Sunday, 29 December 2019

Synthesizer Build part-14: AD/AR Envelope Generator.

An updated/slightly improved version of the LookMumNoComputer simple AD/AR.  Improvements suggested by Sam Battle himself.

This Envelope Generator is a fantastic little extra to put in your synthesizer. It's always handy to have a few extra envelope generators in your synth to trigger filter responses or other parameters. I built a 'proper' ADSR a few pages back and this simple version is just perfect to have as extra. I found this on the LookMumNoComputer website and Sam has also done a video about this on YouTube which you can watch here:

Because the LFO from the last blog post didn't have a synchronization input I needed something that could trigger a filter response when I pressed a key on the keyboard so I decided to build this. I had just enough room left on the panel for the LFO to include this and it only needs a small bit of stripboard to build it up on.

Here is the layout that I made, which is just a copy of the one on the LMNC website but with a few changes (see text below. All potmeters viewed from the front.):

AD stands for Attack and Decay, this is when the switch S2 is in the Trigger position. That means there is no sound after you let go of the key. AR stands for Attack and Release and this is when S2 is in the Gate position and now the tone will fade out after you let go of the key.
It's fun to build it like Sam does in the video with a big arcade button with an internal LED light.
It's pretty straight forward build. In fact, it's so simple that I didn't even test it before building it in and luckily it worked straight away. It didn't work perfectly though. It needed pretty high voltage Gate signals and Triggering didn't work at all. There was a discussion about this on the LMNC Circuit Discussion Group Page on FaceBook and it turned out that Sam had advised to lower 3 of the 100K resistors to 10K and an other suggestion was to remove the diode from the input to the switch and make it a normal wire connection. I implemented these changes in the stripboard layout but I left the diode in place. I also advise to put in a bigger capacitor than mentioned on the original layout.  I was lucky enough to have kept some bi-polar capacitors that I took out of some circuitboards years ago because they came in very handy in this build. On top of the 1µF cap I put an extra 2,2µF bi-polar cap to get 3,2µF in total. (put in 5µF if you can) That gives a bit more time for the release to fade out. With just the 1µF it fades in just a few seconds.
Because I mounted this on the same panel as the LFO I was able to just connect the power leads to those of the LFO stripboard because they both use + and - 12 volt. So no need for an extra power cord and connector. This circuit can also be powered by +/- 15V.

Now, if you want something that is just as small but works a lot better then I can refer you to my 'Synthesizer Extra's No:01 SIMPLE AD/AR using the 7555'

This article is about the Thomas Henry designed AD/AR from 2014. He used the 7555 and his design works very well.

Okay, that's all for this one. If you have any questions about this or other builds on this website then please put them in the comments and I'll answer them asap. And while you're here, leave me a comment anyway!
Until the next one!

Synthesizer Build part-13: THE LFO (MusicFromOuterSpace version).

A very useful, good working and simple to build LFO for square-, sine- and triangle-waves plus a stepless transition between ramp- triangle- and sawtooth waves. A good LFO for beginners to build too. I still use this as my main LFO.

This is the LFO from MusicFromOuterSpace. It doesn't have a sync option but nevertheless it's a very useful LFO and it is in fact now the main LFO in my synthesizer.
This LFO has the following features: Stepless transition between Sawtooth to Triangle to Rampwave with one potentiometer. Sinewave and squarewave with changeable pulsewidth. Frequency control and a switch to go from High to Low frequency setting. 
Frequency Range with switch in 'HI' position = 1 wave every 2,39 seconds to 84 waves per second (239mHz to 84Hz)
Frequency Range with switch in 'LO' position = 1 wave every 7 minutes and 46 seconds to 1,43 waves per second (1,43Hz). The readings you will get will differ a bit from mine due to tolerance fluctuations in capacitor and resistor values.  
Squarewave pulsewidth (or dutycycle) goes from 1% to 99%. The pulse width of the squarewave is set with the same potmeter that controls the shape of the other waves. It also influences the shape of the sinewave.
A very feature rich design and a design with very few components so not much can go wrong. It uses a TL084 quad opamp chip and a LM13700 OTA chip.
I even managed to add a little extra of my own design; all the outputs go from -5 to +5 volt but I added two extra outputs for the saw-triangle-ramp wave and the sinewave that goes from 0 to 10 volt. There was room on the circuitboard to put a little TL082 on and make the two inverting buffers with DC offset potmeters. I'm sorry there's no schematic for these additions, I did it from memory, but this feature is included in the stripboard layout. Remember these 0 to 10V signals are inverted, so the waveshape potmeter works the other way around for these waves.
This LFO is meant to be used with a -12V/0V/+12V powersupply but it works equally well on a -15V/0/+15V powersupply without any changes needed.

Here's the layout, wiring diagram (All potmeters viewed from the front). The layout is verified. I recently built a second one of these LFO's to use as a standalone signal generator and it all worked first time. The green wirebridges indicate connections to ground:

(Last revised: 21-Jan.2021 Updated the old layout with some components re-arranged and got rid of a jump wire. Also a mistake corrected. 28-Aug.-2021: Cosmetic changes, got rid of resistor colour coding lines.) 

Print only:

Here's the schematic for the Music From Outer Space LFO:

Bill of Materials:

It was easy enough to put together although I did manage to forget 2 components, but finally I had it done. Then it was time to test it. Wouldn't you know it, I couldn't get it to work. So I applied the first rule of troubleshooting: Thou shalt measure voltages. Sure enough, my dual voltage supply was broken. I connected it to the power supply of the synthesizer and it suddenly sprung to life!
I added the AD/AR because I had room left on the panel but more importantly to compensate for the fact that this LFO has no sync input. I can now use the AD/AR to trigger a filter when ever I press a key on the keyboard. I even made it with a big arcade push button with internal lighting just like the one from LMNC, because I thought that looked pretty cool and you can press the button to get a loud filter resonance reaction (for instance). If you want to know how to build that I can recommend checking out the LookMumNoComputer website. Click here and you'll be taken right to the AD/AR page. The LMNC AD/AR does have some inherent design problems which means it's not very good with trigger signals. That's why I recommend to build the Thomas Henry designed AD/AR, using the 7555, chip instead. Click here to go directly to that page.

Here's a high resolution picture showing oscilloscope screenshots of the different waves.

Here are some pictures of the stripboard with wirebridges and with components:

Here's a picture of the panel I made for it. Like I mentioned earlier, it is combined with an AD/AR, the version that uses the 7555 chip. I used multi-coloured LEDs on the outputs to indicate positive and negative cycles of the outputs. There's no practical reason why I did that, I just thought it looked cool :)

Please, share and follow this blog and see you on the next one. :)
Oh, and if you have any questions and/or comments please post them below in the comment section or visit the EddyBergman Facebook group.. 

Sunday, 15 December 2019

Synthesizer Build part-12: THE KORG MS20 FILTER.

A good working version of the famous Korg MS-20 filter by Rene Schmitz, with updated stripboard layout and wiring diagram with LP & HP .

This is a filter I absolutely had to include in my DIY synthesizer project, for one because Sam Battle from LookMumNoComputer raves about it and it sounds amazing in his videos and the other reason is that it has the option to go between Low-Pass and High-Pass and I didn't have a High-Pass option in the synthesizer yet. I also added the Band-Pass option in the layout drawing however I tested it and it adds no extra benefit to this filter. Of course, combining this filter in High-Pass mode with one of the other lowpass filters gives you the bandpass option too and with a much better sound! (Check out the video at the bottom of this article to hear this filter in series with the Moog Ladder Filter. A really cool combination.) And if you build the Dual Korg Filter you'll have an even better bandpass option but that's for an other article.
There was a discussion on the Synth DIY Facebook group a while back about the possibility of switching this filter between 6dB per Octave and 12dB per Octave. I've tried it and it works but only in LowPass mode. More about this further down the article.
Btw, I tested this filter on dual 12 Volt and it works just as well as on 15 Volt so no need to change anything if you are feeding it from a +/- 12V powersupply. Of course you need to open Resonance a little more than on 15V but it's all within the normal throw of the potmeters so no problem there.

For this filter I used the 'Late MS20 Filter' schematic from Rene Schmitz which you can find by clicking here.
As he mentions in the text with his schematic, the gain of the opamps is hightened so you can get some weird sounds out of this. That's certainly true. :)
This filter is definitely different from other filters. It doesn't sound like the Moog Filter but it does have that 'ripping the fabric of the universe' synth sound and it's a real Speaker Ripper!. It's a 'Sallen-Key' type filter and it produces really divers sounds. (I always think of this filter as the 'Heavy Metal Guitar' of the synthesizer world.) You can get deep bass tones out of it and if you connect an Envelope Generator like the little AD/AR to the CV1 input you'll get a squarewave changing into a really bassy sinewave as the note progresses. It's awesome to experiment with this filter.
In HighPass mode the Cut-Off Frequency potmeter doesn't work over its complete throw. I found it usually only works over the first 50%. This is also true of the other MS-20 filters I built for the Dual version so this is normal behaviour for this design.
This filter is self oscillating in both the Low and High-Pass configuration. The more Resonance you give it, the more the two yellow LED's light up. I found that by using an LM13600 instead of a LM13700 you can  tame the filter a little. The 13600 seems to be a bit less aggressive although the difference is really small. Btw, instead of the TL074 you can also use the TL084 or the LM324. They are pin for pin compatible and work just fine. I personally tested them all successfully.

About Resonance and Self-Oscillation:
On the oscilloscope you won't see much Resonance ringing on top of a squarewave in Lowpass mode. I've noticed this with all 3 Korg filters I built so far. Maybe one little sinewave bump on the top left corner of a squarewave but it's not like the ARP or Steiner filters where the whole top and bottom of the squarewave is full of Resonance or self-oscillation. (It gets better with lower input levels though.) But in HiPass mode you get much more. However, this is normal for this filter and it still sounds pretty amazing so this doesn't matter.
Beware this filter does not like high input levels! When I feed it 0-10Vpp squarewaves directly from the Digisound-80 VCO it doesn't work right. It is much happier with the -5/+5Vpp levels of the Thomas Henry 555-VCO. It is always a good idea to put an input level potmeter on the audio input of this filter.

6dB vs 12dB.
Although this is a 12dB per Octave filter it is possible to get an output with 6dB per Octave roll-off if you tap the signal from pin 7 of the TL074 (A1 in the schematic drawing) with a 470nF capacitor, just like the normal output. The 6dB won't work in HPF mode because the signal is output from the stage before the one where we input the Highpass audio signal.
I have experimented with 6dB and at first I dismissed the option but having recently rebuilt my filter using (vintage) Polystyrene capacitors (and Polyester box caps for the 470nF output caps) it now sounds much better and the 6dB function sounds better too. So I made new layouts which includes the 6dB/Octave function with a switch to give you the choise between 6dB or 12dB. In the rebuilt filter I also used two BC558 transistors that were matched to within 4 points of their Hfe using a simple multimeter transistor tester. I'm not sure if this is necessary but it can't hurt. See the video below for a demo of the 6dB sound.

LM13700 pinout:
I took the part of the Rene Schmitz schematic that shows the CA3080's and I added the pinout numbers for the LM13700 chip which is just 2 CA3080's in one chip with added buffers which are not used in this filter. I'm not showing the whole schematic on my website because I don't want to draw visitors away from his website. It's Rene's schematic, not mine. Here are the pinout numbers:

Below here's the stripboard layout. I made my own version from the ones that are circulating on the internet to which I added the potmeter connections and the audio in and out, CV in, plus the switch connections for Low-, Band- and Highpass with the altered position of the 1nF cap at pin 12 of the LM13700. So with this and the schematics you should have all the info you need to build it right the first time. BTW, the 2 transistors used on the layout below are BC558 PNP's but you can also use the 2N3906 but those have to be put in the other way around.
Please note: I did add the Band-Pass option to the layout but if I were you I would just leave it out. But do some tests and decide for yourself.

(Last revised 1-March-2020 Corrected mistake with 470K resistor. 07-07-2020 added 6dB/Oct option.)

About the DPDT switch wiring for HP/LP mode: 
Connect the top two pins and the right middle pin together and connect that to the High-Pass input. You could even just forget about the top right pin and bypass it but I thought it was neater to include it. The lower right pin goes to ground. The audio signal goes into the middle left pin, and the lower left pin goes to the Low-Pass input. The Band-Pass switch is simply connected to the Low-pass input and the audio input. If the filter is in Low-Pass mode the BP switch won't have any effect but in High-Pass mode the switch will connect the audio to the Low-Pass input aswell so both inputs get the audio signal thus creating the bandpass characteristic.
A little sidenote: the HighPass and LowPass marking next to the switch on the wiring diagram reflect the function of the actual wires connected to the switch. A Toggle switch works the other way around. If you flip the switch upwards, the middle and bottom pins are connected to eachother. If you flip it downwards the top and middle pins are connected so on the eventual panel, the switch should have LowPass marked at the top and HighPass at the bottom.  This way of marking the switch functions goes for all the layouts on this website that have switches in them.

Close up of just the strip-board lay-out (Print this one and use it for your build. The lay-out is guaranteed, tested and verified faultless). The green wire-bridges are coloured green because they indicate a connection to ground. Don't forget to cut all the copper strips under the IC's.

(Last revised 1-March-2020 Corrected mistake with 470K resistor. 07-07-2020 Added 6dB/Oct option.)

Bill of Materials:

About the components:
I used fairly cheap LM13700 chips I got out of China and they worked just fine, however I'm convinced using the real thing will make it sound better. Get your chips from a reputable source.
I used a DPDT toggle switch (Double Pole Double Throw) to switch between High-Pass and Low-Pass but you could also use a jack socket for High-Pass input with a build-in switch that connects C4 to ground when nothing is plugged in there.
The Cut-Off Frequency potmeter in this build is a 100K one, but you can use any type you wish because pins 1 and 3 are connected to the + and - of the power supply so it is nothing more then a voltage devider. I saw that Sam Battle uses a 4K7 for this in his layout so use what ever value you want. (Beware that the voltage difference over that potmeter is 30 Volts so don't go too low with the value or you'll fry your potmeter. Remember Ohm's law!)
I don't think the Resonance potmeter is that critical either but you better stick to the schematics for that one. Keep it a 100K potmeter. I used a logarithmic one but linear will work fine too. As mentioned earlier, you can use an LM13600 instead of the LM13700 and instead of the TL074 you can use the TL084 or the LM324. They all work just fine.
As I mentioned earlier, I rebuilt the filter recently and used polystyrene caps instead of ceramics. It did make a difference in my filter but the first version I built of this filter was a mess to look at. Building something neat and tidy always makes it work better I found out. So don't expect miracles by changing the caps from ceramic to polystyrene.
One thing that might be worth experimenting with is the 10K feedback resistor over opamp A3 in the schematic. The one by the LEDs. You could put in a 15K trimpot and see what it does if you change the feedback resistance. I haven't tried this yet myself but I'm going to and I'll report here if I get any results.

Here's a picture of the panel I made for it. Because I recently re-built the filter I had one unused mounting hole left in the panel so I put a white 3mm LED in there. It's a bit bright =)

Here's a picture of the insides. Those big brown ceramic caps at the top and bottom are just for de-coupling. They're spread out a bit because they're so big. I advise to use polystyrene caps for all the other capacitors although that's not necessary for the filter to work. I just found polystyrene caps to sound better but you mustn't attach too much importance to the use of polystyrene caps. Any type will do.:

The video below shows a demo of the filter after I had re-built it in July 2020 and added the 6dB per Octave option. The audio in this video is a bit crap and I'm just ploinking along the keyboard to test the sounds from low to high notes. (If I could play properly I would have done that, but there we are... =) 

Here's an other and older video demo-ing some  extra effects. In Low-Pass mode the Resonance Control has very little influence on the sound. In High-Pass mode it has much more effect but in this filter the Cut-Off frequency is what it's all about. That's what you use to get the cool sounds:

This is a test with the Korg filter (in Highpass mode) and the Moog Ladder filter (in Lowpass) in series and using my 8 step sequencer and reverb from the CaraOK effects module:

Sounds amazing doesn't it? Especially with added echo or phase-shift effects. 

WARNING: Beware your speakers!! This filter can oscillate at below audible frequencies and the cones of your bass speakers will take a hell of a beating if you've got some serious amplification going. If ever a filter could be called a 'Speaker Ripper' this one is it. Quite literally. (I added this warning because tonight I almost blew my speakers up with this filter.)

I want to direct your attention to a very useful page from Scott Stites' website. He talks about all the different aspects of this filter, using two of these filters in tandem and his approach to adding a Band-Pass mode to it. If you want to build this filter, you have to read this text I think.: Click here for Scott Stites website.

Lastly, here's an other very interesting document I found by Sound Semiconductor entitled "Designing Voltage Controlled Filters for Synthesizers with the SSI-2164."
It goes into great detail into how filters work and how to design them and places specific emphasis on the Korg MS-20 filter.

Okay, that's it for this one. I hope you enjoyed this and other articles and if you did why not leave a comment. I'd love to hear your thoughts or questions in the comments below. Any questions will be answered by me asap!
If you like this website and would like to sponsor future projects, you can 'Buy me a Coffee'.  There's a link for that underneath the main menu if you're on a PC or Mac. All donations will go towards components for future projects. Thank you!
Right, see you on the next one!

Wednesday, 11 December 2019

Synthesizer Build part-11: ECHO and SOUND FX UNIT and LINE OUT.

100 different sound effects combined with a line-out and head-phones connection for the DIY synthesizer.

Here's a little item I found on eBay and thought it would work great in the synthesizer; and luckily I was right. It is a fantastic asset and makes the synth sound really professional and full. 
A word of warning though, this is not a beginners project. You need to know your electronics to follow my plans. This unit also needs two voltages to run on. The effects unit runs on +5V and the stripboard runs on +/-15V or +/-12V. So you must make sure you have those voltages available in your modular set-up. I recently added a Dry/Wet control so you can dial in as much of the effect as you want in your sound. This makes this unit really useable and perfect for modular synths. (see further down the article)

The unit goes under different names but mostly as the Cara OK ("Karaoke" get it?) or DSP 5V Red Digital Stereo Mixer Reverberation Karaoke Reverberation Board Module (if you copy and paste that into the search bar on eBay, you'll find it). The prices vary from $15 to about $30 us.
This is the cheapest listing I could find on eBay: click here.
It offers 100 presets with reverb, echo and even chorus, phaser, flanger, phase shift and reversal effects and combinations of them together. There's a rotary encoder with which you choose the preset of your choise and then you just press to confirm and engage the effect. It's a favourite with many synth builders I noticed.
I made a special panel for it and combined a 'Line-Out' control and bypass switch option with the module. Here is the schematic or wiring diagram to make this unit part of the synthesizer:

I had to put extra attenuation on the bypassed (normal) signal because it was much louder than the output from the Echo Module, but that was easily fixed as you can see in the diagram above. The panel that I built into my synthesizer also includes a headphones out stereo jack which is not included in this article but it's just an output jack soldered straight to the negative poles of the Left and Right output Electrolytic Capacitors. Weirdly enough, if you listen through the headphones, the echo module is much louder than the normal line out. This is probably some impedance matching issue but it doesn't bother me. It's easy enough to turn a volume knob so I'm not bothered.
The Stripboard can work on +/-12V also. It doesn't matter if you use 15 or 12V. The CaraOK effects unit must have it's own +5V powersupply! You can use a 7805 voltage regulator and connect it to the 15 or 12V of the stripboard to get 5V for the effects unit. This is not further described in this article. I'm assuming you have the knowledge to make powersupplies. If not, check out my article on this subject here.

Here's the stripboard layout (new version):
(All potmeters viewed from the front)

Please note: The GROUND connection of the Effects Unit must be connected to the GND of the stripboard. Everything must share the same ground. I myself made a central grounding point with a solder eye connected to one of the M-3 bolts holding the Effects Unit to the panel. To put it in a simple way: the ground of the 5V powersupply for the Effects Unit must be connected to the ground of the +/-15 or 12V powersupply of the stripboard. The Grounds of the Line Out cables are also connected to this common ground. You can tap that off from the 4th strip above the Left Line Out connection. (2nd strip below the chip).

Print only:

(Last revised: 14-Aug-2020: Corrected mistake with negative voltage supply to the TL074.)

I used a 3 pole double throw toggle switch (ON-ON type) to be able to switch the synthesizer between normal output and output through the effects unit. This is a bit of over-kill because you can just as well connect the inputs together (part S1-A of the switch) and then use a 2 pole switch to switch between the outputs. Better still. I describe down below how you can put in a dual gang potmeter instead of a switch and so have a DRY / WET control. 
I'm going to adapt this article soon and make a new layout to put in this Dry/Wet control permanently. 

I wanted a Dry/Wet control on this effects unit for a while now and I wanted to install it without having to rebuild the whole module.
I changed the 3 pole toggle switch for a dual gang potmeter and I connected the inputs, normally connected to part S1A of the switch, all together. If you do this make sure you keep to the right order with the wires. Best to make a few pictures of the switch connections first before you solder in the dual potmeter, that is, if you built this module already.
Below here is a schematic. You can see the switch has been replaced by a stereo potmeter. This MUST be a linear type potmeter!!
There is a half drop in volume at the half way stage of the potmeter because we have 50K of resistance in our signal path there but other than that it functions fine! It's just a matter of turning up the input or output levels to get it where you want it. You must make sure the output audio goes into a very high impedance input, like a HiFi amplifier because if you pull even a tiny bit of current from this circuit it won't behave normally anymore. But it works fine on audio amplifiers, I guarantee it.

The Dry/Wet control really makes a world of difference! Now you can set it to reverb and then precisely dial in the effect to where you like it. It's fantastic sounding!
Here is a picture of the panel with the Dry/Wet control where the switch used to be:

I made a little demo video. You can hear the drop in volume at the midway point of the Dry/Wet control. (I know it's wired backwards LOL :)  But being able to dial in the effect makes a world of difference and the volume can easily be crancked up by the level controls.

Continuing the original text:
I've put in 4 buffer stages, using the TL074, for the input, normal output and FX Unit outputs Left and Right channels and I gave the latter two adjustable gain by means of two 50K potmeters in the feedback loops of the opamps; one for each channel. The gain is adjustable from 2 to 5.3 times. You can increase that by using 100K potmeters instead of 50K ones. That would give a maximum gain of 8.6 times.

When I first tested this unit I noticed I was receiving an FM broadcasting signal through the effects unit. (There's an FM Broadcast transmitter and antenna on a flat 100 meters from my location). So I took a ferrite ring and wound the audio input wire around the ferrite ring about ten times. I also put ferrite beads in the 5V power-supply line to the effects unit and to the print with the opamps on it. This solved the problem. One more little thing: beware of the little crystal X1 near one of the screw holes on the circuit board. It is rather flimsy and fragile. Take care not to damage it.

The Cara OK is a really versatile unit with lots of really cool sounding effects. Here's an overview of the possibilities it offers. I myself printed a small version of this list out, laminated it and stuck it at the bottom of the panel I made for it. Handy to have around I thought :) :

This picture shows all the connections to the circuitboard:

It's small so it won't take up too much space. Beware that it needs just +5V for power supply. Luckily in my synth build I made a power-supply that delivers dual 5, 12 and 15 Volts so I can feed it right from there. I can really recommend you picking this up and trying it in your build project. It will add a lot of options and is a very useful addition to the filters and its output is in stereo! The sound quality is just great so no problems there. The only thing is the difference in volume I mentioned earlier but that is easily fixed. You can use opamps buffers with it, like I did, but it's not absolutely necessary. I did without them at first but then installed output buffers with variable gain as I mentioned before.
The audio response of this module is so good that it even lets through the ultra low frequencies the Korg MS20 filter produces (see next article) and that can go as low as 10Hz. You can really see the speaker cones move bigtime!
Before I installed the 3-pole toggle switch I had a single pole and a double pole switch side by side to switch between FX-unit and normal line out. So after installing that 3-pole switch I had a hole in the panel left over. I mounted a 3,5mm stereo output jack in that hole as a connection for head-phones. The output jack is connected straight to the audio output on the stripboard. One thing I noticed with this arragement is that the normal line-out through the head-phones, sounds a lot quieter than when the effects unit is switched on. That's probably due to a difference in output impedance because we're effectively putting an 8 Ohm resistance between Line-Out and Ground in the form of the head-phone speakers. This doesn't occur when I listen to it on the normal audio amplifier, at least not if the head-phones are not plugged in. It would be a good idea to build a little head-phone amplifier for this purpose.
Here is a picture of the finished module in my synthesizer:

Here's a picture of what's behind the panel. Now you understand why this is not a beginners project ^_____^  You can see the yellow Ferrite ring with the black 'audio in' wire wound around it above the blue circuitboard and there's also one on the stripboard. I advise you to include these in the power supply line and audio in line. In red you can see the 3 pole toggle switch. This panel works really well like this.

Here's the Line Out Panel I made on the back of my synth, with two gold plated RCA outputs and a 6,3mm (1/4") Stereo Output Jack, which is connected straight to the RCA left and right outputs.

Okay, that's the 11th module I put in the synthesizer. We're nearly there. I have just room enough left for one more module and that has got to be the Korg MS20 filter. But I'm waiting for some supplies from China before I can build it. (Circuit boards for one, coz I'm fresh out at the mo.)

Okay here's an excellent video by Juanito Moore that shows you how you can circuit-bend this device and make it voltage controllable. Click here

Right, that concludes this article. Thanks for stopping by and while you're here, why not leave me a comment or if you have any questions put those in the comments too and I'll get back to you asap.

Tuesday, 10 December 2019


A very simple VCA circuit that works perfectly. Easy to build too. You do need an oscilloscope to calibrate it though.  

Please read the whole text before building this project. 
The VCA is nothing more than a voltage controlled volume knob. It turns the volume up when you press a note on the keyboard and it shuts it down after you let go of the key. This is all done using signals coming from the Envelope Generator or ADSR or from an LFO depending on what you use it for. Don't mistake a VCA for an Audio Amplifier. You can not hang speakers on the end of this circuit. A VCA is used to make sure your synthesizer only produces sound when you press a key on the keyboard. The keyboard produces a Gate signal that is high as long as the key is pressed down and a 1 Volt per Octave signal that tells the VCO which note to play. That Gate signal then triggers the Envelope Generator and the output from the Envelope Generator goes into the VCA together with the audio output from a filter for instance and as the VCA detects the Envelope signal it opens up and lets the audio pass through with the volume or amplitude depending on how high the envelope signal is in voltage. The output of the VCA must later be attenuated to audio line level if you want to feed it into a HiFi amplifier. So it's just a link in the synthesizer chain.
Here is a block diagram to show you the position and function of a VCA in a synthesizer. There seems to be a lot of confusion about this with people new to (modular) synthesizers:

(Image taken from 'The Complete Synthesizer' PDF Book)

I used a very simple design for the VCA which I again found on the Yusynth website.
Here's the schematic:

This is an old design and there are some updated versions out there. It's very simple but it works very well except that in my build the signal came out inverted. This isn't really an issue because it's an audio signal and they sound the same whether inverted or not but my Obsessive Compulsive Disorderly mind wants it coming out the same way it came in so I added a little opamp inverter to the output to set this straight. But you could leave IC2 out and tap the output from pin 1 of IC1 but you must use the electrolytic capacitor on the output. The output needs to go through a 10µF Electrolytic Capacitor (plus connected to output VCA) because I noticed a 240mV DC offset voltage on the output which I couldn't trim away with the potmeters.
Use an oscilloscope to set the trimmer potmeters. You should be able to measure a DC voltage (before the 10µF cap I mentioned earlier) and, with trimmer R18, trim away as much DC voltage on the output as you can and with the other trimmer R14 you can trim the balance of the signal. You set it so the positive part of the wave has the same amplitude as the negative part of the wave, with the zero volt line being the dividing line.
The output level may be a bit lower than the input level, even if the ADSR potmeter is fully opened up. If that's the case and you want to correct that (which is not necessary if you're using it for output into a HiFi amplifier) then you can change the gain of the output buffer opamp. This is not in the schematic but in the layout. If you change the 150K resistor between pins 1 and 2 of IC-2 for a 470K resistor, you should get 3 times gain! That should bring the level back to input level. You can experiment with this yourself. If you put a 1 Mega Ohm potmeter between pins 1 and 2 you can control the gain of the opamp with a knob on the panel. Just a thought ;)

Here's the layout I made for standard 24x55 hole stripboard using only 24x39 holes. Like I said before, I added a signal inverter in the shape of a second TL072. Only one of the two opamps in the chip is used, the other one is properly connected to ground. The panel potmeter values are not critical in this design. Logarithmic pots would be preferable because we're dealing with audio signals but linear will work fine too. You can also use other values like 47K or 100K because one is an input level control and the other a voltage devider switched between 12V and ground so the values have no influence on the working of this circuit.
The layout below is verified and absolutely faultless. I guarantee it. It has been used successfully by many people now.

Print only:

Here's a picture of the finished VCA installed in my synthesizer:

The Level control is for the ADSR input signal and determins the volume of the audio signal. The Gain controls the quietness of the VCA when no keys are pressed on the keyboard and it should normally be set to zero. If you turn it up, the last note you played will become audible.
You must use an oscilloscope to test the signals and trim the offset and signal levels. Put it in DC mode when testing. 
In the picture above you see a yellow patch cable connected to the audio output. That is actually my oscilloscope probe, so I can see the output signal on my scope screen, and the audio output is connected internally, behind the panels, to the Line-Out and Effects Unit to the left. From there the audio signal goes to a stereo RCA output on the back of my synthesizer and from there to the Line-In of my HiFi Audio Amplifier.

Below is a picture of the double VCA that I built on April 25th and 26th of 2020 from the same schematic and layout. These VCA's work like a charm! They are so handy to have, I use them a lot in patches as a sort of Gate to let audio through when there's a signal on the ADSR input.. When the ADSR potmeter is turned fully open (clockwise) the output signal will have the same strength as the input signal. Of course this is also influenced by the type of control voltage you feed it. 
With the double VCA I wired up the inputs in such a way that when there is no connection made to input 2, that input gets the same signal as is present on input 1. So with one input you get two outputs. As soon as a patch cable is connected to input 2 that connection with input 1 is broken (by the socket switch of input 2) and the VCA's work as two independent VCA's.

A look behind the panel:

If, after building this VCA, you have trouble with noise, especially at low volume levels, then you most probably need to replace the opamp you're using. I've had people contact me about this and it turned out the opamp was the cause. An other cause of excessive noise has been bad soldering joints. So make sure your solder connections are good. 
Like I said, I built 3 of these and they all work fine and are absolutely quiet.

Here's an oscilloscope picture of how the VCA lets through audio (the yellow waveform) only when it gets the Control Voltage from the ADSR (the blue line). You can see how the amplitude of the audio closely follows the amplitude of the CV from the Envelope Generator (ADSR):

Okay, that's it for now. Any questions? Put them in the comments below and I will answer them asap. You can also post questions in the new Facebook Group for this website.

Synthesizer Build part-9: THE LFO or LOW FREQUENCY OSCILLATOR.

Before we start, this article is about my first attempt at building an LFO based on the AS3340 chip and it didn't work out very well. If you want to read about the LFO that did work then click here to go to chapter 13

The LFO is an indispensable  part of any synthesizer. It is mainly used to modulate other parts of the synth like the filters or the VCO's.
The LFO I build for my synth project is nothing more then a cut down version of the VCO based on the AS3340 chip. I left out a few things, like the high frequency track and Hard- and Soft sync options and I added an inverter for the saw tooth wave. (I was just getting into synthesizer building when I made this so I made some stupid decisions.)
The low frequency is achieved by using a 100nF capacitor from pin 11 to ground instead of the 1nF used in the VCO. You can experiment with this by using different values of caps on pin 11. In fact I made a switch in the LFO panel with a choise of a 270nF for extra Low Frequency and a 100nF for normal LFO use. The total reach is about one wave every 8 seconds to 20 waves per second.
The reverse sawtooth wave offset voltage must be set to zero volt using an oscilloscope.
Here's the layout I drew for it. The schematic is just the same as the AS3340 datasheet schematic.

Note that there is no squarewave output on this LFO. When testing this design I got so much ringing on the downward slope of the squarewave that I deemed it unuseable for LFO use. I therefore made a separate squarewave oscillator using the CD40106 design from the 8 Step Sequencer from the previous article. I made it on a piece of stripboard that simply hangs from the 100K frequency potmeter which is soldered straight to the PCB. One plus point of this approach is that you can have the squarewave going at a different frequency to the other two waveshapes which could be useful for triggering drum modules etc. Be careful that the squarewave oscillator doesn't touch the panel. When I first build it the contacts from the potmeter touched the panel and the CD40106 literally went up in smoke! My whole attic stank of magic smoke for a day! So put some gaffer tape on the contacts or the panel to be safe.

As you can see in the picture the LFO has a CV input and an FM input to control the waveforms with other modulation sources so you can create more weirdness :)
This LFO has a weird quirck that I actually like a lot. It has a bit of a beat when it dies out. When the wave dies out it doesn't do so quietly but it pops a little. I'm not sure why this is but I don't mind it. The waves consist of a negative and positive voltage part so the middle of the wave is around the zero volt line. That might have something to do with it as most of my modules work with signals from zero to plus ten volts. I don't know, but if you do please let me know what causes the plopping sound in the comments below. :) Btw, don't let this withhold you from building the LFO. It works fantastically.

[EDIT-1: Friday the 13th of December (that figures, LOL)]
I changed an output jack for a new one because it kept getting loose and I switched the panel back on and boom, up went the CD40106 again! Magic smoke (nothing magic about it in my opinion but there we are ^__^ ). So I tried fixing the ringing issue with the squarewave from the AS3340 with as result that now only the triangle wave is still working. It's going from bad to worse with this module so I think I'm going to scrap it and build one of Yusynth's LFO's. Luckily it's not a vital part of the synth but naturally you have to have a good and well working LFO in your setup. I saw a Yusynth design that can be synced to other signals which is really cool because the LFO then engages when you hit a note on the keyboard. That's what I want. So more on this later.

EDIT-2: Friday 27th of December-2019]
I tried the Yusynth LFO, in fact it was an improved design because the original has some issues with the sync pulse. A fellow member of the Synth DIY Facebook group provided a layout and schematics and I build it but I can not get it to work. I can't figure out what I did wrong but I'll get it working one day.
In the mean time I've build an LFO from a schematic I found on the MusicFromOuterSpace website.
This one has no sync option but it is a very cool LFO and a very simple design, with sine-, triangle-, sawtooth-, ramp- and squarewaves. The transition from saw to triangle to ramp is continous by means of a 50K potmeter. The wave amplitudes are +5V to -5V but I added a few opamps with DC-offset functions so I also have outputs for the sinewave and the saw-triangle-ramp waves that go from 0V to +10V.
I had a lot of trouble getting this one to work to untill I checked my symmetrical powersupply and noticed it had +1.4V and -20Volts. So it's obvious it doesn't work that way. (Btw, that is not the cause of the synchronized LFO not working.)
I'll be adding a new article about the MFOS LFO as soon as I built the panel and have it all installed so I can make some pictures or video to illustrate the article with. Stay tuned.

Okay, that's it for this one. I hope you enjoyed it. If you have any questions please put them in the comments and while you're here leave me a comment anyway please! More synth related stuff to come!

Monday, 9 December 2019

Synthesizer Build part-8: 8 STEP SEQUENCER.

A simple 'Baby 8' type Sequencer made with the CD4017 chip. Easy to build and fun to use. No DIY synth should be without one of these.

This sequencer is one of my earlier projects and of my own design although it's more or less put together from bits of other designs like the 'Baby 8' but it works fine for me and is really easy to make and easy to tune although to build it is quite time consuming and repetitive work because a lot of steps have to be soldered eight times. I found it rather tedious work but very worth while. 

Here's the schematic drawing for this sequencer:

Here's the stripboard layout I made for the sequencer. In the schematic I drew in switches that you can add to turn individual channels on or off but I didn't include them in my build because I didn't have the space for them on the panel. In this layout I don't use any transistors either. I thought it was nonsense to make this more difficult then it needs to be. It will work fine without them because we hardly draw any current from these outputs. The CV output signal goes straight into a VCO. The layout has an extra 10µF electrolytic capacitor on the output of the voltage regulator that is not on the schematic. It's for extra noise suppression. You can get away with using a 100nF cap or leaving it out completely.
Be careful when you wire this up, note that the jumper (or wire bridge) for output 5 is connected to pin 10 of the chip so the left bunch of jumpers skips a copper trace at output 5. Look carefully at the layout! If you want to include switches to mute individual channels then put them in series with the diode!

(Last revised: 26-Feb.-2020: Minor cosmetic changes.)

NOTE: All potmeters in the layout are shown from the front side!

Use Schottky Diodes on the wipers of the potmeters. They only have a voltage drop of 0.2V instead of the 0.6 to 0.7 Voltage drop over 1N4148 diodes usually found in sequencers like this. This means you can get deeper tones from the VCO you plug it into. Because of the 0.6 to 0.7 Volt voltage drop over the silicone diodes, the first section of the potmeters wouldn't do anything until you get above 0.6 volts. So with a lower voltage drop there's more throw on the potmeter. As an experiment I also installed a 100K potmeter over the output of the Control Voltage and the wiper goes to the CV output jack. That way you can get even lower tones although, of course, this compresses the dynamic range of the sequencer. With the potmeter fully open you get the normal range of 0.2 to 8 Volts. If you close the pot half way, your range becomes 0.1 to 4 Volts so the spacing between notes becomes smaller. You don't need to include that option, I never use it and it is not included in the layout. But anyway, this is an expirimental sequencer and as a whole it works really well, If you build it you will be happy, I guarantee it. :)
A better solution, and one you should consider if you are comfortable with designing simple circuits with opamps, is to add a DC-Offset feature to this sequencer. That way you can get the lowest notes down to 0 volt without influencing the dynamic range of the sequencer. It's easy enough to do. This is not included in the layout or schematic though.

Here's a close-up of the stripboard:

Bill of materials for the layout version:

Here's a picture of the sequencer:

The sequencer is build up around the CD4017 decade counter chip, using a CD40106 to create the clock pulses which also serve as the 'Gate' pulses.
The CD40106 hex inverter is used as a low frequency oscillator giving off squarewave pulses who's frequency can be controlled by the 100K potmeter. I used a 15µF electrolythic Capacitor although a 10µF will do just as well. But a little higher value will give you slower speeds so you could even try a 22µF cap. The clock pulses can be interrupted by switch S-2 to give you a chance to tune that particular channel. Sometimes it can happen that after using the 'Stop/Run' switch that the sequencer jumps to channel one. If that happens try using a different CD40106 chip. You might have a fake one and they can be quircky in their behaviour.
With S-2 closed the clock pulses go into pin 14 of the CD4017 and with every pulse the chip will output a high signal on a different pin. The order by which the different pins go high is a bit random. Here is the right order: 3,2,4,7,10,1,5,6,9,11. Because of this confusing order, the outputs are set in the right order by the wire bridges to the copper traces underneath the CD4017. From there the pulses can be accessed in the right order to avoid confusion. Following the schematic drawing, the pulses go straight into the base of the 2N2222 transistors which are used here as switches. The Base-Emitter voltage is way more than needed to saturate the transistor and fully open it up. I chose the 2N2222 transistor because it can handle a reasonably large current and there's no need to use any resistors to connect them (although using a resistor in series with the base connection wouldn't be a bad thing because we're using the 2N2222 at near the limit of the operational specs.) From this base connection we also feed the eight LED's which indicate which channel is on at each moment in time. The LED's are connected with 3K resistors to reduce current flow and still provide a bright light.
All the collectors of the transistors are connected straight to the 8 Volt power rail and the emitters are all connected to ground.
It's better to just follow the stripboard layout and skip the whole transistor setup and connect the output of the CD4017 straight to the potmeters. I'm using transistors as a sort of buffer and to make this sequencer future proof for other experiments so I can draw some current from the outputs should that be necessary. But you can just leave them out it you want to. Makes it so much easier.
By setting the different potmeters, you can create the different tonal paterns the sequencer produces.
Because the potmeters are simply used as voltage deviders, it doesn't really matter which value they are as long as it's 50K or over so that they don't draw too much current and as long as you use the same value on all 8 channels.
You can tap the 'Gate' pulses straight from pin 3 of the Speed Control potmeter to the Gate output jack mounted in the panel. The pulses are really clean looking 8 Volt squarewave pulses with a 50% duty cycle so if you use the gate output into the ADSR, it will sound as if a key is pressed every time the sequencer switches to an other note.

A ten step switch is used to select the length of the sequence. It can be anything from 1 to 8. Btw, you can easily make this a ten step sequencer by connecting the last two pins from the CD4017. I made it an 8 step because I didn't have enough space to mount everything horizontally and because 8 steps is more natural for music then 10 steps because you normally have 4 notes in a beat. So multiples of 4 are better. The potmeters on my panel are mounted vertically and I could only fit eight of them below eachother anyway.
Connect the wiper part of the switch to pin 15 of the CD4017 and the wires from 1 to 8 to their relative position on the switch. Connect pins 9 and 10 of the switch together and connect the ninth output from the CD4017 to that. The pulse going into pin 15 of the 4017 will reset the chip and the counter will start over again.
Don't forget to connect pin 13 of the CD4017 to ground.

It is best with this build to make the panel first and connect all the components and do the essential wiring while you have access. Then make the circuitboard and connect the wires to the panel. Solder the resistors straight to the LED's and the diodes to the wipers of the potmeters. Connect the cathodes together and solder a wire from there to the CV output jack.
I used 5mm LED's and I made the holes in the panel by using a drill rather than a hole enlarger bit which I normally use to enlarge the pilot holes I drilled. The drill is usually a little bit less then 5mm and therefor the LED's will sit very tight and don't even need to be glued in place (although it is best to hot-glue them in place anyway).

Do not forget to solder a big 470µF capacitor on the input of the 7808 voltage regulator. Otherwise pulses will bleed through onto the power supply rails and you'll hear the tone sequence even if the sequencer isn't connected to the CV input of the VCO. I also included an ON/OFF switch (S-1) on the panel just to have the option to shut it down. It's the only panel in my synth build to have an ON/OFF switch.

To tune the sequencer, simply set it to the lowest speed and use switch S-2 to interrupt the clock pulses and stop at each channel. Then you can tune that particular channel using a tuner or simply by ear, by turning the potmeter and then you turn S-2 back on. The sequencer flips to the next channel, you turn it off again with S-2 and tune that note, then you flip the switch again and jump to the next channel, etc, etc.
It's very simple and very effective. :)
I tried putting in a momentary switch to jump channels and connected it to the +8 Volt rail but this didn't work very well. Maybe my switch is poor quality though, I don't know. But you don't really need a manual switch if you can use the slow clock pulses to switch channels for you.

That's all there is to say about this. It's one of the most fun panels for the synthesizer but one of the most tedious to build. It cost me 6 hours straight to design and build it but luckily it worked straight away.

Here's a little demo of the sequencer. This was filmed before I put in switch S-2 so I had no option to tune the sequencer at the time of filming. I might make a new video soon:

Okay, that's another one done. I hope you enjoyed it. If you have any questions about this build then  please leave them in the comment section below or in the Facebook Group.

Synthesizer Build part-7: THE MOOG LADDER FILTER.

Okay, here we have a filter who's sound is iconic and instantly recognizable if you know the music by Jean-Michel Jarre, Giorgio Moroder or Vangelis and (thousands of others really). The Moog Ladder Filter produces that sharp, snidey, ripping the fabric of the universe synthesizer sound and also that wet, watery sound you sometimes hear (if you use it it with loads of reverb). That's why I had to include it in my synthesizer build project of course.

Since this article was written I have made a new version of this filter, this time including the CA3046 transistor arrays and it works very well and no need to match transistors with that version. If you want to build this filter I would really advise you to use that layout instead of this one (although this one works fine too of course). You can find it in Chapter 39 (click here).

I used the schematic from Yusynth's website.

Before we start. Most people always want to know if it works on 12V. I tested the filter on dual 12V and it works just fine.
In this schematic the top and bottom transistors are applied in the form of a transistor array chip, the CA3046, but I couldn't get hold of that quickly enough and this early in the build I didn't really trust myself to design a layout including those arrays, so I decided to use all transistors and that works just as well. It makes the layout a lot easier. It is always mentioned that you must use matched pairs of transistors for this filter but really, that's a throw-back to the early seventies when transistors were not as consistent and reliable as they are now so if you have transistors from the same batch they will probably be matched well enough but put them through the transistor tester on your multimeter and match them on hfe value. The only place where the transistors must be matched well is on the place in the schematic where they use the CA3046; the top and bottom of the filter and the output on the side. I personally matched all my transistors by using the Transistor Curve Tracer I described in an earlier article on this website.
I built a second ladder filter as a test for the layout below and I used all unmatched transistors. The layout works fine but using unmatched transistors did not turn out well. I could not get the resonance trimmed correctly and there were enormous differences in volume when using the resonance potmeter. I used a squarewave for testing and the top of the squarewave had an angle to it instead of being horizontal. So you must used matched transistors!
This filter has a few quircks that you need to know about but which are normal for this design.
- The Resonance potmeter has only a small area of influence. For most of the throw of the potmeter you will hardly notice anything. This is normal for this design. That's why we need a reversed logarithmic potmeter for Resonance. To stretch out that last bit of the potmeter.
- When the Resonance is fully open, the output volume drops. This too is normal for this filter and even the original Moog ones have this. Yusynth also talks about this on his website.

The build is quite straight forward but you need to be very accurate. The 50K anti-logarithmic potmeter for the Emphasis or Resonance control was an other thing I couldn't get a hold of so I made my own by using a linear potmeter with a 5K resistor between pins 1 and 2. This works very well, In the layout I used a reversed logarithmic potmeter and I show the alternative that I myself used, next to it.
The input level potmeters are 50K logarithmic ones but if you don't have those just use linear ones. They don't even have to be 50K. You can use 100K or 1M or even 10K if that's what you have available. They're just audio input level pots so they act as attenuators or voltage dividers and the value has no impact on the working of the circuit. This goes for all the level potmeters in all the projects on this website unless it is mentioned otherwise on the layout.
The Frequency Cutoff potmeter must be a 10K!

I made a layout for stripboard including the wiring. I used this layout to build a second filter and it worked straight away so this layout is verified. (All potmeters viewed from the front):

(Last revised: 24-June-2020: Corrected polarity of C3. 15-July-2020 added alternative for reversed potmeter.)

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

Here are a few pictures of the finished circuitboard:

As you can see in the pictures, I added two trimpotmeters which are not on the stripboard layout above. These are two 200K trim pots and they go over pins 1 and 2 of each opamp, to make the gain adjustable. It says in the schematic to 'adjust the value of the feedback resistors according to audio level'. These trimpots make that possible without having to use the soldering iron. It's a bit awkward with the wires but I had to put the potmeters on the print where there was room enough to accommodate them and the wires. Plus I added them as an after thought, so after I made the layout. At least they are all neatly in a row. :-)
For clarity I made a second layout which includes these alterations. If you decide to replicate this then don't forget to remove (or not solder in) the original resistors over pins 1 and 2; the 56K on IC-1 and the 120K on IC-2 because these are replaced by the trimmers, as this layout shows. Lay-out is verified not only by me but I heard from many people who built this filter successfully using this layout.:

I lowered the Control Voltage input resistors from 100K to 5K6. In the schematic they are 100K but after installing the filter in my synth set-up I noticed that the CV hardly had any effect at all when I connected a LFO to it, so I lowered these to 5K6.

After finishing the build I tested the filter on the oscilloscope first and set all the trimmers to the right positions. It's easy enough to do, you just watch the scope for the best response. Then, after installing the filter you can adjust the trimmers to get the best sound out of it. This filter is self-oscillating, meaning that if you have nothing connected to the inputs and you turn Frequency Cut-off and Resonance all the way up the filter will oscillate of its own. There's no 1V/Oct. input though so resonance won't keep track with the notes on the keyboard. An other problem is that the self resonance only occurs at the top of the squarewave and not the bottom part. When you start it up it will self oscillate on both the top and the bottom but as soon as the transistors warm up, about 20 seconds, the bottom oscillations disappear. This is due to the fact that I have used transistors that were not perfectly matched. Using the CA3046 chip would solve this and that's why I made a second Moog Ladder Filter project using the CA3046 chips. You can find that in chapter 39
The first all transistor filter I built didn't have this problem so if you're careful with matching the transistors you should be fine. Please note that the input volume is also an influence on this. If the level is too low you can also have the bottom oscillations disappear.
Here are some oscilloscope images to illustrate what I mean:
Self oscillation when filter is just switched on:

And here's the situation after about 20 seconds. The bottom oscillations are gone. The filter still sounds pretty cool though:

This is something I only discovered a short while ago but well matched transistors or using the CA3046 chip should solve this issue. The first Ladder Filter I built using all transistors didn't suffer from this problem because I was careful to match the transistors accurately.

Here's a video showing the test results on the oscilloscope so you have an idea of what the waveforms should look like. In this first filter I built and which I demo in this video the transistors seem to be reasonably well matched because I do retain some self oscillation on the lower parts of the squarewave:


A demonstration of the sound of the filter:

Here's a look at my PCB for the filter. This is what is actually in my synthesizer and I build it from a different layout that I made earlier especially for the Eurocard format of stripboard:

Here's the panel I made for it:

As you can see in the picture, I've installed a bypass switch for this filter. It's great to have this filter in series with the Korg MS-20 or the ARP2600 filter but sometimes you want to be able to easily switch to one filter. With a bypass switch there's no need to constantly connect and disconnect patch cables so this is really helpful. The switch only works with audio input 1 and it sends the signal straight to the output jack and disconnects the in- and output from the in- and output jacks at the same time. Here's the wiring diagram for the bypass switch:

Okay that's it for this episode. Stay tuned for more synthesizer build articles and while you're here leave a comment please!