Working OMM prototype

I've spent the days since getting back to Norway making an OMM prototype running on an 8 bit PIC18F458 @ 20MHz.

Things are looking very good indeed, except for speed of course. I've implemented:

- node structs
- function pointers and pointer lookup
- matrix array calculation
- DAC output to 14 bit SPI DAC (MAX544
- input buffer reading and outputbuffer writing

In addition I've coded the following node functions:
- Sum
- Invert
- Invert each side (positive stays positive etc)
- Ramp (but without proper interval calculation)
- Delay line (allows loops in network)
- Input
- Output

Next up should be
- Timers for constant output to DAC
- SPI input for controllers (but that is not possible without a second SPI module)
- JSON parsing
- Table lookup for tuning and exponentialization.

Here is the first output from the OMM - the frequency is too low to get a good image on the scope:

A few specifics on the SPI and the DAC:

The DAC accepts a maximum SPI speed of 10MHz. In the tests I did, I ran the SPI at F_osc/4, which means 5MHz when the oscillator frequency is 20MHz. I achieved what I think is a sample rate of about 89kHz but that is without any calculations between samples. In practice then, it seems the maximum DAC rate is slightly less than 180kHz when running the SPI clock at 10MHz (from the PIC that is, it is entirely possible that you can achieve higher speeds otherwise).

Here is  the code for the DAC tryout:

void writeToDac(){
	unsigned int dacout;
	/**** DAC ****/
	SPI1_Init();
	TRISC = 0; //trisc as output
	LATC.B0 = 1;
	dacout=0;
	// The following code runs at about 89kHz.
	while(1){
	LATC.B0 = 0; //must write directly to latch (didn't work with PORTC.B0!)
	SPI1_write(hi(dacout));
	SPI1_write(lo(dacout) & 0b11111100);
	LATC.B0 = 1; //latches values in DAC.
	dacout += 2048;
	}
	}

The first time I got the DAC working it looked like this:

The visible steps are because I chose to use only 32 steps to speed things up and to see what is happening. After removing all delays and maximizing the speed it looks like this:

Here is the SPI clock btw:

On voltages and other practicalities

While dreaming up the OMM I have for some reason constantly kept thinking about the output from the DACs as 5V, with a OMM value of 0 corresponding to 0V and 65536 to 5V while at the same time intending to use a 1V/octave VCOs and having the pitch CV controlled from the OMM.

This, of course, leads to a severe limitation: the synth would not be able to span more than 5 octaves, and no tuning or modulation would be possible for the highest and lowest notes.

There are a few different solutions to this problem:

1) one could use a separate output as octave selector. By keeping this at 2.5V normally, we could add or subtract 2 octaves and still have 0.5V left for tuning and modulation. This will however lead to having to do some tricks when receiving notes outsidere the normal 5 octaves.

2) one can use 10V instead of 5V for the DACs. This gives us the same 10 octaves as adding an octave selector CV. However, this requires a redesign of some modules such as the VCA.

3) one can change the input resistor on the VCOs so their response is 1/2V per octave, again giving us the same 10 octaves.

I am not sure which one to choose. I will have a closer look at how other midi-2-CV converters do this.

I also need to figure out how many octaves to support. If it should be possible to do one-octave pitch bends 10 volts only allows 8 full octaves, and even then other modulations and tuning must be kept separate.

LFOs and modulation

Our numerical LFOs will have an amplitude from 0 to 64k. But when using the LFO to modulate the pitch CV, to get a vibrato effect, we want it to increase the pitch for half the cycle and lower it for the other half. In essence, we would be better off using a signed int. In other cases, and for other operators, we are only interested in the positive parts of the waveform. And in any case the DACs would normally only output a single-polarity signal (though this can be fixed).

Again, I need to check how others are doing this.

Other things to think about when it comes to LFOs is that it should be possible to adjust their starting point and direction (up/down), as well as to choose whether or not to reset the cycle at the start of a note and to sync it to an external clock.

Multiplication vs attenuation/scaling - virtual VCAs

I've realized that we need an additional function, one that treats the inputs like they are between 0 and 1 when multiplying them. This is analogous to how a VCA works, when the CV is at max, the output is the same as the input. When the CV is half of its Max, the output is attentatet by 50% etc. In other words, the output is scaled depending on the CV, between 0 and 1 times the input.

Ramp generation

If the running time of the matrix calculations is constant and triggered by a timer, generating a ramp is simply the task of adding the same number to an accumulator variable every cycle.

But since the number added is never exactly the number necessary for a correct ramp progression (we are dealing with integers after all), we will accumulate an error. This may be ok, but we should keep control of how big the error may become and take actions to prevent it from becoming too big.

If we work with 16 bit numbers, we will get an error of maximum 1/65536 per cycle. But assuming we are running at 44.1kHz, this will accumulate to an error of 44100/65535 = 67% in one second! Not very accurate I'd say.

Lets do the cslculation the opposite way. If we decide that yhe maximum acceptable error is 1% per second, what resolution do we need for our interval?

Well, in one second, we should be at most 0.01 * 65536 = 655 away from the correct answer.  At 44.1kHz, this means that the maximum error per cycle must be less than 655/44100 = 0.015, or 1/67th of the error we get from using an integer.

To remedy this, we may store the accumulator and the value to add as a number with more bits (in doing this, we will multiply the interval, or rather, calculate it differently) and divide the accumulator down to 16 bits when the output is needed.

To get the necessary precision we need to be able to store 1/67ths. By adding 6 bits to the number we can store 1/64 fractions, which is not quite 1/64 but may be good enough, or we can add an additional two bits and get a 1/256 precision, making the accumulator and interval variables 24bit (it may be practical to make them 32bit while at it, not sure).

Converting the accumulator back to 16bit is simply a matter of shifting it 8bits to the right.

Looping or multi stage ramps (envelopes)

When the ramp reaches the desired height, we want to stop adding to it, reset it or even start subtracting from it. This means that every time we have added something to the accumulator, we should check its value and possibly perform an action. We may even have to keep track of state, as what stage of an envelope we are in. The state, as well as the accumulator and value to add should be stored within the ramp node's struct if we want a truly node-type agnostic algorithm that lets us have any number of any type of node (unlike the Matrix 12 which has a fixed number of ramps, LFOs etc).

Maximising clock speeds

Just a short note on chrystals, clocks and instructions. On the PIC24 and PIC32 you have very high maximum MIPS, million instructions per second, counts. To reach these speeds, you should not use an extremely fast chrystal. Instead, you use what is called a PLL, phase locked loop module that is found within the mcu. This expects an input clock of between 4 and 5MHz, but multiplies this to get a much higher internal frequency. You may use a faster external chrystal, but this has to be divided down to be in the interval that the PLL expects by a prescaler (also found in the mcu).

Polysynth voice controller

The hardest thing right now is getting hold of paper. Fake t-shirts, booze and other crap, no problem, but finding a decent notebook is nearly impossible in the town of Patong. I found this at last (the killer bunny in the background is the daily art left by the cleaners):

So, we've looked at some of the tings we want to do on the voice card mcu, but what mcu should we use?

We have some initial requirements:
- dual SPI/I2C interface
- data flowing through the matrix should be 16 bit
- we want the run-through of the matrix to be fast to achieve a high output update rate, maybe as close to 44.1kHz as possible.

This last requirement may not be possible with "home made" samle and hold outputs but let's for the moment assume it is.

So far i have almost exclusively worked with the Microchip PIC16 and PIC18 MCU families. These are 8 bit, but it is still possible to work on 16 bit numbers. However, when it comes to multiplication of 16 bit integers, tings start taking a loooong time. In our matrix we want to do a lot of multiplications, so this is bad. To speed tings up we may precalculate a lookup table, but this would have too many entries to be possible, as the amount of ram available is very limited"

Another limitation is the maximum speed possible. From memory I think the maximum clock frequency for the 18F is around 40MHz. But each instruction takes 4 clock cycles, so the maximum speed is 10MIPS (million instructions per second). If we are aming for 44.1kHz, this leaves us with 226 instructions per refresh! Takling into account that multiplications take a lot of these and you quickly realize that the PIC18 doesn't quite cut it. I know the Matrix 12 probably does not have a lot more power than this, but it used 8 bit variables. If we did the same we could possibly make tings work, but we won't.

If we want to stick with the PIC family this leaves us three options - the dsPIC and PIC24, both 16 bit, and PIC32 which is 32 bit.

The PIC32 runs at up to 200MIPS, 20 times faster than the PIC18F. It packs a whooping 512kB of ram and has plenty of serial i/o and whatnot. It even has built in support for ethernet. Also, it can do 16bit x 16bit and 32bit x 16bit multiplications in 1 instruction! With 200MIPS we get 4535 instructions per update, so we should be able to use a quite large matrix.

So what are we waiting for? The PIC32 wins the shootout, right? Well, there are two things worth considering. First of all, the price. A PIC32 running at 200MIPS and with 512kB Ram costs close to $9 in quantities of 5000 and up. Not something that breaks the bank, but considering we need one per voice, it gets a bit expensive.

More importantly though, the PIC32 comes in surface mountable versions only. This makes it harder to prototype things. We may of course add in-circuit programming posibilities (and this is such a good idea that I would consider doing it any way) but it still is a bit of a hassle.

That leaves us with the PIC24 and the dsPIC. These are so similar that they share the same data sheet. The difference is that the dsPIC has a set of instructions specialised for digital signal processing, like doing large sum-products in a single instruction. Right now I see no use for this in our matrix (funny, as matrix multiplication is possibly something the dsPIC is very good at), so lets see if the PIC24 can do the job.

The PIC24 runs at up to 70MIPS, 7 times faster than the PIC18, so we get 1587 instructions per update. Not as much as we may have hoped for but still something. The PIC24 can also do 16bit x 16bit multiplication in one cycle (though i need to get this confirmed). It has two SPI interfaces and it comes in (among others) a 28pin through-hole version.  The cost is  around $3.50, which is a lot cheaper than the PIC32. All in all I am willing to give it a try. Except for having to redo the digital voice pcb much of the code and circuitry from a PIC24 version can be reused for a PIC32 version. One could also reduce the update frequency to get more time between updates. Besides, part of the fun is using a mcu with limited resources, you have to be clever :-)

The OMMNoM

I mentioned in the previous post that I want the synth to be like the Oberheim Matrix line with its software modulation matrix. Here is my take on it: The OMM (open modulation matrix) and its counterpart, the OMMNoM (OMM Node Manager).

The OMM is a piece of firmware running inside the mcu of each voice. It takes a set of inputs, constants, ramps/envelopes and function generators, and through a series of transformations calculates the CVs that control the voice. It is open in the sense that it is possible and easy to add input sources and transformations to the code and it has an open protocol so anything able to generate a simple string stream may be used to create the OMM model.

The OMMNoM is a program used to create the matrix. I envision this as a graphical tool running on a tablet. Through dragging and dropping nodes representing sources and transformations onto a canvas, and then connecting their inputs and outputs to each other and to cv outputs, one creates modulation matrices as advanced as the ones in the Oberheim Matrix line through a much more intuitive user interface.

In the matrix, any output may be connected to any input. Some nodes, such as summing and multiplication accept multiple inputs, while others, for example linear-to-exponential converters and cv outputs only have a single input. All nodes have one output, except the cv outputs which have none (or rather, they have a physical output that connects to the analog part of the voice). An output may of course be connected to any number of inputs.

So far I've thought of the following nodes:

External inputs
- midi controller messages
- keyboard pitch
- keyboard trigger/gate
- pitch bend wheel
- modulation wheel
- physical input switches
- physical input potmeters
- virtual input switches
- virtual input potmeters
- noise input connected to physical noise module of the analog card (use with sample and hold for example).

Controllable functions/function generators
- ramp generators (inc looping?)
- multi stage envelopes (inc looping?)
- LFOs
- Sample and Hold
- Trigger (sends trigger pulse if input is high)
- Comparator - sets output high when one input is higher than the other
- switch - turns on or off a signal when input is high or low)

Mathematical expressions
- sum
- multiply
- divide
- maximum
- minimum
- average
- exponentiation (possibly with dB control)

Logical operators, all with adjustable thresholds (with defaults)
- and
- or
- not
- xor

Outputs
- CV
- trigger pulse (for analog envelope)
- gate (for analog envelope)
- other binary pins (controlling switches etc)

Other
- numerical constants - these may be entered directly on a node input instead of being a separate node.

The outputs will most likely be named after what they connect to on the analog voice card, but this is not a necessity for the OMM model which just maps the output to a pin or DAC address.

Usage example:

To create an echo with diminishing volume,
an LFO may be connected to the trigger and gate input of an envelope, which in turn is multiplied with a down-sloping ramp. The output from the multiplier should be run through an exponentiation and then connected to the output CV going to the voice's output VCA.

Polysynth voice cards

Voices

I intend to spilt the voice cards into two physically separate PCBs, one containing the mcu and DACs for the various required control voltages, and one with the analog parts like VCOs, VCFs, VCAs etc. This way i may develop the two cards separately and change only parts of a voice if i need to. In short, it reduces the consequences should something go wrong or if i want changes. I may even consider a separate filter card.

(Mostly) analog card

At the moment the analog cards should contain the following:
- noise, switchable between white, pink and red, with its own separate VCA.
- three VCOs, each with saw, triangle and pulse outputs. Each waveform should have a separate VCAS.
- one 2 pole state variable filter (HP/BP/LP and possibly notch)
- one 4 pole LP filter
- one sample player/digital oscillator. This can be used for sampled attacks (Roland D-50 style) or digital waveforms.
- cross modulation between oscillator 1 and 2
- sync (soft and maybe hard) between 1 and 2/3
- a separate output VCA connected to the main envelope.

All these elements will be built as separate, chip-style PCBs, once again reducing the complexity and simplifying testing.

I strongly consider making this an Oberheim Matrix-style synth. More about that in a separate post, but this means that I will move LFOs, envelope/ramp generators, sample and hold and other modulation stuff into the digital domain. A consequence of this is that each waveform may possibly have a separate envelope or have its amplitude modulated in other ways.

 A note on the VCAs as well: if possible, I want to keep them linear. This will make them far less temperature sensitive, as well as slightly easier to build. They may also be turned fully off. It does also mean that exponential conversion will have to be done in software. I have to test this to see if it is feasible and if the result is good enough.

In addition to the necessary CV and switch control inputs, the voice should have the following connectors:
- External sound input (possibly with it's own VCA)
- Feedback from noise to mcu for random/sample & hold functionality
- Feedback from VCOs for tuning
- Separate analog ground for each voice card

Filter section

I have thoughts about making it possible to merge two voices into a six (eight) oscillator monster voice to do super saw-like stuff. Originally I only intended a voice to have a single filter, and in the case of voice merging, which would happen before the filter, I would get a spare filter for each voice.

I'm concidering two ways of using this filter, switchable from software:
- in series with the first one, but with separate filter controls
- in parallel, but with separate output VCAs.


As a wrap up for the post, I have sone other ideal and thoughts that may or may not make their way into the voice cards:

- Move the sample player to the digital board to keep nasty digital transients away from the analog board? (Still required for switches, but they will hopefully not be switched in the middle of playing.
- Suboscillators for each VCO
- Analog envelopes for the filter and/or voice outputs. May use both analog and digital.
- Saw polarity switch or separate VCAs for each polarity?
- Filter cassettes to be able to switch between famous filters. Maybe one cassette per voice card?
- FX busses. I want to include at least a juno-106 chorus module, but possibly also a digital effects module.
- ring modulation

Sketches

Polysynth and digital communication

Life is hard. But not right now. We're on a babymoon, a relaxing two-weeks holiday to sunny Thailand preparing ourselves for the stress of having a baby (thanks for the idea, Theresa). Nothing much happens here, so what better to do while sunbathing by the pool than drink a cold beer and dream about homemade polysynths, eh?

Today's topic: ic-to-ic communication. There are three viable options: parallel diy protocols, spi and i2c. They each require different nummer of communication pins, from 8+ for the parallel, 4+ for spi and 2 for i2c.

I2c, while seemingly the best in terms of the number of pins it requires, has a few attributes worth looking at. First of all, each device on the i2c bus has a unique address. While this is often a good thing, there are times when you want to send the same data to multiplum devices. Secondly, the transfer rate is limited, often to 100kHz or 400kHz. This may not pose much of a problem if your microcontroller cannot handle higher speeds anyway, but it is worth remembering.

Spi on the other hand usually has an address bit input for each device. When this is high, the device receives whatever data comes through the pipe. This means that multiple devices may receive the same data at the same time.

So, which one should we choose?

For my polysynth i imagine 8 voices. If possible I intend to control two voices at the same time from one mcu, if not, they will each have their own mcu. I want separate mcu's to be able to have a fully multitimbral synth, as well as being able to combine two voices into a more complex voice.

Using spi for the master mcu to the voice mcu's means I can send the same data to multiple voices at the same time, which is nice when doing polyphony.

For the voice mcu to dac, port expander etc, the choice is not as clear. First of all, if one wants separate external (master-to-voicecards) and internal (within voice) busses, one either needs a mcu with two serial interfaces or either use bit banging or do parallel communication within the voice.

I think i will use a combination of spi and separate pins for the internal communication. For example, i want to transfer data to a sample player/digital oscillator on the voice card. However, i also want to trigget its envelope  or start its sample at an exact time. Using a single dedicated pin for this may give me tighter controll timewise.

The same goes for DACs and shift registers. Data as spi and clocking as dedicated pins. Maybe. I'll have to test this.

Oh well, time for a drink. Iced coffee maybe?

Exponential VCA with temperature correction

As promised in the previous post, here is a new take on the exponential VCA. This time the CV input has been modified and a tempco resistor added. This is the same scheme used in many VCOs for the pitch CV.

As the 1k 3300ppm tempco resistor is the most used tempco resistor in DIY designs, possibly making it easier to get hold of, I have decided to use it for this design as well. This, however, means that the input CV has to be amplified before it is attenuated by the resistor voltage divider formed by R21 and TR1.

There are two ways to build this circuit - trimmable and not trimmable. For the not-trimmable version, use the exact values found in the diagram - 57k (47k + 10k) on the CVTRIM input and 23k (22k + 1k) in the feedback loop. All resistors should be 1% or better metal film resistors.

For a trimmable version you may replace the CVTRIM input resistors with a 10k resistor. The input voltage should be around -2.6V, and you may connect a 20k or 25k potmeter between 0 and -15V to achieve this. This potmeter will subtract a voltage from the input CV, which affects the maximum and minimum amplification possible. Both move in the same direction, if the maximum drops, the minimum will also get lower.

You may also change R11 to a 18k resistor and R19 to a 10k potentiometer. This pot will affect the "distance" between the maximum and minimum amplification. Turning the potmeter to the right and increasing the resistance of R19 will make the maximum amplification drop, but at the same time the minimum gets higher (i.e. you will not be able to attenuate the signal as much).

CV (linear) and response (Exponential) without trimming. No tempco used but temperature is 24 degrees celcius. Trim voltage is -15V, not 15V as it says in the picture.

The maths behind this

The maximum input to the transistor base when no correction CV is added should be -0.2072V. Using the R21/TR1 resistor voltage divider means that the output from the IC1A opamp must be -11.8V.

This voltage should be reached when the CV is 5V, which means that the CV has to be amplified -11.8V/5V = -2.362 times.

We will almost achieve this if we select a 23k feedback resistor and a 10k input resistor.

To reach -70dB, the correction CV must be 0.106V at the transistor input. This means it has to be 6.042V at the opamp output due to the resistor voltage divider. If we  choose to input -15V at the CVTRIM input, we need an input resistor (R1+R3) = 23k * -15V/6,042V = 57.1k

Or

If we choose a 10k CVTRIM resistor, the input correction CV must be 6.042V * -10k/23k = -2.63V

UPDATE: As pointed out by MB in the comments, there is an error in the schematics. The output from IC1A/B should go to the top of the 56k resistor and the Q1/Q2 transistor bases should be connected to where the 56k meets the 1k tempco, as shown in my original sketch here:

Exponential VCA, first try

After getting a very good result from the linear VCA, I felt like making an exponential version as well. The Lockbox VCA has an exponential converter, so why not try to use the same thing here?

I have written a long text about the exponential converter and learned a lot about it, so I had everything I needed in terms of theory ready. Reading this will give good insight to why the exponential converter is like it is and how it works, but it is not necessary to follow the reasoning in this post.

The most important result from the document is that the output current from the converter, Ic, can be written as:

   Ic = Is * e^(Vb/Vt)

where

• Is is a constant reference current
• Vb is the voltage at the base of a transistor in the converter, which is a fraction of the CV.
• Vt is a constant that - unfortunately - changes with temperature.

The exponential curve

My big question when I started looking at the exponential VCA was how exactly the exponential curve should be. Where should it start?

Now, this may sound strange, one would always like the control current to be 0A when the CV is 0V, and at its maximum when the CV is at max. Unfortunately, the formula above will NEVER be zero. When the CV is 0V, e^(Vb/Vt) equals 1, and so Ic = Is. We may add a negative voltage so that e^(Vb/Vt) becomes less than one, but it will still never be zero.

The question then is, how close to zero do we have to get before we cannot hear the signal passing through the VCA any more?

Offness

I tried some component values and looked at the result on the scope. It looked good, but when I tried connecting the output to an amplifier, I could easily hear the sound even when the CV was 0V. After some googling, and realising that I had what I needed in my bookshelf, I discovered a few nice rules.

Douglas Self writes about this in the chapter "Volume and Balance control" of his book "Small signal audio design".

He states that a good volume control should cover at least 50dB, and at least -70dB attenuation is needed to get a good "offness", where you cannot hear much of the signal any more. But what the heck does that mean?

I will not go into details about decibels, but here is a rule of thumb: If you increase a signal 10 times, you have raised the volume by 20dB. Increase it a 100 times and you have raised it by 40dB. Similarly, divide the signal 10 times and you have decreased it by -20dB. Divide it by 100 and you have decreased it by -40dB.

Volume controls (and VCAs) work by attenuating the input signal. The input should be at its highest and the volume control only "pinches off" parts of it, reducing the volume. So, as a consequence of the previous paragraph, when the volume is reduced to 1/10th of the initial volume, it has been reduced by -20dB.

To figure out how much -50dB and -70dB are, we can use the formula

   Change in dB = 20 * log(Output/Input)

Where log is the 10-logarithm.

Refactoring the formula gives us that

   Output = Input * 10^(Change in dB / 20)

At -50 dB:

   Output = Input * 10^(-50 / 20) = Input * 0,00316

At -70 dB:

   Output = Input * 10^(-70 / 20) = Input * 0,000316

In other words, the output is 0.00316 times the input when reduced by -50dB and 0.000316 times the input when reduced by -70dB. So, now we got something to aim for.

What attenuation to aim for

So, should you go for the -50dB or -70dB slope? Well, that depends of course. I have build both, and there is a significant difference in the offness. There is a faint but clearly audible sound from the -50dB VCA even when the CV is at 0. With the -70dB VCA I had to walk up to the speaker and put my ear next to it, and even then could only hear an extremely low sound (my wife actually heard it before me).

The problem with the -70dB contra the -50dB is that what you gain in offness, you lose in fine control of the higher-volume parts. You have to turn the volume pot of the -70dB one a bit before you reach the starting point of the -50dB one so you get less pot travel for controlling the rest. The difference is not extreme but it's necessary to be aware of it.

I would probably go for the -70dB, but if you do not need the output to be completely off (maybe the output is masked by the output of other sounds?) you may choose the -50dB version.

-50dB curve. Straight line is CV, 0 to 5V. Curve is response of a 5V input signal. Note: not finely adjusted so top misses 5V a bit.

-70dB curve. Straight line is CV, 0 to 5V. Curve is response of a 5V input signal. Notice how the bend is sharper and starts later than on the -5dB one.

Calculating vital parameters

We already know from the linear VCA that a maximum control current of 1.515mA will give unity gain within the circuit used, so we'll use that as a starting point. I will only show calculations for -70dB, but it is similar for -50dB and -100dB.

With a maximum current of 1.515mA, the minimum current must be

   0.000316 * 1.515mA = 0.479 uA for -70dB attenuation.

To see what this means for input CV voltage, we need to use the exponential formula again.

Remember,

   Ic = Is * e^(Vb/Vt)

which means that

   Vb = Vt * (ln(Ic) - ln(Is))

Vt varies with temperature, it is in fact

   Vt = ((degrees in celcius +273.16)*1.38*10^-23) / (1.6*10^-19)

This means that at 20 degrees celcius ("room temperature"), Vt = 25.3mV

Is is a constant that we choose ourselves. From my initial trials I found that an Is of 15V / 510kOhm = 29.4uA worked fairly well, so I chose that for my further calculations

Now we  can find Vb:

   At 0.479uA, Vb = 25.3 * 10^-3 * (ln(0.479*10-6) - ln(29.4*10^-6)) = -100.6mV

   At 1.515mA, Vb = 25.3 * 10^-3 * (ln(1.515*10-3) - ln(29.4*10^-6)) = 96.27mV

In other words, the voltage span needed to control the VCA from 0 to -70dB is

   96.27mV - (-100.6mV) = 196.8mV

Ah, but our CV spans 5V, and it starts from 0, not -100.6mV? Well, that can easily be corrected by an opamp summer with gain < 1.

By using a 1k feedback resistor and a 25k input resistor, our 5V CV is reduced to a 0.2V CV. To move the starting point to -100mV we only need to add a negative voltage. If we use the negative supply rail, -15V, we have to divide it by 150 to get to -100mV. As it has to run through the same 1k feedback resistor, we need to run it through a 150k resistor to get this attenutation.

A word of confusion

I have neglected to mention one thing. The exponential converter described above requires a positive Vb and it has its positive reference current Is and control current Ic running down into the collectors of the exponential converter transistors.

The output of the opamp summer connected to the CV circuit however, inverts the voltage giving us a negative CV. At the same time, the LM13700 requires a positive current running INTO pin1.

Fortunately, a PNP-transistor based exponential converter works exactly opposite of the NPN based one described in my texts about the exponential converter. Instead of a positive Vb it requires a negative one, and its reference current Is must run out of the collector. So by connecting the reference current generating resistor to -15V instead of 15V we are good to go.

Temperature is a bastard

Although we now have a working exponential VCA, it only responds accurately to the CV when the temperature is exactly 20 degrees celcius. This is because of the Vt mentioned earlier.

To see just how bad this gets, we can do some quick calculations:

Temperature	Vt	Ic max	Percentage of 20 degrees
0	23.6mV	2.02mA	133%
10	24.4mV	1.74mA	115%
20	25.4mV	1.52mA	100%
30	26.1mV	1.33mA	88%
40	27.0mV	1.18mA	78%
50	27.9mV	1.05mA	69%

This may pose a bit of a problem. It is likely that the temperature will be higher rather than lower than 20 degrees, so the current will probably never be too high. Still, it may be a good idea to add a trimpot to the voltage that moves the starting point so that you can adjust your VCA to work best at your desired temperature. This will also alter the maximum attenuation, but not dramatically.

An exponential converter that works better with temperature changes will be suggested later.


Suggested resistor values for -50dB, -70dB and -100dB:

attenuation	Gain, R15	CV, R4	Trim, R3	Trim if variable voltage/pot.
-50dB	1k	33k	292k	220k
-70dB	1k	25k	150k	120k
-100dB	1k	18k	84k2	75k

If a trim pot or variable voltage is connected to the CVTRIM input, use the resistor value in the last column. If not, connect CVTRIM to -15V.

Dual exponential VCA with -50dB attenuation, 0-5V CV

The Xonik VCA revisited

I have spent some time tweaking the LM13700 based linear VCA. I read somewhere that one should aim to use the highest possible value for the control current I_abc, which makes sense:

The way the VCA works is

• The input voltage is divided down to something the LM13700 can handle
• The voltage is attenuated/amplified by the LM13700, the highest output signal is found when the control current is at its highest.
• The output current from the LM13700 is converted into a voltage and amplified by the output opamp.

Now, the lower the maximum controlling current is, the higher the output amplification must be. The higher the amplification, the more noise you amplify as well.

With this in mind, I took a closer look at the circuit and the LM13700 datasheet.

The datasheet states that the maximum Iabc the IC can handle is 2mA. Any more than this and the device self destructs.This is why some circuits out there have a 10k resistor connected between the transistor collector and the IC. The Iabc input (pin 1) of the LM13700 will stay close to the negative power rail, and the collector can never move above 0V (more than this and it would turn off as the collector has to be more negative than the emitter, and the emitter is connected to the CV inputs opamp's virtual ground). This gives a maximum voltage across the resistor of 15V, and a current through the resistor and into the LM13700 of 15V/10k = 1.5mA.

This can safely be changed to a 8.2k resistor which will give a maximum current of 1.83mA. It could actually be reduced even further, from my experience the Iabc input never goes much below -14V, but at least this is still on the safe side.

After some experimentation I decided to use a 3k3 resistor on the CV input, R2, when using a 0 to 5V CV. The maximum input current will then be 5V/3.3k = 1.515mA (A higher CV input will make this go higher, but it will be limited by the 8.2k resistor on the LM13700 input so we will never break the LM13700).

With a CV of 5V, I then tried different values for the input and output signal resistors. Assuming we want the input signal to be allowed to swing as much as possible, I added a 15V signal through a potmeter to the input and watched when the LM13700 started clipping. I ended up using a 27k input resistor, R1. To get the output signal back to the same amplitude (The opamp will start clipping with an input signal of about 13V, so I tried this with a 10V signal) I had to use a combination of a 22k and a 1.2k resistor in the feedback of the output amplifier, R15 and R17.

I also tried various other configurations. One may want to use a 0 to 10V CV instead of the 0 to 5V (the CV has to be positive, so if you want to use a -5 to 5V CV you have to bias it into 0 to 10V first. A negative CV will not harm the VCA in any way, it will just be fully turned off when the CV is less than 0V). If you are certain your input signal will never be above 5V or 10V you can also get away with less attenuation/amplification and thus less noise.

Here is a table of good values to use:

CV	Signal swing	Input resistor R1	Output resistor R15 (+R17)	CV input resistor R2
0-5V	+/-5V	5.6k	4.7k	3.3k
	+/-10V	12k	10k + 560R	3.3k
	Max	27k	22k + 560R	3.3k
0-10V	+/-5V	5.6k	4.7k + 680R	6.8k
	+/-10V	12k	10k + 1k	6.8k
	Max	27k	22k + 1.2k	6.8k

You can of course use a potentiometer instead of the output resistors, this will let you tweak the output perfectly. My experience however is that this is unnecessary as long as you use 1% resistors and the values above.

Dual VCA circuit using 0-5V CV

Ps: The additional input resistors are changed from two 510R resistors to one combination of 470R + 22R and one 510R. While this adds an additional resistor, it is well worth it in terms of balance / centering. Alternatively use a 1k potentiometer instead, with one side connected to the positive input and the other to the negative input. The center pin should be connected to ground.

Input vs output signal. The topmost wave is the input. The output wave is slightly off center.

Input vs output signal with a different balance - the waves overlap nicely.

PPS: The output buffer inverts the phase of the signal, this can be remedied by adding a second opamp with unity gain.

Disclaimer: The Xonik VCA is mainly inspired by the LM13600 VCA found on the Bergfotron pages and the linear Lockbox VCA 2 found on the Lockbox Synth pages

Xonik VCO

I have spent a lot of time lately understanding the exponential converter and temperature correction built into it. I have even written a long document about it that I will publish as soon as I find a place to put the files.

Now, after understanding most of it, I have built a tiny VCO. The core is the well known Terry Michaels core from the electronotes articles, the same as the ASM-1, Ken Stone Catgirl synth and the PicoVCO (which in turn is presumably the same circuit as the Hearn Morley HMVCO1c). The waveshaping is a somewhat modified version of the picoVCO for the triangle part combined with the pulse circuit from the Yusynth VCO. All wave outputs are +/-5V.

The PCB layout is heavily inspired by the PicoVCO v2.0, but I have kept the linear FM input and moved the triangle symmetry potmeter off board so that all pots are on the motherboard. The symmerty circuit is also somewhat changed. Physically it is about 21 x 33 mm and it has 13 pins, so it's slightly larger than the PicoVCO.

The new super-mini-dual-VCA

I had a little time to tinker with Eagle again yesterday, and the result is a 40% smaller dual VCA. I've now put parts on both sides. I had to remove the separate CV input and trimmer outputs - instead of the trimmers you can put four 510 Ohm resistors on the board. The physical size (excluding pins) is now 27.3 * 16.8 mm.

I must admin that I like the original a little better as it had labeled pins and a nice big Xonik logo on it, but this one is very cool for being so small.

VCA works as expected

I tested the dual VCA today. Except for a mixup of output pin 1 and 2, it works as expected. At 10V the amplification is just slightly higher than 1. The VCA is linear, so at 5V it looks roughly like 50% amplification

VCA hooked up to the bread board. The two 1k trim pots are for adjusting the symmetry around 0V. 

Response at CV=10V. The outermost triangle wave is the output, and it's slightly larger than the input.

Response at CV=5V, the output is about half the size of the input.

7 saw controller board populated

I finished populating the 7 saw controller board tonight. It has not been tested yet but it looks good :-D

I didn't have any 22k 0204-size resistors, and I can't find any 3M3 or 4M7 resistors anywhere in that form factor, so I used normal 1% metal film ones instead. It looks a bit wonky but will work fine.

I have yet to test both the dual VCA and this board so not sure it works yet...

7 saw controller board

tiny LM13700 based dual VCA

Silk screen looks great :-)

Working on analog multiplexing

I am trying to do analog multiplexing. It's a bit harder than key/digital multiplexing, and I have made this initial design. Let's see what the nice people at electro-music think of it.

First multi-sub-oscillator board populated

The first 4 x sub oscillator board has been populated. This is the top board, a second pcb with four more sub oscillators will be mounted below it. This time I've made sure connections from the lower board actually go through the top board, so no cables have to be attached to the lower board (as opposed to the oscillator boards where you have to connect cables to both boards.

This is the rev 1 board which has the same bug as the single sub oscillator, so I have soldered 100nF caps to the flip flop feedback lines to make them flip and flop like it should.

Quad sub oscillator

Close up, note the tiny 100nF caps soldered directly onto the smd components' legs.

New panels

This week I've finished five new panels and prepared prints for three more. I've done two more oscillator panels, two sub oscillators and one 7-saw controller. I've also printed two more sub oscillators and one controller. Since the edges of the first oscillator panel didn't get as sharp as I wanted when folding the plastic around the edges, I've decided to cut the foil along the edges this time.

Plexi glass panels before attaching the adhesive film

Three oscillator modules - the first one to the left has plastic folded around the edges while the new ones are square cut for a sharper edge. I've also realized that I will run out of orange potmeter caps. Unfortunately, Re'an has stopped producing these so it's hard to get more. I've found some similar looking ones on Ebay, they seem to be 1mm wider but it may work out ok.

Finished sum oscillator and 7-saw control panels

Close-up of the sub oscillator. Note the input polarity switch, setting this to the opposite of the input wave will give that cool saw-within-saw wave shown in an earlier post

Tiny but crucial bug fix on the suboscillator board

A couple of weeks back I populated the first single suboscillator board, but much to my dismay it didn't work. I didn't have time to inspect it further, but I suspected it was related to the CD4013 flip flop as I had some trouble getting it to work originally. I ordered some more from a different source and they arrived yesterday.

I swapped the old chip for the new one, but it still didn't work. I then rebuilt the first part of the circuit on a breadboard, and it sort-of worked, but both flip flops in the package seemed to flip and flop at the same frequency, which they definitely shouldn't.

Today, I tried adding in a 2k2 resistor between the D and not-Q ports, as well as a 100nF cap from D to ground, a trick I found somewhere online when I originally designed the circuit. This made the circuit work again.

Doing this on the PCB would require me to cut some traces, which would be a lot of trouble, especially on the SMD-based quad suboscillator.

Luckily, when i replaced the 2k2 resistor with a wire, while leaving the cap in place, it still worked :) I soldered two caps onto the pcb as well, and it too worked like it should. I have not tried the board at various frequencies, but for the time being it seems to work fine.

While searching for solutions to my problem, I came across this post named "cd4013 weird shit" - that really sums up my thoughts about this whole problem too...

The fully populated sub oscillator board. It looks a bit like a bug of sorts, don't you think?

A closeup of the 100nF capacitor. Also note the beautiful curved cutouts, I just love the boards OSHPark are able to produce!

Final (?) Saw mix and detune circuit tested

This week I've spent at least 20 hours recalculating and building the saw mix and detune circuits on a solderless breadboard. I changed the mix and detune inputs to accept 0 to 10V instead of 0 to 5V, and also added an offset input to be able to use -5 to 5V inputs. The detune output still ranges from 0 to 5V, but the mixer slopes are now 0 to 10V as well, as this is what the VCAs I use expects.

I did a lot of up front calculations, but the final circuits were adjusted by ear. I am pretty pleased with the result, enough to go ahead and redraw the PCBs. I have decided to place the two VCAs on a separate, tiny PCB, which can be reused for other projects. I will also try to fit all parts on a single PCB instead of the two that the initial design uses, but this probably requires me to use SMD components again - the prototype uses almost 30 op amps and more than 50 resistors and takes up one and a half proto board :-)

The result is quite good, although not as good as I initially hoped and dreamed about. I am still impressed with the result though, especially at lower frequencies - it will make one hell of a bass sound. I may try to make a "computune"-style automatic tuner, to keep the oscillators better synced. I will also add a chorus, filter and perhaps distortion module, but for now I will focus on making the controller PCB.

The prototype fills two breadboards and includes all the necessary electronics for generating separate detune slopes for each oscillator, mixers to mix the single input pitch CV with the detune CV, mix CV slope generators, two VCAs and two audio mixers. To the right, behind the breadboard, is the 8 oscillator module.

Before protoboarding this circuit, i finished the 8 oscillator module, complete with separate fine tune pots and 8 CV in / 8 saw out connectors. It looks very cool!.

I shot a few videos for a friend of mine - the quality is crap but it illustrates how things work:

Oscillator mixing

Detune

Quad saw mixing

Before new years I tried hooking up a simple mixer to mix four saw waves. The mixer consisted of half of a TL072 and five 47k resisors.

While I DID get a mixed output, it sounded worse for every new oscillator I connected. Today I hooked up an oscilloscope to try to see if I could figure out what was going on, and in fact I quickly realized the problem.

When mixing two +/- 5V signals with unity gain, they will occasionally reach +/10V. Mixing three oscillators will result in +/-15V, and four results in +/-20V. The problem is of course that above 14.8V (or thereabouts), the op amp starts clipping, chopping of the tops. This does not sound very nice at all.

By reducing the gain resistor to 12k, the sum of the four oscillators will be within the original +/-5V. And indeed, the output sounds so much better! Now I'm closing in on the buzzing, swarming multi-saw sound I am looking for :-)

I have to rethink the oscillator mixer for the 7-saw though, to prevent clipping at all times. Next up is breadboarding the mixer again :-D

How to make cheap front panels

Due to our recent car purchase, I expect money to be a bit tighter than usual in the next months. I still need some cool front panels for my upcoming polyphonic/super saw synth. I have therefore been looking for ways to create decent panels at a low price instead of using Schaeffer/front panel express or similar services.

Today I completed the first prototype, a panel for the 8-saw oscillator, and this post explains the process in detail.

I have the luxury of co-owning a Silhouette Cameo cutter with my wife. This machine, while meant for scrapbooking and cutting vinyl for arts & crafts, works perfectly for cutting all sorts of soft materials. While you can get away with just an x-acto knife or similar, the machine makes the job so much easier and the end result much better.

Requirements

For the panels, you need the following:
- 2 mm thick perspex/plexiglass or similar hard plastic glass
- Adhesive-backed glossy photo paper
- Adhesive-backed transparent plastic

I got my perspex from Biltema here in Norway, but you can get similar stuff elsewhere. Make sure it is 2 mm, any more and you may not be able to attach the mini jack sockets, any less and the panel will bend too easily.

The photo paper I used is called Skyhorse Premium Quality Glossy Photo Paper, 130g, self-adhesive cast-coated. I got it off ebay in the UK. It is rather thick, and the ink I use (Canon CLI-55xx) is not absorbed well enough to protect it from wear and tear - it smudges easily even after drying for some time. This is however no problem as long as the paper is covered with the adhesive transparent plastic

The adhesive plastic I use is the kind used as a dust cover on books. It comes in glossy and matte finish, I've chosen the glossy one.

The process

Perspex

First I designed the panel in Illustrator. When doing this, I added cross hairs to all circular components, to tell me where to drill holes. When satisfied with the design, I made a copy of it and removed all but the outline and cross hairs. I then printed this design onto a piece of the adhesive photo paper, trimmed the edges using a paper cutter and attached it to the perspex. To make it easy to remove after drilling, I put the paper on top of the protective film that covered the perspex.

I then drilled all holes using a 1 mm drill bit in a drill press. It's much easier to get an accurately positioned hole using such a small bit first. Remember to also drill holes for any key tabs on the potentiometers.

Some of my holes are very close (1 mm) to the edge of the perspex. I drilled these first, in case they crack the perspex, so that as little work as possible is lost. Be very gentle and do not put a lot of pressure on the drill.

I then drilled 3 mm holes, even in those places where the end hole is supposed to be larger than that - still to get a better accuracy.

Then I used a conical drill bit to drill any holes larger than 10 mm. Finally, I drilled any remaining 6 and 7 mm holes.

After drilling all the holes I used a scroll saw to cut the edges and straightened them by sliding the panel along a metal file. This is the end result:

Cut and drilled perspex

Note that I tried to drill the holes for the potentiometer key tab only partially through the perspex. This was a bad decision as the plastic extrudes slightly, making the hole visible on the front even after attaching the photo paper. Had I drilled it all the way through, it would probably have been invisible.

Photo paper

After creating the raw perspex panel, I printed the panel design onto the adhesive photo paper.

The Silhouette cameo comes with a plugin for Illustrator that lets you add what is called registration marks to your drawing. Later, when you want to cut out parts of the design (holes for pots etc), the Cameo recognizes these registration marks and knows where the design is located on the paper. The plugin costs $40 but is well worth the money.

Panel design with registration marks in the corners

Next up is to cover the panel with the transparent plastic to increase its durability. I decided I wanted the edges to be well protected as well, so I came up with a way that let me put the plastic around the edges and onto the back. If you do not need this, just put plastic all over the panel and cut along the edges of the front panel (after creating holes for the pots etc).

In my case, I had to cut away parts of the photo paper around the edges of the panel so that the plastic film would not stick there. As the photo paper already has a non-stick backing paper, you only need to remove the photo paper itself. However, it is hard to cut through the top layer only, so I did instead cut all the way through:

Cut two grooves about 10 mm apart.

After cutting two grooves, I turned the paper over and put some Scotch magic tape over it to keep it in place. I then did the next side the same way etc:

Finally, I removed the photo paper from the parts I had cut out:

I could now cover the whole paper with plastic film:

 After covering the paper, I attached it to the cutting mat of the Silhouette Cameo:

Based on the same Illustrator design, I created outlines for the potmeter holes etc in the same position as the previously created cross hairs. I made sure to make the holes about 1 mm larger that the holes I drilled, as the silhouette is not accurate enough to get a perfect hit. The design is then sent to the machine and "printed" onto the panel. Due to the thickness of the plastic and paper combined, I had to set the cutting knife to 8, and even this proved to be a bit too little in some cases. Ideally you want to cut all the way through the photo paper but not through the backing paper.

I now carefully removed all the cut out parts using a sharp hook

Then I cut off the edges using a paper cutter. The corners are cut diagonally to be able to wrap them around the edges

Finally, I removed the panel from the backing paper and attached it to the perspex

After attaching the front panel components (knobs, jacks etc), all edges are covered and the result is quite pleasing. The plastic is slipping slightly from the photo paper close to the edges, this will probably not happen if you do not fold the plastic around to the back. Also, when tightening the DIN-jack sockets the plastic bulges slightly, this can possibly be prevented by cutting the plastic closer to the screw holes.

All in all I think the result is very good, especially considering the price and the time it took to create it.