’m not quite sure how to drive the S/H module.

Enlightenment Dawns…

At this point I took a proper look at the two provided “cheat sheets” that came with the system and things start to make a little more sense.

There is a “Bass and Drum” patch that shows how to use the LFO as the clock with a reset after four steps, triggering the EG on step one and noise via the S/H on all four steps. The EG is driving the filter and you get a pretty quirky noisy pulse with a nice “pew” on every fourth step.

I’m not entirely sure what “Jacuzzi + The Drone” was all about – that just seemed quite a few farty noises that I couldn’t get to do anything interesting (to me). I might have to give this another go later.

“ARP Lead” shows how to use the EG as a clock source and the GATE sequencer to drive a pitch level on certain steps of the clock. It also shows how the stylus can be used to set the base pitch for the sequence.

I’ve also played with using the clock module as a frequency divider to get a sub-octave.

At this point, we can see how creative use of the modules starts to show how the components can come together for something interesting.

Good Points

The modules do seem very full-featured. The oscillator for example, has VCO and LFO modes, with three output waveforms and the option for FM. It also has an option for -5V to +5V or 0 to +5V to support full audio output or use as a CV for other modules. I’m not entirely sure I understand at this stage what all the LEDs are telling me, but I guess I’ll figure that out.

There are some nice conventions too: blue headers for inputs, red for outputs, for example. And it is obvious some modules are designed for plugging next to each other.

The use of jumper wires makes for very easy integration with other things, so I’m looking forward to seeing how I can hook it up to my Volca Modular and my Educational DIY Synth Thing.

The modules are fully open sourced and documented online here: https://github.com/microrack/modules

The PCB design is very neat and tidy and seems quite a master of minaturisation.

The electrical specification is pretty complete with lots of features designed for robust protection when using with other devices. All provided modules have protections for shorts, reverse polarity, over and under voltage, and so on. The mechanical specification is also fully published. All details here: https://specs.microrack.org/

In short, once you get stuck in, everything seems very well thought out.

Conclusion

My initial hope was for use in an educational setting, but I’m not sure the bare PCB approach would be up to that. I don’t know if there would be ESD issues, but I do think that repeatedly plugging in and out of the breadboard would eventually take its toll on the power pins. Also, I think it would be far too easy to get the power connections either shorted or mixed up, especially if creating prototype additional circuits.

As a functioning synth for my own messing about, I’m going to have to invest in a couple more modules I think. I will need at least one mixer/VCA, and probably one of the headers-to-jacks modules too. On the main site, it still lists modules as for pre-order – I suspect that is because the Kickstarter units are still on their way to backers.

I wasn’t originally interested in MIDI to CV or the microcontroller DIY board, as I figured those would be fairly easy to do myself. And whilst that is still the case, I’m still wondering about their addition to create more of a stand-alone setup to play with.

It is interesting to note that there is a Eurorack mounting kit, but it is quite expensive, so I’m not sure I’m too fussed about that myself. But I might attempt to knock up some kind of 3D printed frame for housing a couple of modules. In fact, some users in the Microrack forums have done exactly that.

On balance, this is a very well thought out kit in general terms, but I think the choice of initial modules has veered towards the “beat box” rather than actual synth, but that is perhaps just my preference. But the three sample patches, and some of the ideas starting to appear in the forums, do show the potential of even what I already have.

At this stage, I’m looking forward to some proper playing around and I plan to digest some of the design information to see what I can do with the DIY side of the kit too.

I haven’t talked about the audio of course. It is pretty Lo-Fi, which fits right in for me of course.

There are already a number of videos up online from people far more knowledgeable than me, showing it in action. So I’ll leave it to those to show what it can do audio wise.

Like many systems, I think this will reward any time spent getting to know it and what the modules can do.

Kevin

Three new boards for my #StackSynth arrived from @aislerhq yesterday. They are buffer, #ADSR, and #VCO. The VCO is the first with #SMD components on the bottom side. Two resistor values are missing in my SMD collection: 30k and 68k. I already soldered the ADSR and tested it. The buffer will follow tomorrow.
#diysynth #soldering #synthesizer #envelopegenerator

In the first part I described the concept and basic design parameters I was looking for and then in Part 2 I looked at the detailed electronic design for a board based on the ESP32 WROOM module I have. In this part I go into detail about the code.

• Part 1 – This introduction and high-level design principles.
• Part 2 – Detailed design of an ESP32 based PCB.
  • PCB Design
  • PCB Build
• Part 3 – Software design.
• Part 4 – Mechanical assembly and final use – todo.

Warning! I strongly recommend using old or second hand equipment for your experiments.  I am not responsible for any damage to expensive instruments!

The ESP32 EduMod Hacked Synth Thing

As mentioned in the last part, the principles and code I’m using are taken from these previous posts:

• ESP32 and Mozzi.
• ESP32 and PWM.
• ESP32 DAC Envelope Generator.

I also now have a built PCB design to make development and testing a lot easier.

Recall the following design choices:

• The two ESP32 DACs are the output for two envelope generators, each driven by four pots (in standard ADSR configuration), a trigger and a gate signal.
• There are ESP32 PWM outputs for two VCOs, each with four waveforms (sine, triangle, saw, square) and an analog input (pot or CV) for frequency and amplitude.
• ESP32 PWM outputs are also used for a LFO with rate and depth analog controls.
• The ESP32 UART is used for MIDI in.
• ESP32 digital IO is used for the envelope GATE and TRIGGER signals, although they are level inverted, i.e. they should be read as active LOW.

Othe points to note:

• There are three “native” analog inputs used for the two VCO pitch and amplitude CVs. All other pots are read via a CD4067 16-way multiplexer.
• Software will determine how the VCO CVs interact with the potentiometers.
• The UART RX pin will be used for MIDI and the UART TX pin will be used for one of the PWM outputs, so there will be no serial port (via USB) for diagnostic output.
• The ESP32 is a dual core microprocessor, so the aim is to split the functionality across the two cores.

High Level Design

The tasks will be split as follows:

• Core 0:
  • Runs the PWM timer at 32,768Hz (although the sample rate will be half this – more on that later).
  • Manages the PWM outputs for the two VCOs and the LFO.
• Core 1:
  • Runs the EG timer for the DAC at 10,000 Hz.
  • Handles the reading of all the IO: “real” analog values; multiplexed analog values; digital IO.
  • Handles MIDI.

Multicore ESP32 Arduino

Multicore support for the ESP32 on Arduino is provided using FreeRTOS tasks. For a FreeRTOS application, the default Arduino setup()/loop() will run on core 1. However, once tasks have been created to run on the second core, the default loop is no longer used (I believe).

FreeRTOS enables multitasking on a single CPU (core), each with its own internal priority relative to other tasks. But I’m only using it to start a single task on each core. This is achieved using the xTaskCreatePinnedToCore() function. The last parameter specifies which core to use and the 5th parameter indicates the priority – both are set to “1” as there is only a single task on each core.

The Arduino startup code has thus been reduced to the following:

void core0Entry (void *pvParameters);
TaskHandle_t core0Task;
void core1Entry (void *pvParameters);
TaskHandle_t core1Task;

void setup () {
  xTaskCreatePinnedToCore(core0Entry, "Core 0 Task", 4096, NULL, 1, &core0Task, 0);
  xTaskCreatePinnedToCore(core1Entry, "Core 1 Task", 2048, NULL, 1, &core1Task, 1);
}

void loop () {
  // No main loop here
}

void core0Entry (void *pvParameters) {
  Task0Setup();

  for (;;) {
    Task0Loop();
    vTaskDelay(1); // Allow FreeRTOS IDLE to maintain watchdog
  }
}

void core1Entry (void *pvParameters) {
  Task1Setup();

  for (;;) {
    Task1Loop();
    vTaskDelay(1); // Allow FreeRTOS IDLE to maintain watchdog
  }
}

I’ve simulated the setup()/loop() pattern in each of the task entry points. One thing I quickly found was that if the tasks run with no release of the processor, then FreeRTOS will fail with a watchdog error.

This is why each task includes a vTaskDelay(1) call to allow FreeRTOS to do whatever it needs to do as part of its “idle” processing.

The other thing to be careful of, which would be the same even in a single core application, is that the timers and their timer interrupt routines also allow some general processing time on each core. This will become important further in the discussion.

Task 0: PWM Generation

The essence of Task 0 is the code from ESP32 and PWM. In order to simplify things I’ve created a basic oscillator class as follows:

class COsc {
public:
  COsc ();
  ~COsc (void);

  void Setup (int pwmpin, int wave, bool lfo=false);
  void QuadSetup (int sinpin, int sawpin, int tripin, int squpin);
  void ScanPWM (void);
  void SetFreq (uint16_t freq, uint16_t vol=MAX_VOL);
  void SetMVolts (uint32_t mvolts, uint16_t vol=MAX_VOL);
}

Initially the plan was to have one instance per PWM output. The pin and required waveform is configured as part of the call to the Setup() function.

But then I decided it would be more efficient to utilise the fact that the same index and accumulator can be used to read through each of the four wavetables, so created the idea of a “quad” oscillator. This just requires the four PWM pins to be used to be specified as part of the call to QuadSetup() and eliminates several calls into the ScanPWM function for separate oscillators.

There are two ways to set the oscillator’s frequency output. One is to pass in the frequency directly, the other is to pass in a millivolt reading from a potentiometer based on a 1V/oct scaling. The “zero volts” frequency is preset to be 65.406Hz, i.e. C2.

The frequency is derived from the millivolt reading using the following:

void COsc::SetMVolts (uint32_t mvolts, uint16_t vol) {
  // Set frequency according to Volt/Oct scaling
  //   Freq = baseFreq * 2 ^ (voltage)
  //
  float freq = FREQ_0V * (powf (2.0, ((float)mvolts)/1000.0));
  int f = (int)freq;
  SetFreq(f, vol);
}

With the default settings this class can be used for both VCO and LFO, but the lowest frequency for the LFO was just 1Hz. Consequently, I’ve added the option to specify a “LFO” setting as part of the single oscillator Setup() function. If enabled, then the oscillator runs 10x slower than default. This allows for frequencies down to 0.1Hz, but does mean that the frequency or millivolt parameters are now treated as being 10 times smaller than their values would imply.

Note when MIDI is added into the mix, I just use the NoteOn message to change the base frequency used for the oscillators.

The ScanPWM() function is designed to be called from the interrupt routine running at the sample rate.

The oscillators are thus created as two Quad oscillators (one per VCO) and two single oscillators for the dual waveforms of the LFO:

#define OSC_VCO1 0
#define OSC_VCO2 1
#define OSC_LFO1 2
#define OSC_LFO2 3
#define NUM_OSC 4
COsc *pOsc[NUM_OSC];
int pwmpins[NUM_OSC][4] = {
  {17,18,5,19},
  {21,22,1,23},
  {16,0,0,0},
  {13,0,0,0}
};
int oscwaves[NUM_OSC] = {
  OSC_WAVES,  // VCO 1 (quad)
  OSC_WAVES,  // VCO 2 (quad)
  OSC_TRI,    // LFO
  OSC_SAW
};

void oscSetup() {
  for (int i=0; i<NUM_OSC; i++) {
    pOsc[i] = new COsc();
    if (oscwaves[i] == OSC_WAVES) {
      // This is a quad oscillator
      pOsc[i]->QuadSetup (pwmpins[i][0], pwmpins[i][1], pwmpins[i][2], pwmpins[i][3]);
    } else {
      // A single oscillator - these are the LFOs
      pOsc[i]->Setup (pwmpins[i][0], oscwaves[i], true);      
    }
  }
}

The timer configuration for Task 0 is as follows:

#define TIMER_FREQ  10000000  // 1MHz * 10 (0.1uS)
#define TIMER_RATE  305       // 30.5 uS * 10

Task0Timer = timerBegin(TIMER_FREQ);
timerAttachInterrupt(Task0Timer, &Task0TimerIsr);
timerAlarm(Task0Timer, TIMER_RATE, true, 0);

This generates a timer interrupt running at 32,768Hz from a 10MHz timer

However, servicing 10 PWM outputs means that there is no time remaining once interrupt processing is complete for any other operations. Consequently, I set things up so that the PWM outputs are updated in two blocks of 5, meaning that the effective sample rate for PWM purposes is actually now 16,384Hz:

int toggle0;
void ARDUINO_ISR_ATTR Task0TimerIsr (void) {
  toggle0 = !toggle0;
  if (toggle0) {
    pOsc[OSC_VCO1]->ScanPWM();
    pOsc[OSC_LFO1]->ScanPWM();
  } else {
    pOsc[OSC_VCO2]->ScanPWM();
    pOsc[OSC_LFO2]->ScanPWM();
  }
}

The main Task 0 Loop just pulls in the analog IO values that have been read as part of the processing of Task 1. These values are then used to set the frequencies of each VCO and the LFO.

Task 0: VCO Control

The VCO potentiometers and CV inputs have to be combined to determine the frequency and amplitude of the PWM signal.

In terms of amplitude, I’ve chosen to use the maximum of the pot or CV input to set the amplitude of VCO 1. Note that VCO 2 has no amplitude input, although with hindsight, I could have included a pot even though I’ve run out of GPIO pins for a CV input.

This means that the pot has to be turned fully “off” for the CV to become significant.

In terms of setting the frequencies, I initially thought about some kind of frequency modulation, which would have been good for linking the two VCOs together, but then decided actually I wanted a more straight forward relationship between pot and CV to allow the use of the LFO or the EGs for pitch. The result is going for a simple summation of the two inputs which gives a 3 octave range (started at C2) for each of the CV and pot, but when combined could give up to a 6 octave spread of frequencies.

When MIDI is enabled, that will set the base frequency for the summation rather than defaulting to C2. This will set the base for both VCOs, so the pots/CVs could be used for a detuning style effect.

Note: Due to the issues with the choice of pins for the inputs, the base level of the CV input for VCO2 is floating at around 300-600mV. The default (unconnected) configuration introduces an element of detuning. If the CV inputs are not wanted, then it is recommended to connect them to GND to eliminate this effect.

The result of taking this approach, combined with the frequency of scanning the inputs, means that if one VCO is used to set the frequency for the other, then it probably won’t track the voltages accurately enough for a smooth modulation. But it does generation quite a wacky “sample and hold” style aliasing effect, which actually I quite like.

I could try to do something about that, but I think in reality this is a use-case where a purely analog VCO can be added via breadboard as part of the experimentation which can be driven from the PWM VCOs – that should enable a number of possibilities for more complex oscillator modulation.

There is also some clipping at higher LFO frequencies if used as the amplitude input. This needs to be investigated to see if that is a bug in the code or handling of the CV somehow.

It is possible that this, and the aliasing of pitch, is a consequence of the frequency of scanning the analog inputs or the speed of the analogRead() function itself – it’s pretty slow. Some investigation is required to experiment with changing the frequency of the reading or rewriting to use the ESP32 API rather than the (slower) Arduino analogRead() function (more on that later).

Task 1: IO Scanning

All the IO is scanned as part of Task 1’s main processing loop. This has to cover the following:

• Real analog values from the inputs corresponding to the VCO CVs.
• Multiplexed analog values for all the pots for the VCOs, LFO and two EGs.
• Digital IO for the EG gates and triggers.

I started off reading the “real” analog IO associated with the VCOs in task 0, but found that it just wasn’t being scanned frequently enough to function in any useful manner. Also, the calls to analogRead() incur some kind of processing overhead that appears to interfere with the timing of the PWM timer interrupt, leaving “blips” in the output.

I started to think about writing a FastAnalogRead() function to use the ESP32 SDK directly rather than the Arduino analogRead() function, but before doing that moved all the IO processing over to Task 1. It turns out that this seems to work adequately so if I’ve not done that at present.

But one consequence of this chain of thought is that I now have two sets of functions performing my own analog IO reading:

• MuxAnalogRead()
• FastAnalogRead()

Which both perform in very similar ways. I’ll describe the API for MuxAnalogRead(), but the same principles were used for FastAnalogRead() too.

void MuxSetup(void);
void MuxLoop(bool bVoltageReading=false);
void MuxLoopSingle (uint8_t pot, bool bVoltageReading=false);
uint16_t MuxAnalogRead (uint8_t pot);
uint32_t MuxAnalogReadMilliVolts (uint8_t pot);

The two Loop functions are designed to perform the actual reading from the hardware. Loop will read all known values in a single scan. LoopSingle() will read a single pot.

The functions MuxAnalogRead() and MuxAnalogReadMilliVolts() are used to retrieve the previously read values.

Scanning is performed using either the standard analogRead() or the ESP32 API function analogReadMilliVolts(). This is why a bVoltageReading parameter is required as part of the two Loop functions. It would be possible to calculate one from the other for every reading, but I’ve opted not to do that in order to keep the calculations required as part of a read as minimal as possible.

This allows the scanning of the hardware to happen on one core (in Task 1 in this case) yet be read from the other (Task 0).

Digital IO processing just requires monitoring the trigger and gate inputs for a HIGH->LOW transition and call the appropriate triggerADSR() or gateADSR() functions as required, although there is a bit more logic required to work in the MIDI input.

The actually scanning of IO is split over the different scans of the Task Loop as follows:

#define S_MUX (NUM_MUX_POTS-1)
#define S_FASTIO (S_MUX+1)
#define S_DIGIO  (S_FASTIO+1)
#define S_ALGIO  (S_DIGIO+1)
#define S_MIDI   (S_ALGIO+1)
#define S_LAST   (S_MIDI+1)
int taskState;

void Task1Loop(void)
{
  if (taskState <= S_MUX) {
    // Read each MUX pot in turn
    switch (taskState) {
      case 5: // VCO 1 CV
      case 7: // VCO 2 CV
        // These are read in millivolts...
        MuxLoopSingle(taskState, true);
        break;
      default:
        // These are read as 12-bit raw 0..4095
        MuxLoopSingle(taskState, false);
        break;
    }
  }
  else if (taskState == S_FASTIO) {
    Task1FastAlgIOLoop();
  }
  else if (taskState == S_DIGIO) {
    Task1DigIOLoop();
  }
  else if (taskState == S_ALGIO) {
    Task1AlgIOLoop();
  }
  else if (taskState == S_MIDI) {
    Task1MIDILoop();
  }
  else {
    taskState = S_LAST;
  }
  taskState++;
  if (taskState >= S_LAST) {
    taskState = 0;
  }
}

Task 1: Envelope Generation

The core of the envelope generation code is from ESP32 DAC Envelope Generator, but once again the code has been refactored into a class as follows:

class CEnv {
public:
  CEnv ();
  ~CEnv (void);

  void setADSR (int eg_a, int eg_d, int eg_s, int eg_r);
  uint8_t nextADSR (void);
  void triggerADSR (void);
  void gateADSR (bool gateOn);
  void dumpADSR (void);
}

One change is setting the ADSR parameters in a single call. The values are assumed to be potentiometer readings in the range 0..4095.

Two envelope generators are created by Task 1:

CEnv *pEnv[NUM_ENV];

void envSetup() {
  for (int i=0; i<NUM_ENV; i++) {
    pEnv[i] = new CEnv();
  }
}

The nextADSR() function has to be called for each envelope generator from the timer interrupt routine.

The timer configuration for Task 1 is as follows:

#define TIMER_FREQ  100000  // 100kHz (0.01mS)
#define TIMER_RATE  10      // 0.1mS

Task1Timer = timerBegin(TIMER_FREQ);
timerAttachInterrupt(Task1Timer, &Task1TimerIsr);
timerAlarm(Task1Timer, TIMER_RATE, true, 0);

This generates a 10kHz sample rate for the EGs from a 100kHz timer signal.

Task 1: MIDI

I’m using the standard Arduino MIDI library. There are a couple of points to note however:

• MIDI will set the TRIGGER and GATE according to the reception of NoteOn and NoteOff messages.
• I will need to keep track of which note is playing so that overlapping NoteOn/NoteOff messages don’t confuse the GATE handling.
• As I’m reusing the TX pin as a PWM output, PWM initialisation must happen after MIDI initialisation, otherwise MIDI will assume the use of both RX and TX.
• Serial MIDI by default implements “software MIDI THRU” so this needs to be disabled using setThruOff().
• MIDI will set the base frequency used for the VCO controls, so the pot and CV will add to the MIDI frequency.

It isn’t clear what should happen with regards to frequency on NoteOff. Here are some options:

• Reset back to default base frequency on NoteOff (the “0V” state), but this has the disadvantage that the frequency changes as soon as the gate signal initiates the Release state of the ADSR.
• Wait until the envelope has finished the Release stage to reset the base frequency. This seems sensible when using the EGs, but has the disadvantage that when just using MIDI to set pitch, it will not revert back immediately on “key up” – it will only revert back when the envelope is complete, which is confusing if the EG is not being used.
• Don’t reset the base frequency, this means the note will continue on the last used frequency until a new note is played.

I’ve opted not to reset the base frequency on the basis of: if MIDI on/off handling is required, then it will probably be using envelopes anyway. If not, then having only MIDI on setting a sustained frequency sort of makes more sense to me.

Closing Thoughts

The code will continue to evolve as I keep tinkering with it, but the current state can be found in the GitHub repository for the project here: https://github.com/diyelectromusic/ESP32EduModHackThing

There are still a few quirks and things I might continue to investigate and possibly optimise. Specifically:

• The performance for reading the “real” analog CVs for the VCOs.
• The frequency handling with respect to MIDI.

But in general I think it is starting to get pretty usable.

Kevin

https://diyelectromusic.wordpress.com/2024/05/26/educational-diy-synth-thing-part-3/

#adsr #define #envelopeGenerator #esp32 #lfo #oscillator #vca #vco

In the first part I described the concept and basic design parameters I was looking for. In this part I go into detail about how I’m going about implementing them with an ESP32 WROOM board.

• Part 1 – This introduction and high-level design principles.
• Part 2 – Detailed design of an ESP32 based PCB.
  • PCB Design
  • PCB Build
• Part 3 – Software design.
• Part 4 – Mechanical assembly and final use – todo.

Warning! I strongly recommend using old or second hand equipment for your experiments.  I am not responsible for any damage to expensive instruments!

The ESP32 EduMod Hacked Synth Thing

The principles and code I’m using are taken from these previous posts:

• ESP32 and Mozzi.
• ESP32 and PWM.
• ESP32 DAC Envelope Generator.

From the experience gained so far, I’ve gone for the following:

• Use the ESP32 DACs for the output for two envelope generators, each driven by four pots, a trigger and a gate signal.
• Use the ESP32 PWM outputs for two VCOs, each with four waveforms (sine, triangle, saw, square) and an analog input (pot or CV) for frequency and amplitude.
• Use another ESP32 PWM output for a LFO with rate and depth analog controls.
• Use the ESP32 UART for MIDI in.
• Use a TDA7052A as the final VCA/audio output stage.

Adding all this up, and taking the view that the VCOs will be supporting simultaneous waveforms on four pins each (rather than any kind of waveform selection input), I’m going to need some kind of IO expander. The easiest thing is probably to use an analog multiplexer for the twin ADSR and LFO controls. I’m designing this around a CD4067 analog multiplexer which provides 16 input channels and will require 4 IO pins for control and 1 IO pin for the analog value to read.

This leads me to the following pin usage, using the same cheap ESP32 WROOM module, I’ve used for my other experiments so far:

One thing I hadn’t considered when defining the above pinout (and subsequently designing a circuit around it) is the ESP32’s strapping pins. There are six pins that are used as configuration pins on boot to set the various mode and operation of the device and of course it assumes the use of the UART. Once booted, they can usually be used as normal GPIO pins, but if connected circuitry changes the state on boot, then unexpected things may occur.

See the discussion below for further details, implications, and things I’d wish I’d realised before putting a design together 🙂

The analog inputs for the Mux are defined as follows:

Basic electrical properties and design principles

• All internal control signals (CV, triggers or gates) will be using 3V3 levels. All internal audio signals will be DC biased to the 0-3V3 range.
• Internal patch points will be using jumper wires to make it clear that these are not “proper” modular signals for general patching outside of the unit.
• A special input will be provided for a MIDI IN socket.
• CV, trigger and gate signals will include basic protection to limit them to the 0-3V3 levels for use within the system.
• An output stage will be provided that will support either a directly connected small (8Ω) speaker or a mono line out jack.
• Power will be duplicated to a header on the side of the unit to allow for easy connections to breadboards.
• The PCBs will fit in a 100x100mm footprint to keep costs down, but I might end up with several per unit.
• All code will run on a single ESP32 WROOM dev module that seems easily, and cheaply, available.

There will be some, but limited protections on internal jumper connections. I’m working on the basis that it should be fairly cheap to build, all chips should be relatively cheap and easy to replace (and therefore socketed), so educated experimentation is encouraged, but uneducated plugging in will probably break something.

As long as everything stays within the 0-3V3 range, it should probably be ok, though.

ESP32 Strapping Pins

Section 2.4 of the ESP32 datasheet (“Strapping Pins”) has all the details. Essentially a number of pins are pulled high or low using the internal pullup or pulldown resistors. These define the boot configuration for the board and can be overridden by external pullup or pulldown resistors.

One or two other pins have additional circuitry too on a typical DevBoard.

The key GPIO pins used as Strapping pins are:

• GPIO 0 is the boot button, so I’m not worried about that. However to enter boot mode both GPIO 0 and GPIO 2 must be LOW. If my use of GPIO 2 is an issue, then I’ll have to flash the board before plugging it in. Once plugged in and booting from flash, then the state of GPIO 2 is irrelevant to the ESP32. GPIO 2 is pulled LOW via an external 10K resistor though on my dev board too.
• GPIO 2 on my DevKit board is also connected to the onboard LED with a 1K resistor to ground. This means that it’s use for an analog input was probably a mistake! It is possible to remove the onboard LED if required, but otherwise this means that the analog input range will be somewhat decreased to around 2.3V.
• GPIO 4 is pulled LOW. I don’t know what the significance of this is, but as this is connected to a CV input stage, when nothing is plugged in, this will be default pulled LOW externally too. I don’t know what will happen if a CV is present on boot…
• GPIO 5 is internally pulled HIGH. It is connected to a PWM output circuit, so I might be ok. To be determined…
• GPIO 12 (MTDI) is pulled LOW. It is connected to the MUX address pin, which is a digital input pin. If that works like a microcontrollers input pin then it will have a high impedance in which case, this should be fine.
• GPIO 15 (MTDO) is pulled HIGH. This determines if the boot is “silent” or not – i.e. if startup messages are sent to the UART. As I’m using the TX pin (GPIO 1) as a PWM output, it would be advantageous actually to have this LOW to stop any output, but either way it shouldn’t be a big deal. This pin is connected to a CV input stage with a default pull LOW resistor if nothing is plugged in, but I suspect the internal pull-up to be the stronger of the two and most likely to “win” out.
• GPIO 1 and 3 are TX and RX respectively. RX is connected to the MIDI IN circuit, so this means that board can’t be programmed while connected anyway. TX is configured for a PWM output circuit so there may be spurious output signals on startup for a short period of time on this output.

There is a good summary of some of the implications of attempting to use the strapping pins on this page here.

From the above discussion, I think I might get away with it for the most part (although GPIO 2 is annoying), but I’ll have to see once the boards are made up and do some proper tests. At present, I suspect the most significant ramification of the above is that I won’t be able to program the ESP32 module whilst plugged in, but that is likely to be as a result of the MIDI circuit on RX rather than anything else.

Update after testing:

• GPIO 0: not relevant.
• GPIO 2: as mentioned above, GPIO 2 will not present the full range as the analog input. I believe this is related to the voltage drop across the LED so one option is simply to desolder the LED or its resistor from the board. Also, when not connected the minimum reading is equivalent to 400-600mV rather than zero.
• GPIO 4: this largely works, but when not connected does seem to not quite read full zero, but that could just be noise.
• GPIO 5: no noticeable impact.
• GPIO 12: no noticeable impact.
• GPIO 15: this works fine with an analog input connected, but like GPIO 4 doesn’t quite get to zero when not connected.

ESP32 VCOs and LFO

These will be based on the ESP32 and PWM experiments – I’m planning to include two VCOs and a LFO.

The VCOs will take an analog input – a control voltage for frequency which can come from a pot or an external signal. One of the VCOs will also include a CV for amplitude, to allow some simple shaping (e.g. from the LFO). Both will output four PWM signals: sine, triangle, saw, square – each on its own output pin. I’ll use the built-in tone() function for the square wave – that won’t need a DDS approach.

The CV input will be using a 1V/oct scale, so the input range will be relatively narrow – just over three octaves. I do plan to hook up a MIDI input to the ESP32’s UART so that can feed a frequency directly into one of the VCOs without going via the analog stage, to give a wider range over MIDI only.

Of course, I need to decide what note/frequency 0V will start at.

Control Voltages

Each VCO analog input stage will consist of a pot which can act as a manual control voltage or as some kind of control for an external control voltage. But should the pot add to the external voltage or limit (attenuate) it? If it adds, then it would provide the ability to set an offset to the external signal. If attenuating then it affects the range of the external signal.

I originally went with the following circuit, which is largely based on the input stage of the MiniMo (ATtiny85 based) synth.

But the operation of the pot wasn’t quite what I wanted. With no CV, then the pot allows adjustments between 3V3 and 0V. But when a 3V3 CV is applied to the input, then there is effectively now a voltage divider happening which reduces the peak-to-peak range of the input signal by half and then moves the DC bias between 0-1.5V through to 1.5-3V (or there abouts). Given the limited voltage range anyway, I really wanted to be able to pick up the full 0-3V3 range of the CV if possible.

If all I want is attenuation then one way to achieve this would be to use a switched jack, where the switch will link the pot to the CV when a jack is inserted and to 3V3 when it isn’t, but I wanted to be using jumper wires with no switches.

I’ve gone for summation of the signals, so in the end I decided I’d just do it in software and wire the pots in directly to an analog input then construct a pot-less CV input stage.

As the CV input will be limited at 0-3V3 it will need to be protected against over and under voltage. I’ve seen a range of approaches to achieving this, but in the end I took the basic principles from the modules designed by Hagiwo, resulting in the following circuit:

As I understand things, for the CV input, the BAT43 zener diodes will “clamp” the input CV to GND or 3V3 in the case of under or over voltage; whilst the values of the resistors (1K/680K) mean that there is a sensible impedance but little voltage change. In some cases, I’ve seen the use of a voltage divider here to change a 5V signal into a 3V3 signal. That might be ok if the microcontroller then still treats the 3V3 signal as a “full range” (5 octave) signal, but in my case I want to preserve any 1V/Oct input on any intermediate signal, even if that then leaves me with a smaller three octave+ (0 to 3V3) range.

PWM Output Filtering

Each PWM output will be in the 0-3V3 range, at typical audio frequencies, so a simple low-pass filter will be added as follows:

This is a simple one-stage low-pass filter with a 3db point of just under 5kHz which hopefully gives a nice smooth output. I tried 470, 680 and 1K resistors and 68nF and 100nF capacitors and couldn’t see a significant difference at my test 440 Hz signal, so went with the lower values for a higher cutoff to support more audio frequencies of notes.

The LFO will have two pot controls – rate and depth (amplitude) and no external control. It will provide a triangle and saw PWM output.

Note: If the capacitor is left off the square wave signal then a sharper square wave is possible, so I’ll leave these as optional on the circuit board itself, but will probably leave them out in my build. This makes sense as the sharpest square wave would be a continuing series of odd harmonics, so if I’m filtering out any above 5kHz that will take away quite a few!

In fact, as already mentioned, in code it probably makes sense for the square wave to be a tone() signal rather than a PWM output anyway, in which case the sharpest signal would be with no resistor either, but I suspect some resistance is necessary to protect against whatever the pin ends up driving. Keeping with 470 will match the other outputs, but an even sharper signal is possible with 220 instead. I’ll stick with 470 to keep everything matched. I will be using tone() rather than PWM in the code.

Envelope Generator

The gate and trigger connections for the two envelope generators will allow external connections. This could be a button or an external active HIGH signal. To get these into the microcontroller, requires a circuit like the following (I’ve found various versions of this online, but I think the definitive version came from Rich Holmes – at least on the forums I was looking at anyway):

Again, as I understand things, the use of a diode and transistor provides over and under voltage protection, the resistors provide current protection, and the 1M to GND resistor ensures the input isn’t floating if unconnected.

One issue with this circuit is that the signal to the microcontroller is inverted – i.e. it will be default HIGH until the gate is activated in which case it will become active LOW. The code in the microcontroller has to take this into account.

There will be four of these circuits, one for each of the gate and trigger inputs for each of the two envelope generators.

A manual gate or trigger can be built from a switch that connects the EGn_EXT_yyyy signal to 3V3, with a suitable capacitor for debouncing.

Other External Signals

An external connection will be provided for MIDI IN which will be my standard 3V3 compatible, H11L1 optoisolator based MIDI circuit, so no surprises there. I’m not providing MIDI out.

This will go directly into the ESP32’s RX pin for UART 0 and processed in software to generate an internal CV/gate/trigger signal. This CV/gate/trigger won’t be visible outside the microcontroller – it will be the one part of the system that remains entirely within software. This means that MIDI should extend to a full range of pitches, not just those within the 0-3V3 1V/oct range.

There will be no special hardware for USB, so if USB MIDI is supported in the end, then it will be via the ESP32’s built-in USB port, but that might complicate the power circuitry.

VCA and Audio Output

As already mentioned, the last stage will make use of the TD7052A combining a VCA and output stage amplifier in one. This is basically the circuit I ended up with from my experiments.

The TDA7052A is designed to drive an 8Ω speaker directly from its differential outputs pins 5 and 8), but I’ve also seen it used with a single output to send to a line socket, so I’ve tried to support both here.

Ideally I’d have been able to use a switched jack socket to disable the speaker when a jack is plugged in. Instead, this is another area where there is a compromise and people will have to use one or the other only.

The resistor values on the input have been chosen to attempt to match the input ranges required of the TDA7052A, so ~0-1.2V for the control voltage and ~0-300mA for the audio input. This is the point where the audio signal finally gets the DC bias removed and from a 0-3V3 audio signal becomes a +/- 150mA signal instead. But recall that the TDA7052A will then internally bias it to around 2.6V. Also recall that the VOL signal will sit at 1.25V until pulled low by the transistor and incoming control voltage. One again I’m using a BC557 (PNP) transistor here.

The output is mono and will eventually produce around a +/- 400mV output at the socket when powered from a 5V or 9V supply.

From my experiments it seems to output a 1.5-2V signal with an 8Ω loudspeaker across pins 5 and 8 when powered from 5V. It was slightly increased with a 9V, but the signal started to distort.

To be honest, I’m not convinced of the utility of the speaker output, so it may be that if a speaker is required it will be better taken from the line out and through a secondary small amplifier of some sort. Or it may be that prototyping this section on a breadboard is a little futile – I’ll have to wait until it is on a PCB and try again.

Power and Battery

This has been left about as simple as it is possible to be:

All connectors and switches have been left with pin headers. The 7-12V header could be connected up to a 9V battery if required. The power switch can be omitted by jumpering across J9. This is fed into a LM7805 to produce a steady 5V VCC supply.

Note VCC will be fed into the ESP32’s VIN pin to create the 3V3 power signal that supplies the rest of the circuit. It will also power the TDA7052A directly.

Also note that if nothing is plugged in here (i.e. neither battery or DC), then it should be possible to power the unit from the ESP32’s USB port. If this is the case, then VCC will come from VIN from the ESP32 and the jumper J2 should be removed to disconnect VCC from the LM7805. In this case the ESP32 VIN (which comes from the USB 5V) will also be powering the TDA7052A.

According to the schematics I’ve found online for my dev board (which seem to agree with what I can trace out on my board) the on-board power circuit is as follows:

I’ve found some discrepancy in the capacitors values – another schematic suggests C1 here is 100nF. More seriously, I found one schematic that shows VCCUSB connected to VIN on the NCP1117 rather than VIN. I’m pretty sure this is a mistake and it certainly doesn’t match what I have on my board.

Taking VIN on the module pinout as being the input to the NCP1117 and with a maximum dropout voltage of 1.2V (as per the NCP1117 spec) then the input range for VIN should be 4.5V up to 20V.

Powering it from the 7805 should be perfectly adequate and in fact could probably have been omitted. But in the interests of providing some stability to the PCB when in use, along with wanting a known voltage for the amplifier, I’ve opted to include the regulator. It also means I’ll be able to test parts of the board without requiring the ESP32 plugged in, and might provide an element of protection for the ESP32 should an incorrect power supply be plugged in.

But it should be quite possible to leave it out and direct VIN directly to the input power connection if required.

There is no protection for using USB and powering the system from the regulator, so this has to be decided as a build option to be one or the other, but being able to use USB during testing may be useful.

Panel Design

I’m hoping to get all this within a 100×100 footprint for ease of manufacturing. Initial efforts seem to show that it will be possible. This is the panel design I’m going for.

Design principles:

• MIDI, power, audio output, and any external links for breadboarding will be left off the main panel – the idea being that they will probably be in the side of a case. Or they could be an additional panel element if required.
• I’ve deliberately not included a filter stage – that will be an area where breadboard experimentation will be encouraged! It can be included by taking the output of one of the VCOs and reconnecting to the input of the VCA.
• I might double up on some of the connection points to allow several patch leads – e.g. to make it easy to link to a scope.

PCB Design

I have a PCB design ready, further details available below:

• ESP32 WROOM Educational Modular Synth Thing PCB Design
• ESP32 WROOM Educational Modular Synth Thing PCB Build Guide

Closing Thoughts

I’m writing this after some experimentation and initial breadboarding, so I’m hoping much of this is from a position of knowledge (although obviously not enough knowledge to have anticipating the strapping pins issue!).

The only compromise I think I’ve had to make is that I’ve left out a filter stage. At some point, if this rather mad contraption actually works, then I might consider designing a second panel with some additional modules and that might include some filter options.

Kevin

https://diyelectromusic.wordpress.com/2024/05/07/educational-diy-synth-thing-part-2/

#adsr #envelopeGenerator #esp32 #lfo #oscillator #vca #vco

Following on the back of my ESP32 DAC Envelope Generator and in particular my note at the start that it was essentially the code algorithm and none of the electronics that might make it useful, I started to try to find the simplest possible circuit that Just Might Work as a voltage controlled amplifier (VCA) for demonstration purposes. It turned out to be quite a “rabbit hole”. This post details where I ended up.

But first, just to be clear, I’ll repeat my warning from last time – don’t hook this up to anything else unless you really know what you are doing (unlike me). This is my fumbling around with a little knowledge, largely being “dangerous”, based on random circuits I’ve found online. If that isn’t enough to scare you off, I’m not sure what is…

Warning! I strongly recommend using old or second hand equipment for your experiments.  I am not responsible for any damage to expensive instruments!

If you are new to electronics and microcontrollers, see the Getting Started pages.

Introduction

I’m after a voltage-controlled amplifier (VCA) that can be controlled with my 0-3V3 signal and that could be used to shape an audio signal. This can be thought of as a “black box” with an audio input and output and a control voltage.

But of course there are a large number of considerations if one is to do this properly.

So naturally I threw all that out the window and started searching online for “simple VCA circuit” to see what comes back.

I’m not after any specific quality of VCA, I just want to see the art of the possible with regards to fewest components, simplest design, and useful demonstration of the envelope generator working.

The designs I’ve found can be categorised largely as follows:

• Opto-electronic based designs – e.g. optoisolators, vacrols or similar.
• Transistor based designs – often paired with diodes.
• Op-amp and amplifier based designs – these are starting to get more complex, but seem to produce some pretty usable designs from what I can see.
• Dedicated chips – e.g. the AS3330, AS3360 and compatible devices.

I’d suggest that I’ve probably listed these in order of simplicity (simplest first) which also seems to be inversely proportional to quality of output (that is, the best is last).

Opto-electronic Based Designs

These had instant appeal to me as they appear to be pretty simple on the face of things. Essentially the control voltage modulates the light emitting side of an optoelectronic device such as an LED; which in turn modulates a light sensing side which controls the level of an audio signal.

Here is one of the simplest designs I found using an optoisolator: https://www.reddit.com/r/synthdiy/comments/lreytp/simple_vca_in_eurorack_format_on_stripboard/
This includes a circuit for an LED to indicate activity, which I wasn’t so bothered about, so I’ve simplified it further to the following:

This is about the simplest thing I’ve found and it does kind of work to a degree. I found myself with some cheap optoisolators so used those. The key points to consider for this:

• The resistor on the CV input has to limit the current through the optoisolator’s LED.
• The optoisolator’s LED will only start conducting when the CV input reaches the LED’s forward voltage.

I used a PC817 and PC814. The only difference as far as I can see is that the 814 allows for AC operation, incorporating two diodes in the input side.

For both, the forward voltage of the LED is nominally 1.2-1.4V, which is almost half the CV level! This means that attempting to use the envelope generator “as is” significantly reduces the effectiveness of the levels.

There might be a way to bias the input signal, but I’ve not found a simple way in electronics to do this. Ideally, the voltage range for the input would end up in the region of 1.2-4.5V (i.e. up to 3V3+1.2V), but I don’t know how to do that myself for a slow changing voltage like this – at least not without getting more complicated (and probably involving OpAmps) I must admit to being seriously tempted to give this idea a go as suggested by “Chip” on Mastodon as a very crude possibility, but I’m not sure how the DAC would like that…

Another option might be some simple amplification so that the voltage drop across the LED is a smaller fraction of the whole range.

One simple way to avoid the additional electronics is to introduce an offset in the code so that the output of the DAC is in the range 1.2-3.3V, so that is what I’ve done for now and that seems to work pretty well.

On the oscilloscope trace below, I have the output to a 8Ω speaker. We can see that it is ridiculously noisy… but it does work! It sounds a lot better than it looks and there is no filtering or processing going on other than what is shown in the circuit above.

Using a minimum ATTACK_LEVEL of 70 seems to ensure there is little audio leaking, but it is possible to hear where the voltage drops below the forward voltage of the LED – there is a slight blip. In the photo below it is clear to see the non-linearity of the response (top) compared to the control voltage (bottom), especially in the rising ATTACK stage.

The audio in this case is the ESP32 generating a 440Hz tone using the Arduino tone() function, so it is a 0-3V3 square wave being generated.

Most designs using optoelectronics tend to use a VACTROL. This is effectively an LED glued to a LDR in a sealed enclosure. They can be bought or made (apparently VACTROL is a brand name, but that is what everyone calls them).

Here are some circuits and tutorials using a VACTROL as a VCA that look quite interesting and probably worth following up:

• Kirstian Blastol’s Modular in a Week video “Vactrol VCAs and CV Attenuators”. This video shows how to build your own vactrols using a variety of methods. His “Schematic_Vactrol” is essentially the same circuit as shown above for the optoisolator…
• Benjie Jiao’s “Passive Vactrol VCA” which is a very similar circuit again, described as a “low pass gate”.
• “Voltage to resistance” from thesquarewaveparade (search for thesquarewaveparade VtoR.jpg) has a range of increasingly complex, but still relatively straightforward to understand circuits for a voltage-controlled resistance.

But the best discussion on how to make best use of a Vactrol, and hence by inference (by me) probably also an optoisolator can be found here:

• Rich Holme’s “How to Vactrol”.

I just used exactly the same circuit as for my optoisolator, but the Vactrol I’m using seems to have a larger forward voltage. It is marked VTL5C9, but I’m not sure it’s an “official” Xvive one – but if it is, then the datasheet states a forward voltage of 2.5V which is rather a lot when my control voltage tops out at 3.3V!

To be any use at all, I’ve had to set the minimum ATTACK_LEVEL to 110 which has essentially halved the resolution. But once again, it does work for some definition of “work”.

The longer lead is the anode for the LED side – then I’ve just used it in place of the optoisolator.

Transistor Based Designs

A number of simple VCA designs use a single transistor or a transistor and a diode. I found this one being discussed online so gave that a go:

As far as I can see this is sort of “reversed” in the sense that rather than the control voltage controlling the audio, to me it looks like the audio is controlling the control voltage signal… the end result is still audio only when the control voltage enables it though.

I didn’t have a BC549B transistor, which I believe is a common low-noise NPN transistor, so I used a 2N3904 instead. Note that the pin assignments are different between the two!

One of the advantages of this design appears to be good tracking of the envelope as can be seen below. The actual output level isn’t very high though – the blue scale is 200mV compared to the yellow’s 1V below. I suspect that is a result of attempting to drive the 8Ω speaker directly. I guess this would need to go into some kind of buffer or amplifier stage with some sensible impedance matching at the very least, to be particularly useful.

There is some discussion of this circuit here: https://www.modwiggler.com/forum/viewtopic.php?t=168810. I believe it is only working in the simple case shown above as the audio signal is biased to a 0-3.3V signal rather than a +/- signal.

A bit further down in the linked thread there is an example of how to cope with an unbiased input audio signal by adding a DC bias to the transistor base, so I tried a variant of that as follows:

As I’m using a 3V3 control voltage, I’ve used 680kΩ and 100kΩ to bias the transistor at around 420mV. A coupling capacitor removes any existing DC bias to the audio signal prior to feeding it in.

The final signal at the loudspeaker is pretty low as before, but if I take the output from just before C1 it is looking pretty good to me although we can see the output is still positive only.

Once again, the trick now would be to buffer or set up the load on this signal to preserve it as an actual audio out.

This is pretty crude, but then the discussions implied nothing less. But for a handful of largely passive components, this does seem to show some promise.

For a proper discussion on the workings of transistor based VCAs, I can recommend:

• Moritz Klein’s “Designing a classic transistor-VCA from scratch”.
• AudioPhool’s “Single Transistor Voltage Controlled Amplifier (VCA)”.

OpAmp Output Buffer

For completeness I briefly experimented with adding a LM358 OpAmp based buffer (or “voltage follower”) circuit as follows:

This is connected instead of the speaker in the previous circuit so the input is coming from the right hand side of the 100nF capacitor, so the input is unbiased. But the OpAmp is running from 5V and there are two resistors on the input which adds a 2.5V bias.

The output is pretty good, but I can see the top of the envelope being clipped in the positive direction. I’m not entirely sure why – I thought powering it from 5V (rather than 3V3) would give enough headroom for a decent signal, but maybe not.

Also my limited electronics knowledge is failing me in understanding how the signal shown above (measured at the collector output for the transistor) becomes the signal shown below (measured after the final capacitor in the buffer circuit).
Once again it is probably something to do with my lack of understanding of output impedance and related topics.

It might work better with a LMV358 rather than a LM358 as apparently the LMV version runs better at lower voltages with output much closer to the power rails (“rail to rail”). It might also be possible to adjust the bias resistors and re-align the signals somehow…

Other Transistor Circuits

I also found a couple of circuits where the control voltage is fed into the base of the transistor to control the audio passing through it in a more “traditional” (or at least, “expected” by me) manner. But I wasn’t really able to get them working, so I didn’t take those any further. One I never actually tried as the LMNC “painfully simple VCA” which is essentially just a diode and transistor.

I also found a circuit describing a “swing VCA” but again my initial experiments didn’t seem to give me anything useful, so I’ve largely ignored those too.

OpAmps and Amplifier Based Designs

In one of the discussions I found talk of a TDA7052A in use as a VCA. It was posted by elektrouwe in the discussion here: https://electro-music.com/forum/topic-63383.html&postorder=asc. Further on in the discussion was another circuit, posted by Hammer.

The original TDA7052 is a 1W mono audio amplifier apparently designed for battery led operations and to be powered by 3-18V. The TDA7052A is an upgrade that adds DC voltage volume control, but I believe the power requirements have increased slightly to requiring a 4.5-18V source.

A good discussion for how to use them can be found here: https://electro-dan.co.uk/electronics/tda7052.aspx.

In particular, note that elektrouwe states:

“TDA7052A has a gain of ~ 56x, which means Vin should be in the 100mV range, otherwise you need an input voltage divider. You MUST use an input coupling cap., because the chip generates an internal DC bias voltage.“

Pretty much every other example circuit I’ve seen shows the two outputs (pin 5 and 8) connected directly to a 8Ω loudspeaker, but the design above is only using one of the outputs. The two outputs apparently provide an inverting and non-inverting output option.

Here are some key points that might be relevant to its use as a VCA (from the “TDA7052A/AT: 1 W BTL mono audio amplifier with DC volume control” datasheet, dated July 1994, sometimes braded Philips, sometimes NXP):

• “The maximum gain of the amplifier is fixed at 35.5 dB.”
• “The DC volume control stage has a logarithmic control characteristic.”
• “The total gain can be controlled from 35.5 dB to −44 dB.”
• “If the DC volume control voltage is below 0.3 V, the device switches to the mute mode.”
• Positive supply voltage range: 4.5V to 18V.

From the graphs in the datasheet (shown below), the response appears pretty linear when the DC volume control voltage is between 0.4 and 1.2V.

So I think the summary is that to use this as a VCA then the control voltage needs scaling to the 0.4-1.2V range (or thereabouts) and the input signal needs to allow for a maximum 35.5dB gain at the high end of that range. The article I linked to earlier describes the original TDA7052’s 39dB gain as equivalent to a voltage gain of 90 times, so to keep within the 0-5V presumed output range, that would require an input audio signal in the 50-100mV range.

I took an approach that largely used elements of each circuit. From here, we are told:

“The DC volume control is at pin 4. The TDA7052A produces a voltage of around +1.125V at this pin, as well using the current at this pin as a volume reference.”

So I thought the approach of using a PNP transistor to modulate that according to the control voltage was probably the way to go.

To get my CV (which is in the 0-3.3V range) down to 0-1.2V I used a voltage divider of a 2MΩ and 1MΩ resistor dropping the range down to approx 0-1.2V. To keep the audio signal within a sensible range to allow for the full gain, I used another voltage divider, this time using a 1MΩ and 100kΩ resistor dropping my 0-3.3V test signal down to around 0-300mV, which is perhaps still a little high, but it was fine for a test. Any final circuit may have to account for a line-level audio input (probably) so will need to be adjusted accordingly.

Note, from my experiments, the audio input (pin 2) seems to be internally biased at around 2.6V. This was with either a 3V3 VCC or 5V VCC.

So here are the components used:

• 1x TDA5072A
• 1x BC557 PNP transistor
• 1x 1KΩ resistor
• 2x 1MΩ resistors
• 1x 2MΩ resistor
• 2x 1uF electrolytic capacitors
• 1x 10uF electrolytic capacitor

The circuit is designed for a 0-3.3V CV input and is powered from a 0-5V supply.

And here is my test circuit:

And this seems to work fairly well. In the following trace (I’ve swapped the traces over – blue is now the CV and yellow the output), we can see the effect of both the logarithmic response of the DC volume control of the TDA5072A and possibly some maxing out of the volume when at full (I’m not sure tbh).

But it certainly sounds convincing to me and is perhaps the most promising solution so far.

The LM13700 Transconductance Amplifier

The LM13700 appears to be a massively useful circuit in synthesizer designs! I’ve found it used in at least three designs online and it has a chapter of its own in “Make: Analog Synthesizers” by Ray Wilson.

I’ve not got any at the moment, but here are links to the designs and some further discussions about it:

• https://benjiaomodular.com/post/2021-12-17-lm13700-vca/
• https://electricdruid.net/design-a-eurorack-vintage-vca-with-the-lm13700/
• https://www.davidhaillant.com/simple-vca/

This will be one to come back to.

Closing Thoughts

This was a bit of a diversion, and really I was after the cheapest, simplest, VCA I could manage to build.

I really like the simplicity of the optoisolator/vactrol approach, and for higher signal voltages I can see that it would work well, but for 0-3.3V signals the non-linearity from the use of the LED is just too great. If I can find a simple way to add ~1.5V to the signal then this would probably work pretty well.

The transistor designs were very encouraging, but I suspect I’d really need a little more electronics knowledge to be able to sensibly use them in a circuit.

The TDA7052 is the easiest for me to understand at present, but I would like to return to the topic if I managed to get hold of some LM13700 equivalent devices at some point.

One group I never mentioned is dedicated VCA/EG chips, like the AS3330/33360 or the CEM2164. Again this is probably a subject to return to at some point, but I’m getting quite well past simple, cheap, lo-fi at that point – these are used in proper modular synth designs.

Just a reminder – none of these circuits have any input protection, there is no buffering, and no thoughts of impedance – they are literally the bare bones that might have a hope of kind of working (if at all). They are just part of me learning how some of these things work.

They are not intended for any real use and certainly not for use with any equipment that isn’t disposable.

Kevin

https://diyelectromusic.wordpress.com/2024/04/20/esp32-dac-envelope-generator-part-2/

#envelopeGenerator #esp32 #lm13700 #optoisolator #tda5072 #vactrol #vca

I’m continuing my series of experiments with the ESP32 by considering how I might use the twin DACs onboard the WROOM module as a Lo-Fi, 8-bit envelope generator. I’ve not looked at envelope generation before, so this is a good excuse to see what it is all about.

Important Note: This is NOT an envelope generator circuit or standalone device at present. It just outputs the waveform to the DAC. There is no electronics here that would make that a usable signal in any kind of controlling manner at present. This is mostly thinking about the code to produce the waveforms.

In short, don’t hook this up to anything else unless you really know what you are doing (unlike me).

Warning! I strongly recommend using old or second hand equipment for your experiments.  I am not responsible for any damage to expensive instruments!

These are the key tutorials for the main concepts used in this project:

• Arduino ESP32 DAC
• m0xpd’s “Signal Processing on the Arduino” and ADSRduino
• Mozzi ADSR implementation

If you are new to microcontrollers, see the Getting Started pages.

Parts list

• ESP32 WROOM DevKit
• 4x or 8x 10kΩ potentiometers
• 2x 1kΩ resistors
• 2x push/toggle switches
• Breadboard and jumper wires

The Circuit

I’ve ended up wiring potentiometers to eight analog inputs, buttons to two digital inputs and put my oscilloscope on each of the DACs to see the output.

The potentiometers are wired in the usual VCC-signal-GND manner (although only one is shown above). The buttons are pulled down as the signals are meant to be active HIGH signals.

The Trigger input is a pulse indicating when a key would be pressed and signifying the start of the envelope generation. When triggered the Attack stage of the envelope will begin immediately followed by the Delay phase. The Gate input is held and meant to indicate while the key is pressed and when it is released. Whilst on, the envelope will remain in the Sustain phase. On removal of the Gate signal the Release stage of the envelope will start.

Note that the plan is for the Trigger to allow retriggering of the envelope at any time and that for removal of the Gate can also happen at any time and start Release. It is also quite possible for there to be several triggers whilst the gate is still active.

It is also possible for the trigger and gate pin to be the same in which case trigger happens on the rising edge along with gate ON and gate OFF will happen on the falling edge.

Here is the full GPIO list for this experiment.

I’ve used the same GATE and TRIGGER signals for both envelope generators, but it would be quite happy with four independent inputs.

Everything here is working with 3V3 logic levels, including the envelope voltages produced.

In the photo below I’ve simplified my wiring by using my Analog IO Board PCB to give me eight potentiometers directly wired into the ESP32.

Envelopes in Mozzi

I’ve already used envelopes in my experiments with ESP32 and Mozzi, but they are applied in software to modulate the amplitude of the Mozzi synthesized output. And really, if using a microcontroller for synthesis this is the natural way to do things.

By way of an example, in Mozzi, envelopes are created on startup, have their parameters changed as part of the control loop, are triggered on and off usually in response to note events, and then have each instantaneous value calculated as part of the audio loop an applied to the sample value.

The essence of their use in Mozzi is as follows:

#include <ADSR.h>

ADSR <CONTROL_RATE, AUDIO_RATE> envelope;

void HandleNoteOn(byte channel, byte note, byte velocity) {
  envelope.noteOn();
}

void HandleNoteOff(byte channel, byte note, byte velocity) {
  envelope.noteOff();
}

void setup () {
  envelope.setADLevels(ADSR_ALVL, ADSR_DLVL);
  envelope.setTimes(ADSR_A, ADSR_D, ADSR_S, ADSR_R);
}

void updateControl(){
  IF ADSR values have changed THEN
    call setADLevels and setTimes again as required
}

AudioOutput_t updateAudio(){
  Calculate new 8-bit sample
  return MonoOutput::from16Bit(envelope.next() * sample);
}

All that would be required to get this to output just the envelope would be to change the return statement in updateAudio to return the envelope value directly.

AudioOutput_t updateAudio(){
  return MonoOutput::from8Bit(envelope.next());
}

There are several more example sketches in Examples->Mozzi->07.Envelopes.

There are several issues with this approach that stop me using this for what I want to do:

• This only supports one output. I might be able to configure two envelopes and get one output on the “left” channel and one on the “right” channel, which I think then map onto the two DACS…
• I want to integrate this with some of my ESP32 PWM messing around too, which isn’t easy when Mozzi is determining all the outputs. There is an option to use a user-defined function for the output, but at this point I’m doing a lot more myself anyway…

And anyway, I wanted to work out how an envelope generator could be implemented myself. So I didn’t use Mozzi and got to work on my own implementation.

DIY Envelope Generation using Timers

I had an initial look around at any existing envelope generator implementations for Arduino, having a look at both ADSRduino and the Mozzi ADSR implementation.

In the end I opted for a simpler design of my own, deciding to manage the ADSR as a state machine in code with calculations for how much the envelope level needs to change per tick of a timer. I’m just implementing simple linear updates for each stage.

Setting up the timer is the same as for PWM, but this time I’m using a 100kHz timer with an alarm every 10kHz. This gives me a 0.1mS “tick” which is more than adequate for generating an envelope.

I’ve opted to map the potentiometers onto the ADSR parameters as follows:

• ADR are mapped using: 1 + potval * 2.
• S is mapped directly to a value in the 0..255 range, reflecting the 8-bit DAC output.

The time values are in units of 0.1mS so can specify a duration for any of the three stages between 0.1 and 819.1 mS. For pragmatic reasons, when using these values in calculations, I always add 1 so I don’t ever have a divide by zero (which causes the ESP32 to reset).

All values relating to a level are in 8.8 fixed point format, so are essentially 256 times larger than they need to be to give more accuracy in calculations.

The ADSR state machine has the following functionality:

Idle:
  Do nothing

Trigger:
  Start Attack

Attack:
  Increase level to maximum from current level one step at a time
  IF level reaches maximum:
    Move to Delay

Delay:
  Decrease level to sustain level one step at a time
  IF level reaches sustain level:
    Move to Sustain

Sustain:
  Stay at same level while Gate is ON
  IF Gate is OFF
    Move to Release

Release:
  Decrease level down to 0 one step at a time
  IF level reaches zero
    Move back to Idle

For each stage I maintain a step value, which is how much the level has to change for that specific step. This is calculated as follows:

Num of Steps for this stage = Time of the stage / SAMPLE RATE

Usefully, if I’m measuring the time of the stage in mS then I can use a SAMPLE RATE in kHz and the calculation still works. So the step increment itself can be found by:

Step increment = (Required end level – Starting level) / Num steps

Step increments can be positive or negative of course depending on whether the output is rising or falling.

As already mentioned I’m using 8.8 fixed point arithmetic for the levels. The biggest concern was watching out for automatic wrapping of the 16-bit values whilst performing calculations, so I’ve removed that as a possibility by using signed, 32-bit values for the step increment and stored level.

All the parameters associated with an envelope are stored in a structure:

struct adsrEnv_s {
  int32_t  env_l;
  int32_t  steps;
  uint16_t attack_ms;
  uint16_t attack_l;
  uint16_t delay_ms;
  uint16_t sustain_l;
  uint16_t release_ms;
  bool gate;  
  adsr_t state;
} env[NUM_DAC_PINS];

And there are a number of functions for manipulating the envelope. This is the point where really I ought to be branching over into “proper” C++ and making this an object, but I’ve stuck with C, structures and arrays for now.

The final implementation has a few extra steps in the state machine corresponding to the transitions between stages. This just makes calculating the new step values clearer at the expense of adding an extra timer “tick”‘s worth of processing time to each stage.

Two complications come from how the gate and trigger need to be handled.

The gate has to be checked in each of the stages and if the gate goes to OFF then the state needs to switch over to Release.

The trigger needs to come externally to the main state machine, but in order to ensure I’m not attempting to update variables at the same time that that they are being manipulated by the interrupt-driven state machine function, the trigger just updates the state to a “trigger” state so that on the next tick, the state machine will update itself.

The full set of states recognised now stands as follows (stored roughly in the order they progress through):

// ADSR state machine
enum adsr_t {
  adsrIdle=0,
  adsrTrigger,
  toAttack,
  adsrAttack,
  toDelay,
  adsrDelay,
  toSustain,
  adsrSustain,
  toRelease,
  adsrRelease,
  adsrReset
};

There is the option of configuring a timing pin so that both the time within the interrupt handler, and the period of the timer can be checked.

There is also a TEST option that manually triggers different stages of the ADSR and dumps the level of one of the envelope generators out to the serial console. This makes tweaking and debugging a bit easier.

The main loop handles the IO updates:

Loop:
  FOREACH DAC/EG:
    Read Trigger pin
    IF Trigger pin goes LOW->HIGH THEN
      Trigger ADSR

    Read Gate pin
    IF Gate pin goes LOW->HIGH THEN
      Turn on ADSR Gate
    IF Gate pin goes HIGH->LOW THEN
      Turn off ADSR Gate

    Scan each pot and update ADSR if changed

Here is a trace of both envelopes with different ADSR values running off the same trigger and gate:

Find it on GitHub here.

Closing Thoughts

To get any practical use out of this will require some electronics. I can’t just hook the DAC up to something else and expect everything to place nicely, so that is something to consider next.

But for now, although the code is more complex than I original thought, thanks to having to handle the interplay of triggers and gates, it seems to work pretty well.

Kevin

https://diyelectromusic.wordpress.com/2024/04/07/esp32-dac-envelope-generator/

#dac #envelopeGenerator #esp32 #mozzi

After some minor adjustments, I now transferred it onto a soldered #perfboard.
Short demo video: https://makertube.net/w/jiuDuVyXH15G8uXPVBcdno
#soldering #electronics #diysynth #555timer #NE555 #ADSR #envelopegenerator

#prototyping an #ADSR #envelope generator on a solderless #breadboard for my #diysynth experiments. I used René Schmitz' design as reference, but using a #bipolar #NE555 instead of the required #CMOS variant, since I didn't have any of those.
I captured the output using my cheap #oscilloscope and used a simple push button.
It seems to work fine, though. I can live with the higher power usage.
Next step: transfer it to a soldered perfboard.
#envelopegenerator #electronics #555 #555timer