Synthesizer Build part-25: DUAL BUFFERED MULTIPLE with LEDs.

A very practical module that replicates any voltage on the input and splits it 4 ways. Now also in Eurorack size. This is also known as a CV/Gate Expander but it's the same thing =) After seeing Sam Battle's version with the bi-coloured LED's I've added a second version of my own with the bi-coloured LEDs because I think it will be in demand. =) The version with LED's is half way down this article.

I have finished the case for the second stage of my DIY synthesizer so now I can continue building modules and writing about them here. So the first thing I wanted in the new case was a Buffered Multiple. I have built a number of filters that need a 1V/Octave signal but I only had one output to provide it so I designed this little circuit. You can connect anything you want to the inputs: Audio signals, LFO signals, 1V/Oct. CV, Gate signals, you name it. Any voltage presented at the input is replicated at each of the four outputs and if you need more you can connect one of the outputs of the first stage to input 2 and so get a total of 7 outputs for one input. 

I've added a Eurorack version with LED and, as a bonus, at the very bottom of this article there is a layout for a 3 x 4 Triple Buffered Multiple also in Eurorack format.

This module will work on both +/-15V and +/-12V.  Running it on 15V just means it can handle a bit higher voltages but as the average synth only uses control voltages with a maximum of +/-10V it really won't matter.

Here is the schematic drawing showing one stage. As you can see it's just 4 non-inverting buffers with all the inputs connected together. The module is just two of these on one piece of stripboard. I've put 100 Ω resistors on the outputs for some extra protection. Because there's (practically) no current flowing these resistors won't influence the output voltage. I've tried 1K resistors but they influence the voltage and bring the notes down by a few Cents. I did a lot of testing on this and 100 Ω is really the maximum. Of course you can use 1K and retune your VCO's but I want to keep it as clean as possible. If you don't feel comfortable with resistors on the outputs you can leave them out all together. In my own version I left them out too and replaced them with wirebridges. The IC's have built-in short-circuit protection anyway.

I've had some kind feedback in the comments of the Buffered Multiple not working right with Gate signals in combination with the Behringer TD-3. The Gate pulse would stay high. The solution was very simple, just put a 1MOhm resistor between the input and the Ground.  This is something I forgot but you must never leave an opamp input floating so there must be a resistor between the input and ground. I've adapted the schematic and put in the 1M resistor on the input. All the layouts are updated too.

To be quite clear because a lot of people miss read this: 

THE RESISTORS IN THE OUTPUTS ARE 100 Ω NOT 100K!!! If you put in 100K resistors you won't get much signal out at all. If in doubt put in wirebridges instead!!

Here's the verified stripboard layout. Just two IC's and some wire bridges and resistors. I recently added de-coupling capacitors for the chips because it was kindly remarked upon in the comments below that they were missing and it is good practise to include them.

Please don't forget to put a 1M resistor between between the input and ground (on both sides). The circuit needs these because opamp inputs should never be left floating. 

(Last revised 25-Feb.-2022: updated the layout and added 1M resistors to inputs.)

Stripboard only. The 100 Ω resistors can be replaced with wirebridges. In fact I would use wirebridges instead of resistors to begin with. I chose 100 Ω because that's the highest value that still doesn't cause a drop in voltage but you can easily do without them and use wirebridges instead. The resistors are there to offer some protection to the chip should you connect a live signal to one of the outputs by mistake, but most chips have internal protection from that anyway:

Bill of Materials:

BI-COLOURED LED VERSION:

Here's the layout for the version with the bi-coloured LEDs:

Wiring Diagram:

(Last revised: 11-feb-2022: added 1M resistor to inputs.)

Print only. 

BOM for the LED version:

Make sure you choose the bi-coloured LED's with two legs and not the ones with 3 legs. They won't work in this set-up. They must be two-legged. You may need to alter the value of the 18K resistors according to the brightness of the LED's you're using but I find 18K to be a nice middle value. Not too bright not too dim either. I've altered my own Buffered Multiple to include the LED function and These 18K resistors work just fine with the Red/Blue LEDs I'm using. But you can use 5K6 if 18K is too dim in your case.

If you take a look at the demo video in the article about the "Really Good AS3340 VCO" you can see this Dual Buffered Multiple at work to the left of the VCO I'm demonstrating.

Below note E1 the Red light, which is positive voltage, doesn't light up but above E1 you can see it light up. The higher the note, the brighter the LED gets and it can handle voltages as high as 15V without problems. Above the lowest 2 octaves the LED shines about as bright as it will ever get. For the blue part of the light the threshold for the LED is higher, about 3,5 Volt. So it will need a tiny bit higher initial voltage to turn on, but that's not really a problem. But for accuracy it's better to use Red/Green LED's because of the lower threshold voltage of the green LED vs the blue one.

I've added 6 times gain to the LED opamp stage so that the LED will start shining at the lowest of voltages. The 510K resistor determins the Gain. The formula for calculating the gain is 1+(510K/100K)=6.1
Of course you could also use two normal LEDs; one for the positive cycle and one for the negative cycle. It's up to you.

The extra gain of the LED output stage works very well in practise. The LED has quite a good range in brightness and thus gives a good indication of the strength of the voltage present on the outputs.

Here's the schematic for the LED version. I've connected it to output 4 of each of the two sides but you can choose any output that's convenient. Because the LED is buffered it takes no voltage away from the output. This schematic drawing shows only one side because the second side is identical to this one.

This module is very useful to have in your setup because sooner or later you're going to need at least one of these, just like you need at least one mixer/passive attenuator.

Here are a few pictures of the finished product. I made a little L-Bracket (also visible on the layout) so I could mount the print at 90° to the panel.

This is the version with the LED's installed. The right LED is displaying the voltage of the Gate signal which is +5V. The left LED is displaying the 1V/Oct signal which is a bit lower and thus the LED shines a bit dimmer.

The print up close. As you can see mine has more wire bridges because I forgot that some connections can be made directly under the IC. So I did it the hard way. The first picture below is the original module without LEDs. The second one is after I converted it and added the LEDs.

Because I added the LEDs later I had to use some jumpwires on the print to connect it all together.

EURORACK VERSION.

Single buffered multiple with one input and five outputs.

This is one I recently built for my eurorack case. It has 5 outputs because I added a dual opamp for the LED and that uses only one side so I had an opamp left over which I added to the outputs. I used 68 Ω resistors from the outputs to the sockets to give the opamps a little bit of protection should a signal carrying cable be plugged into an output by mistake (opamps don't like that). I didn't use IC sockets for this build because I've run out of those. I soldered the IC's straight in but that works fine and it also means bad IC-socket contacts won't be an issue so it makes it more reliable.

This layout is verified.

Here's the schematic. It's more or less a copy of the above schematic and I've put in the Bi-Coloured LED section with the 2nd opamp of the TL072. I lowered the value of the output resistors a bit from 100 Ohm to 68 Ω to be absolutely sure there's no voltage drop but you can use wirebridges instead of resistors too. The output resistors serve as a short circuit protection but opamps have that protection built-in nowadays. If you want de-coupling caps then put some 100nF ceramic caps over the power pins of the chips to ground as seen on the layout.

Here's the Bill of Materials for this version. The list has 100 Ω resistors instead of the 68 Ω in the schematic. Either of the two will work fine, even wirebridges work fine:

Here are some pictures of the finished eurorack module. It's 6hp wide (3cm) and 5cm deep. I used hot-glue to connect the stripboard straight to the back of the panel and next to the output sockets to which the stripboard is also glued for some extra support. Works very well and it's very sturdy. I soldered the power lead straight to the stripboard and also secured it with some hot-glue. 

 

As you can see in the last picture there's one ground wire going through all the ground lugs of the sockets and the cathode leg of the LED. From there a copper wire connects it straight to the ground of the circuitboard. Don't rely on the conductivity of the metal front panel for grounding. Always connect socket grounds with wire to the circuitboard.

TRIPLE BUFFERED MULTIPLE.

As a little bonus for you, here's a Eurorack friendly version of a Triple Buffered Multiple. So 3 inputs and 12 outputs in total (no LEDs). If only one of the three inputs is used the signal comes out of all 12 outputs. If you use more then one input the signal gets divided up over the outputs. (See text on layout).

This module will run fine on either dual 12 or 15V so also useful for Kosmo sized systems.

If you want LEDs on this module you can choose to connect a Bi-Coloured LED to outputs A4, B4 and C4 and mount the LEDs next to those particular sockets or leave those 3 sockets out. The LEDs will probably pull down the voltage a bit on those outputs (but not on the rest.). Use a 15K resistor as a current limiter in series with each of the LEDs. Add decoupling caps if you wish over the powerrails nearest to each of the chips. A 100nF from + to ground and a 100nF from ground to - for each chip so 6 caps in total. There's enough room for that above each chip. You can use normal ceramic type caps.

I left in 6 empty strips at the bottom so you can hot-glue the stripboard to the back of a Eurorack panel and use the side of the sockets for extra support, so the components will stick out above the sockets (or below the sockets, which ever way you look at it ^___^). If you have an other way of mounting it and don't need the extra empty space then just saw it off.  

Okay, that's an other one done. Number 25! A bit of a milestone for my synth build :)

If you have any questions or comments just leave them in the comments below please. 

See you on the next one!

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Synthesizer Build part-24: ADSR with 7555 (YuSynth design)

A great ADSR. Works perfectly and it's a very simple design, no trimmers to set. With verified stripboard layouts and now also in Eurorack format. There are also Eurorack Gerber files available for download for this project.

What does an ADSR or Envelope Generator do?

The Envelope Generator is generally better known as the ADSR which stands for Attack, Decay, Sustain and Release. These are the four amplitude phases a note goes through when you press a key on the keyboard. If we didn't have this ADSR in combination with the VCA, we would constantly hear the oscillator sound but we only want to hear it when we press a key on the keyboard right? So as soon as a key is pressed down, a Gate signal goes into this ADSR to tell it to produce an Envelope Signal. This envelope signal then goes to the VCA (Voltage Controlled Amplifier) where it opens up the VCA and so the amplitude of the envelope signal determins the volume of the sound coming out of the VCA.

The attack is the speed of the initial rise of the note, when you first press the key. Set it to zero and the sound is instantly there. Turn it open and the sound is going to take a while until it gets to full volume.

Decay is the time it takes for the note to go from the peak attack level to the sustain level. 

Sustain is the level of the note as you keep the key pressed down. It is usually set a bit less loud than the peak Attack level. (If we set Sustain fully open it will be on the same level as the peak Attack level and then it won't matter how you set the Decay because there's nothing to decay to.)

Then we have Release and that is the amount of time it takes for the note to fade out once you let go of the key. So the envelope generator produces a signal that determines the volume of the note over time and this signal is being used by the Voltage Controlled Amplifier (VCA) which interprets it as an output level. In some Minimoog synths it is also called the Loudness Contour.

Now of course the envelope output is a control voltage so it doesn't mean that you need to use it for the above mentioned purpose. You can connect it to anything that can be controlled with a control voltage like the filter cut-off or the pulse width of a squarewave or even the pitch of an oscillator. This opens up a miriad of options but let's not get ahead of ourselves here. If you're just starting out with synth building, you need the ADSR to open the VCA and the fancy stuff will come automatically with more experience. And this ADSR is perfect for beginners but also for seasoned builders in need of a good working ADSR. This is my ADSR of choise really.

My building experience:

This is the fourth Envelope Generator I present on my website and I think this one is the first that worked as it should straight away. No trimmers to set in the circuit either. I just used the schematic from the Yusynth website made a layout and built it. On the website he has two versions, an old and a new one. I built the new one. I can say without any doubt that this design is perfect if you want a good and reliable ADSR to pair with your VCA or to drive a filter. And because the circuit is so simple, even a stripboard version like this one would be robust enough to put in a rig you take on tour with you because, providing the panel is sturdy enough, there's practically nothing that can go wrong on the circuitboard.

This is the schematic. The opamp numbering on the schematic is different on the layout, I used the opamps in a different order but it works the same.

And this is the stripboard layout I made for it. It is verified, I used it for my build and it worked first time. Because it's so simple a design I didn't even test the print after building it. I made a frontpanel and wired everything up and then I plugged it into my synth and it just worked.

You can see the components are rather stretched out over the stripboard. This is something I did in all my early projects to make troubleshooting easier but I'm sure you could make a more compact layout if you need it. (There are also Gerber files for a Eurorack version available at the bottom of the article.)

Stripboard only. 

Bill of Materials:

If you want to add some extra outputs with buffers then below here is an extra layout that you can add to the ADSR to provide you with two extra normal outputs and two inverted ones. Of course you don't need to use the inverted output signal, you can use all four outputs for the normal signal. It doesn't matter what kind of signal is presented on the inputs, it will be replicated on the two outputs. (Two outputs for each input). This is an all purpose design so you can use this board for anything you like, even other projects like VCO's.

The wiring of the potmeters may look a bit strange with pin 3 left unused on three of the four potmeters, but I assure you that this is the way it should be wired up. Just follow the layout. It'll work fine I promise you. You can see in the schematic drawing that these pins are left dangling in the wind so that's what we do.

The ADSR triggers with a gate signal with a threshold of 3 Volt. The output envelope is 10Vpp. There's a manual trigger button on the panel (which is useful for testing). The envelope generator has two outputs. There's a normal output and an inverted output with a switch that lets you choose between +10V to 0V or 0V to -10V. There's also a switch to change the duration times with 'Fast' and 'Slow' settings. Use a DPDT ON-ON switch for the Fast/Slow function and a SPDT (ON-ON) switch for the Inverter voltage function. In Fast mode the duration for Attack, Decay and Release can be set between 1mS and 1Sec. In Slow mode they can be set from 5mS to 10Sec. These times are generated by the 1µF and 10µF electrolytic capacitors C4a and C4b. In the text on the Yusynth website it says to use Tantalum caps for this but I used normal Electrolithic Caps and this works just fine. I hate Tantalum caps anyway, they always blow up on me, LOL. If you want longer times you can install bigger caps. You could even take a 3 position switch and add a third cap of, for instance, 47µF to generate really long times. I haven't tried this myself so I can not guarantee it works but I don't see why it shouldn't.
There's a LED to indicate the level of the envelope. The LED remains lit very dimly if there's no Gate signal present and the ADSR is at rest. This is normal for this circuit. It simply indicates the ADSR is ready to fire.

Make sure you use three logarithmic 1 Mega Ohm potmeters for Attack, Decay and Release. Otherwise it will be difficult to set the parameters accurately. For Sustain we use a normal linear 10K potmeter.

It's interesting to note that all the 1 Mega Ohm potmeters control time parameters (Attack time, Decay time and Release time) while the 10K linear potmeter controls a level. The Sustain level.
You can run this envelope generator on a dual 12 Volt powersupply without any changes only the envelope output levels will go from 0 to 8 Volt instead of 0 to 10 Volt.

EURORACK LAYOUTS:

I recently made layouts for Eurorack in both the 10 pin and the 16 pin versions. In the 16 pin version the Gate input is connected to the eurorack-connector's gate pin but also has a separate input socket. If you want to disconnect the Gate signal from the eurorack-connector if you're using the normal input socket, then you must solder the gate connection from the eurorack-connector to the switch of the gate input socket instead of using the wirebridge as shown on the layout for the 16 pin version.

(Remember there are also eurorack Gerber files available at the bottom of this article.)

I've had confirmation that this layout works. So it is now officially verified.

Eurorack 10 pin version:

A very observant reader drew my attention to the fact I had forgotten the connection from pin 6 of the 7555 to pin 14 of the TL074 so I had to tuck that in later and that's why it runs underneath the chip socket of the TL074.  You can also choose to make that connection directly on the backside (copper side) by soldering a small wire inbetween those points, that's up to you.

Print:

Eurorack 16 pin version:

Print:

Here are some screen shots from the oscilloscope. These are from the 'Kosmo' sized ADSR but that shouldn't matter in the end result of course:

The normal envelope:

Inverted 0V to -10V:

Inverted +10V to 0V:

Here's a image showing the fastest rise time this ADSR can reach. It's just under 1 milliSecond or 980µSec.

Well, that's all there is to say about this project really. A very satisfying build because everything worked as it should right from the start. I would say that this is the perfect ADSR. The panel potmeters work over their complete throw, unlike some other E.G.'s I built, and you can set all the parameters very easily. If someone would ask me what ADSR to build I would certainly recommend this one. You can easily add on extra outputs if you so desire. You can add a TL074 for instance and wire up some extra outputs and/or inverted outputs. That's easy enough to do.
Okay, to close off, here are some pictures of the finished product. I made a copper bracket to keep the print in place behind the panel. That way I could use just one M3 bolt. I soldered all wires straight to the copper side.

Finally: there are now Gerber files available for this particular module (for Eurorack) which I uploaded to MediaFire from where you can download them for free. 

Just click here: --- DOWNLOAD GERBERS ---

EURORACK VERSION WITH LOOPING OPTION:

Now there's also a Eurorack version of this ADSR that has a looping function included. Paul Darlington, a member of the EB Facebook group, came up with this brilliant addition to the schematic. Here's a link to the files he posted on Github: --- CLICK HERE ---

You are not allowed to use these files for commercial purposes! They are published under Creative Commons 4.0 license. 

Here's a picture of the faceplate of Paul's ADSR module:

Okay, that's all for this article. If you have any questions or comments please leave them in the comments below or post them on the special Facebook Group for this website.

If you find these projects helpful and would like to support the website and its upkeep then you can buy me a Coffee. There's a button for that underneath the menu if you're on a PC or Mac. Or you can use this PayPal.Me link to donate directly. All donations go towards the website and future projects. Thank you!

Synthesizer Build part-23: DIGISOUND 80.6 LOWPASS FILTER.

A very cool AS3320 design that sounds amazing! With verified stripboard layout and new schematics.

After having taken out the Sequential Pro One lowpass filter to make room for the Korg filter, I needed a new use for the AS3320 chip that was inside it. I found the Digisound 80 point 6 lowpass filter module on this awesome website that has all the schematics for the entire Digisound 80 modular synthesizer.
You can configure the filter for any type you want (it's all in the original text) but we are going to build the lowpass filter because for subtractive synthesis the lowpass is the best sounding and most useful of all the filters in my opinion.
I first made a new schematic drawing because the original had those zigzag lines for resistors and I find the rectangular way of drawing resistors easier and you can put the value of the resistor inside the box. Makes it less complicated to look at imho.
Anyway, here's the new schematic drawing:

So after that was finished I made a stripboard layout. It is verified because I used this twice and both times the filter worked perfectly. Furthermore it has been used successfully by others in their builds. Make sure you work accurately though because I wouldn't consider this a beginners project. The layout includes a second audio output with 3 times the gain of the original output. This is of my own design and is not included in the schematic drawing. It is this output that is wired up to the output jack-socket in the layout below. The original output is marked on the layout too. More about this further down the article:

(Last revised: 19-March-2020. Added a second audio output with 3 times gain compared to the normal output. 25-5-2023: Removed colour codes from resistors, added colouring to wirebridges.)

Stripboard only. Beware that some stripboards are sold with 56 instead of 55 holes horizontally. The layout is 55 holes wide:

Cuts and wirebridges seen from the COMPONENT SIDE! As always, mark the cuts on the component side with a Sharpie and then stick a pin through the marked holes and mark them again on the copper side and then you can cut them with a sharp hand held 6 or 7mm drill bit. 

Bill of Materials:

The panel potmeters used are all 100K linear types but the value isn't that important. Since they are all connected to either a powersupply voltage or an audio signal you can use any value you like from 10K upwards. I myself used three 100K potmeters for the Coarse, Fine and Resonance and I used three 10K potmeters for the audio and CV level controls. This works just fine. 

You can choose to include the Frequency Fine control potmeter or leave it out to save more room on the panel. I personally never use it but it is there if you want to play the filter as an oscillator when it is in full self-oscillation mode with the Resonance turned up full. You'll need to tune the self oscillation pitch to the chromatic scale of notes so in that case a fine tune knob will be very useful. But I personally never tried this so I don't know how well this filter responds to that. If you have any experience with that then please put it in the comments below so I can share it in this article.

This is a 24dB/Octave, 4-pole LPF and it is self oscillating unlike the Prophet One filter I used this chip in earlier. That one refused to self oscillate. I used simple ceramic capacitors for the 220pF caps and this works fine. There's no need for fancy polystyrene caps ^___^.
The 1µF electrolytic cap C7 at the input may seem to have the wrong polarity. Usually a cap like that would have the positive pole connected to the point where the signal comes from and negative to where the signal needs to go. In this case it is mounted correctly because the input opamp is an inverting buffer with a negative gain reducing the amplitude of the 10Vpp input signal by a third to an amplitude the chip can handle. In the output buffer the signal is then inverted again to a positive signal with a gain of 3 to give us the original amplitude.

Calibrating the filter:
There are three trimmer potmeters on this print and you can set them as follows:
RV8, the 100K trimmer, is used to trim away the DC voltage on the audio output. Measure the DC output voltage with nothing connected to the input and turn RV8 until it reads zero.
RV7, the 20K trimmer is an interesting one. It's used to have the filter track 1V/Octave oscillators correctly but I simply tune it for best sound. If you have a squarewave on the input and you turn this trimmer you can clearly hear the over-tones, the harmonics, change in pitch. You should be able to hear the frequency beating effect of the note from the VCO against the tone of the resonance. Trim until there's no frequency beating but also listen to the tone while changing the cut-off frequency and trim until it sounds right to you. There's a full description of the proper way to calibrate this filter in the original text, which is in the link I mentioned earlier in this article.
The last trimmer is the one in series with the current limiting resistor for the AS3320. Simply measure the resistance and set it so the total resistance of the trimmer with the 1K resistor equals 1,5K. You could also just put in a 1K5 resistor but turning this trimmer does have a little influence on the sound but you'll have to try it to know what I mean. I just set it to 1K5 and left it at that. Turning this trimmer all the way to zero resistance won't damage the chip though, eventhough it needs a 1K5 current-limiting resistor, 1K won't hurt it. The one thing I learned building this filter is that the AS3320 is quite a robust chip. I made a few mistakes building it the first time and the chip has had voltages (through resistors) placed on the wrong pins, short circuits and all sorts of other mishaps but it survived all that without a scratch. Thank goodness because I only have one of them at the moment :) Luckily I was able to test if the chip still worked by placing it back inside the old Prophet One filter and seeing if that still worked. That was very useful.
Anyway, you can use this filter with a dual 12 Volt power supply, but in that case the current limiting resistor should be 1,2K in total. But it's really not that important. Simply connect it to +/-12V and it should work fine.

This filter sounds amazing! It has its own distinctive sound and I can not say it sounds like the Korg or the ARP or the Moog Ladder filter. It sounds like a Digisound 80 filter, although it comes veeeeeeery close to the ARP in sound. This one sounds a bit more well behaved, if you know what I mean. The sound of the Resonance is clearer than in the ARP which has a Rensonance that is sharper and rougher in sound. But that's the only  difference I could hear so it occupies a solid second place over the Korg-MS20 and the Moog Ladder filter in my personal top 5. The ARP filter is still number one because it's a real rebel and I love it. But hey, remember, this is all just my personal preference. You may judge it quite differently. Actually, I find myself using this filter more often than the ARP filter somehow.
The output from the Digisound 80 LPF is a bit more attenuated than the other filters I built and that's why I used the left-over opamp in IC-1 as an output buffer with a gain of 3. There's a 150K resistor from pin 6 to ground and double that value, 330K, as feedback resistor from pin 6 to pin 7. (You can use any value over 10K for both resistors, as long as the feedback resistor is twice the value of the resistor to ground.) This brings the volume up to the same level as the other filters I made. As I mentioned earlier, the amplitude is first divided by 3 and then multiplied by 3 again in the output opamp but, at least in my filter, I found the sound still lacking in volume compared to the other filters. That's why I wired up the left-over opamp as an amplifier with an extra gain of 3 to bring it up to normal. I'm not sure if it's just my filter or if this is normal, that's why I left the original output un-touched so you can use it if you think my solution is too loud. You could also install a potmeter of 500K instead of the feedback resistor so you can manually set the gain. (Put a 50K resistor in series with the potmeter so the feedback loop can't go to zero Ohm.)
The original audio output is marked on the layout so you can choose which one you want to use.
The first time I build this filter I had used a coupling capacitor of 4,7µF over the audio output because I measured a big DC offset voltage on the audio output, but then I read the original text and found out you can trim that away with trimmer RV8 so I took out the cap and trimmed the DC away and now it's all as it should be.
This filter has an input for 1V/Octave but unlike the ARP filter it's not necessary to use this. The filter will work fine without it but if you connect a 1V/Oct. source to it, the filter will track the octaves better. The sharp synthesizer sound we all love, will be more prominent if you use the 1V/Oct input. It actually makes the filter sound better.
Like I mentioned before, this filter has 2 potmeters for the Cut-Off Frequency but I advise to only use the 'Coarse' control. Fine is only for Polyphonic synths. I included it in my build so I could hear its effect and write about it here, but I normally don't use it. It stays in the middle position because turning it just changes the sound a tiny bit. It can be handy though to tune it into a certain harmonic frequency because this filter brings out the harmonics of a square wave really well, but all in all; leave it out.

Here's a picture of the finished panel built into the synth.:

As you can see in the picture I also included a bypass switch on my panel so I can put other filters in series with this one and if I only want to use one filter I can bypass this one and send the signal straight to the next one without having to change the patch cables. For instance, having this filter in series with the Korg MS-20 in High Pass mode sounds pretty amazing too! That way you have a Band-pass filter made up of two different filters. The bypass switch is only connected to the 'Audio-1' input though. If you want to see the wiring diagram for this switch, you can find it in the article about the Moog Ladder Filter, in which I also installed a switch like this.

Here's a little video with a demonstration of the sound of the Digisound 80.6 LPF (EDIT: at this stage in building my synth it still hadn't occurred to me that you need to connect an AD or ADSR to the CV input to get that typical synthesizer sound. #facepalm (We live and learn LOL):

Lastly I want to share with you the efforts of LookMumNoComputer Forum member Doolang who successfully built this filter using my layout. He made it so it fits the Eurorack standard by cutting the print in half and connecting the copper strips together with wire. This works like a charm and he did the same with the Steiner-Parker filter which also worked fine.

So that's the Digisound 80.6 lowpass filter done. I can really recommend building this. It has some very recognizable synthesizer sounds that you should really have available in your synth. Make sure you use good quality stripboard though. The first one I built had problems because strips of copper would become loose and break. So I rebuilt it with better quality stripboard. Make sure you use the filter with the 1V/Oct connected to get the best out of it. This will make resonance follow the notes you play (filter tracking). It'll also work with out 1V/Oct of course.

Okay, thanks for being here and if you have any comments or questions just put them in the comments below or in the special Facebook Group for this website.

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Synthesizer Build part-22: RING MODULATOR (Yusynth design).

An excellent ring-modulator to serve as an extra source of weird sounds. This one is simple to build and works very well.

Next to the filter and the FM and Sync capabilities of a VCO, I think the ringmodulator is one of the most important ingredients for sound shaping in any synthesizer and an often overlooked one. No synth should be without one. The ring-modulator was something I always wanted to include in my DIY synthesizer and I was thinking of doing it the old fashioned way with audio transformers but they are very expensive. So I went looking for designs that used semiconductors and came across the Yusynth design. I ordered some MC1496N IC's from a shop near where I live because I didn't want to wait for components from China, what with the Corona virus going on etc. and I wanted to be sure I got real MC1496's and not fakes, which is so often the case with IC's from China. The MC1496 is a chip you will find in many vintage synthesizers and also in the Macbeth studio systems Dual Oscillators for Eurorack (a favourite of Colin Benders). They have a built in ringmodulator that uses this chip. Anyway, I got the chips the next day and I set out to make a stripboard layout.

A ringmodulator takes two signals and multiplies them. If you input just one signal you won't get anything out because if you multiply by nothing (zero) the outcome is nothing. So both inputs need a signal going in for this to work. (That's why you can use Ring-modulators as crude VCA's when switched to DC mode) One is called the Carrier wave and the other is called the Modulator wave. You can modulate the amplitude of the carrier and so get AM modulation. You can get the most beautiful results from this circuit. It can produce very cool bell like sounds or very gnarly AM modulated sounds. You can spend hours experimenting with it. I myself usually combine the output from the ringmodulator with the output from one of the VCO's going straight into a mixer so I can add both signals up and then send that through a filter and/or the lowpass gate and boy do you get phat sounds like that!

Click here to go to the original Yusynth article.  

Schematic for single ringmodulator:

Here is the (verified) stripboard layout I made from it. The original schematic is for a double ring modulator but I only built one. But it's just the same circuit repeated. I used this layout for my build so it is tried and tested. Note that the electrolytic capacitor on the AC output jacks has it's polarity reversed from those on the AC inputs.

Stripboard only:

Bill of Materials:

There different types of 1496 chips for sale, like MC or LM, but the letters in front of the numbers don't matter. They just indicate which firm produced the chip. They will all work fine in this circuit as long as they're 1496's.

TUNING THE RINGMODULATOR:

You can find the tuning procedure on the Yusynth website. I'll describe the procedure here:

There are two trimmers to adjust: A1 and A2. These trimmers are used for cancelling the input signals. The settings are quite simple, you will need a signal generator that delivers a sinewave signal with a 10V peak to peak (+/-5V) output amplitude. You can also use a VCO in your synthesizer to do this. 

The Yusynth website talks about audio amplifiers but it's easier to use a VCO with sinewave and a VCA from your synthesizer to do this and make sure you connect an oscilloscope to the output signal with a frequency counter enabled so you can set your VCO to 1kHz.

Here is how to proceed :

1. Set your audio generator (or VCO) to 1kHz and set the output level of the generator to 10Vpp.
2. Connect the output AxB of the module to an audio amplifier or your VCA. Be sure to set the input potentiometer of the amplifier to a low value: the output level of the module is 10Vpp and most of the audio amplifiers expect an input level that doesn't exceed 1Vpp. If you use a VCA it'll be okay.
3. Connect the signal generator to the input A with a capacitor (AC-INPUT). There, you must hear the 1kHz signal at the output AxB.
4. Adjust the trimmer A2 so that the 1kHz signal can no longer be heard at the output.
   
5. Disconnect the signal generator or VCO from input A and then connect the signal generator to the input B  with a capacitor (AC-INPUT). There, you must hear the 1kHz signal at the output AxB. Adjust the trimmer A1 so that the 1kHz signal can no longer be heard at the output.

Now the module is ready to operate. A last check can be done by connecting the 1kHz signal to both inputs A and B (AC inputs): there you must hear a signal which is twice as high in frequency (2kHz).

It will be a bit of a balancing act to get these signals on the zero volt line on your oscilloscope. Turning one potmeter might create an slight offset voltage that you can trim away with the other trimpot. If all is well you should easily be able to tune the ringmodulator in a few minutes.

If only one input is used there should be no output (because something times zero = zero) and if there's a signal on both inputs you'll get the two signals multiplied out of the output. That's why a ringmodulator can also be used as a crude VCA.

When I first tested my ringmodulator I kept measuring a negative offset of -9V on the output if I stuck a signal into input B. It turned out that my TL072 was faulty. I stuck a new one in and all was well.

PICTURES:
Ring modulation is a very interesting way of combining two frequencies and can get very complex very quickly if you use waves that have a lot of harmonic content like squarewaves. More on the theory involved is here on this Wikipedia page about ring modulation.

I didn't think I could fit any more modules in, but this one was just small enough to go into the wood panel above the modules, next to my Sample and Hold module. Here's a look at the finished panel and how I fit the stripboard behind it with a little copper L-Bracket I made myself and soldered onto the stripboard.

Ignore the sawdust specks in the bottom picture, LOL. I took it just after putting the module in and the synth was covered in fine dust.
Okay that's another one done. This is getting to be quite a big and powerful "sound design machine." It's about time I made a new case that I can put on top of this synthesizer so I can keep building ^___^

If you have any questions please leave them in the comments below!

If you find these projects helpful and would like to support the website and its upkeep then you can buy me a Coffee. There's a button for that underneath the menu if you're on a PC or Mac. Or you can use this PayPal.Me link to donate directly. All donations go towards the website and projects. Thank you!