Archives for : resources and tips

Akai S1000 Upgrades

Installed the LED based display. It is quite bright if not too bright actually.

So I’ve drilled a hole in the back and installed a 1k LOG pot there.

And installed 22 ohm resistor in series. Together cutting on the LED supply current. Display can now be dimmed. At fully dimmed position it is roughly as bright as stock EL Foil based one. Important, the resistor is crucial, else you might damage the LED unit.

I’ve installed a FF based Gotek drive. Actually the stock M0 jumper setting turned out to be working perfectly fine but I have moved it to S0 position later, to be on the safe side. FF is superior to HxC as it will never make the unit freeze during floppy disk change phase (if you change it too fast).
EDIT: I just noticed the Gotek had an old HxC firmware. While the FF was the latest version.

New power supply in. This is a must! It is known that many Akai S1000’s have been damaged due to faulty PSU’s that are now showing their age (expired resource). This Akai came with a Meanwell RS35-5 power supply which previous owner sent separately cause he didn’t know how to install it. I’ve drilled two holes in the bottom of the unit and screwed the new power supply in. It is crucial to either restore the existing power supply or purchase a new one. Else you are risking to seriously damage the unit!

You can use existing wires just fine. As long as you know what you are doing. Disclaimer: I will never answer any question that involves electricity.

Ok the memory cards can be tricky to install correctly and work if you have mixed sized boards. Here is the setup that I have found via trial and error. Slot 1: 2MB Slot 2: 8MB Slot 3: 8 MB Slot 4: 2 MB

Once you’re done, toss it on top of the S1100 just so you can show others you have both.

Matrix 1000 Factory Presets

So you changed the battery in your Matrix 1000 and now all of the presets are gone. You go to the web to search for the factory patches, only to find incomplete archives. Typically people would modify a few of the factory presets, probably forget about it and then upload it online not knowing their bank is no longer genuine. I know the feeling, cause I downloaded all of the banks from the web pages, forums, groups, etc only to learn the patches that I hear do not match the official patch list. So I decided to figure out what’s going on and where the original factory bank is. TLDR: it isn’t on the web (well, it wasn’t until now), I had to manually rebuild it, took me two days to gather everything and rebuild it piece by piece as Lieutenant Columbo would do. A lot of help was in the patch listing names, i.e. it says Brass yet there is a helicopter preset on that location, but in another copy you actually find that Brass patch and figure out this preset was not replaecd or modified with something else. One of the banks that helped me a lot was Moby’s own M1000 bank. It was so fortunate that he did not edit from the front of the bank (which is what people normally would do) but from the back of the bank and that was the game changer. At this point I knew I have something useful to start with! The rest was easy. I went into the patch librarian and moved the correct patches from several banks into the final edit version.

All in all here is the archive for which I guarantee 100% it is accurate and contains all of the factory presets for the Matrix 1000 synthesizer. Patch names can be downloaded from below as well. Keep in mind only two first banks are the user area, the rest is the ROM section which can not be deleted or manipulated. Also to clear one little myth that floats around, not all of the patches that are in the User area originate / exist in the ROM. Yes, many are from the ROM section, but there are at least 60 patches that do not exist in the ROM section and I believe they were exclusively designed for the Matrix 1000.

Matrix 1000 Factory Patches – as the title says, this is the archive that contains Bank 0 and Bank 1

Matrix 1000 Patch Names – the above file is not much use without knowing the patch names. So here they are

Why would anyone want factory patches?
Some people prefer things being genuine, and let’s just add to the fact that back in the 90’s Matrix series (6, 6R, 1000) were all over electronic music. It was literally the go-to synth for that genre, as it had MIDI yet was very affordable, typically half the price of Xpander and MKS-80 which was something not everyone would afford. Don’t let the “ahh the early 90’s while analog synths were cheap” phrase fool you. MKS-80 and Xpander and Matrix 12 were NEVER cheap, even back then out of reach for many, because they all had MIDI. Now…. the Jupiters 4 and 8 were cheap… You could buy either of the two or the Matrix 6 for the same amount of money, but the Matrix had MIDI, which the former did not.

Feel free to comment, or add something or just say thanks in the comment section below. And now it’s time to enjoy the Matrix 1000.

Korg DSS-1 Factory Library for Gotek Flash Floppy & HxC owners



Assuming you bought a Gotek Flash Floppy (eBay et al.) and decided to upgrade your Korg DSS-1 there are probably a lot of questions bothering you. To save you time this page is here to provide all the basic steps to get your system up and running. This setup could theoretically work on HxC, but you will have to ask / search on their forum about the configuration file. I spoke with the author of HxC he is a great guy and always willing to help so don’t worry you’re safe. The images from this library are in .hfe file format and will work on HxC, while for the setup you will probably have to look on forums. If your Gotek is a Flash Floppy type then ignore previous three sentience’s and continue reading on!

Hardware setup
If you installed a Gotek drive (ideally the one with the OLED display), buy a USB stick as small as possible. Format it as FAT32, or if working on a Mac on OSX this is known as MS-DOS style partition. Inspect the Gotek drive and make sure it has a jumper on S0 pins and make sure other pins do not have any jumpers. If you want sound (of virtual floppy clicking) you might want to buy one of those tiny PC speakers and install it into a Gotek by connecting it onto pins marked as JB.

Software setup
There’s nothing really to set up. Simply extract the .7z archive onto USB stick and you can use it immediately. Flash Floppy configuration file (FF.CFG) is already in the archive. If you want you can edit it for your own fine tuned setup, it’s just a standard text file with each line described what it is and what it does. If you decided to upgrade or downgrade your Gotek with and need a good reference on how to setup the configuration file for your own suits, or just feel like a nerd and what to know what each flag does, please follow this link. To remind you, the one which is included in this archive is working just fine.

DSS-1 Library
This the reason you came here, right. But please read this first. While there are various web sources online that provide DSS-1 Library, unfortunately many of them are incomplete / contain corrupt data or contain duplicates and duplicates of duplicates, or are in a format that does not work with Flash Floppy and HxC. This one is different. I’ve decided to start from zero and slowly build or better to say precompile a “new” library that contains all of the DSS-1 images from online, converted into .hfe format, all of the duplicates removed, and corrupted disks replaced with valid ones. There are a total of 144 disks. They are all in .hfe format ready to be used in Flash Floppy and HxC Gotek drives. The library can be downloaded from here:

Korg DSS1 144_disk_Library (64MB) Kindly: do not ask me to add any commercial disks in this library I do NOT support piracy!

What’s inside?
What’s the use of such a huge library without anyone knowing what’s inside. Well we can certainly change that. I took some time and built this large table that covers all 144 Floppy Disks. The table is located here: One huge table

I have a Gotek but don’t have Flash Floppy or HxC what to do?
Fair enough! I assume there are some folks who bought a native “raw” Gotek drive or have one lying around unused or just want to save a buck or two. Don’t worry we got some good news for you. If you know your work with a screwdriver, a paper clip and have USB-A to USB-A cable, you can easily upgrade your Gotek to Flash Floppy for literally free of charge. The instructions are super simple and available here: Gotek FlashFloppy EZ Installer

Below there is a comment section. If you think there is something that can be improved or just wanted to say thanks, you’re welcome. Now go play that DSS-1! Those of us who are lucky owners know how good it sounds and leaving one gathering dust is a sin. 🙂

Akai S-950 Upgrades Part4: DD & HD floppy to image conversion

So your Akai came with a bunch of DD and HD floppies, you converted them to images, started a Gotek drive only to be greeted with a message saying unformatted disk, or no disk in drive, etc and you have to facepalm yourself. I know the feeling!

The source of the problem is that Gotek HxC does not like having both DD (double density) and HD (high density) disk images on the same USB stick. Even if you set the configuration file to Auto, and use .hfe rather then .img files it just refuses to work. In fact it took me whole day to find the combination that works. But eventually i found a solution to have the content of both DD and HD images on the same USB stick. You will have to do exactly as described in here, else the things just won’t work.

A bunch of DD and HD floppies that we will eventually put onto this small USB stick.

First thing you will need a Windows based computer with a floppy disk drive. You will install HxCFloppyEmulator software onto it. You will then insert your Akai floppy and press Floppy Disk Dump. The program might ask you to install an extra driver in here, so install it if it says so. Once the floppy is being read, use Export, and export it as .hfe file. Make sure you label the files sequentially as DSKA0000.hfe, DSKA0001.hfe etc. I suggest you dump first all of the DD disks, then the HD disks into separate folder.

Here are the same floppies in .img format. I used Omniflop on my old WinXP machine to read the floppies because i thought i would use them as .img files. In the end this method just didn’t work right with mixed DD & HD content, so i advise you to go directly .hfe export via HxC2001 software. Pro tip: Don’t bother with OmniFlop.

Keep in mind what is being described in here applies only if your Akai came with mixed DD and HD floppies. If it only came with HD then you are set already and can write the config file to your FAT32 formatted USB stick, with following settings.

These settings work 100%. Save this config to your USB stick and you’re set.

Now comes the tricky part. You will need to use two USB sticks temporarily. This is in fact the only solution that worked. Eventually it will all end on one single USB stick so this is just the temporary phase. Take the second USB stick, make sure it is FAT32, start HxC2001 and make another configuration and export it onto that USB stick. The configuration will differ slightly, instead of AKAI S950 HD set under the Mode, you will have to choose S950 DD. Think of these two USB sticks as two different floppy drives: one is the DD (double density) the other is HD (high density). Hence one will contain only the DD images, while other only the HD images.

Now count the number of DD floppies. Let’s say you have 12 of them. That means you will need to generate 12 HD empty floppies. Go into HxC, make sure you have set it to S950 HD and generate an empty floppy image. Please read HxC “Floppy Emulator Software – Step by Step Guide” to learn how to generate an empty floppy image. Make sure to select Predefined Disk Layout for Akai S-950. Once the file has been generated you will need to copy it 11 more times. Pro tip: You can in fact copy them a few more times, because it doesn’t hurt to have extra spare few empties for your own sampling purposes, and ONE extra empty that you can archive, so that you don’t have to start HxC2001 software each time you want an empty S950 floppy image. Always make sure to label these new files sequentially. For example: Let’s suppose you have 15 HD floppies that you converted into images and have put them onto an USB stick. The are labeled DSKA0000.hfe – DSKA0014.hfe, that means you will need to label your empty HD images starting with DSKA0015.hfe and continuing sequentially up. More importantly there can be NO gaps between file numbers. The sequence always must be continuous, ie: 0017, 0018, 0019…

The image above shows that we have two USB sticks. On USB stick one we have DD floppy disk images with appropriate HXCSDFE configuration file that we generated earlier for S950 DD floppies. The second stick contains our HD disk images, with appropriate HXCSDFE configuration file for HD images, plus our new empty images that we just generated (shown in blue), plus a few extra empties (shown in purple color) for our own sampling.

What we are doing here is getting 12 HD floppies on which we will save our 12 DD floppies. Because for some reason Gotek does not like having both HD and DD flopy image files on the same USB stick. And this is the source of all the problems, and why we are doing this workaround at the first place.

Now it’s our time to start the “conversion process”. Our goal is to convert our DD images into HD format, so that we can work with HD images only, because that’s how Gotek HxC wants to work, and there aren’t many alternatives around. First you will insert the USB stick that contains DD floppy images. You will start the Akai, and load whole disk using DISK / 02 Clear mem & load disk. Once the DD disk has been loaded insert the second USB stick and save the content of your memory onto the HD disk using DISK / 05 Clear volume and save entire memory. And that it pretty much it!

Remove the second USB, insert the first one and repeat the process. Don’t forget to choose next disk on your Gotek drive using Next button. In our example we would insert USB stick 1 and load a file called 000, we would then insert USB stick 2 and save onto the file called 015 (shown on Gotek’s LED display). Once you finished all DD floppies and they have been saved onto HD images, you can toss away the USB stick with DD images and from now on only use second USB stick which is all HD images.

Doggo approves!

Akai S-950 Upgrades Part3: Floppy drive to HxC Gotek conversion / upgrade

So i eventually made an order, despite the fact the floppy was working just fine, floppy emulator is a way better solution. Just a few days later, the package was here with HxC modified Gotek drive. Special thanks to Acid Mitch.

The floppy is held by these four screws on the bottom of the unit. Remove them.

Now you need to remove the power supply board in order to reach the back side of the floppy drive. And with it being reachable, detach the ribbon and power connector from the existing floppy drive.

Remove this metal frame from the floppy drive. Do NOT rotate anything. Leave the floppy drive aside and take the new Gotek drive and connect it exactly as the old floppy drive was connected.

The result should look exactly like this.

Insert the new drive into the unit, and connect the ribbon and power connector, then screw in the four screws that hold the floppy drive in place.

And here it is. Floppy out, Gotek HxC in.

Akai S-950 Upgrades Part2: LCD display upgrade and modification

Some people are just greedy. In fact it was one of the eBay listings that made me inspire do a thread like this. Just take a look at these prices for these “kits” and below i will show you what is the actual “kit” in here. It’s one single strip connector that cost $0.5 and two wires that are already on your Akai! So someone soldered this strip onto their LCD screen and called it a “S-950 Kit”. Very funny!

So, instead of going for this:

Go for this:

Now in order to use your new display you will need to take out the old one and take a look at the strip connector. That is the type you need. It has to be the same shape, bent, rather than straight. Keep in mind, Akai already has everything else to connect the display. The ribbon cable can simply be detached / attached, no desoldering of the ribbon needed, making this super simple modification. You just need to solder this new strip connector and you are done. Make sure you cut the new strip connector to be 14 pins and not 16, else you won’t be able to attach existing 14 pin plastic connector to it.

Here it is ladies and gentlemen, this connector is what makes S-950 LCD display “kit”.

Make sure to remove the noisy inverter. Your Akai no longer needs it. DO NOT throw away the connector that was attached on the front panel, instead cut the wires exactly as shown above while keeping the connector – you will need that connector later to provide power supply for the LED backlight.

With the new display ready, glue three spacers onto three holes where the original screws went thru. I’ve made spacers from the voltage regulator spacer but i cut the inner part of it (the one that goes into power regulator). Every electronics store has these insulators, since you need them to insulate your voltage regulator before you screw it to the metal case – else the regulator will short. And don’t throw away original screws. Believe it or not, but they will fit perfectly.

As I’ve promised, original screws fit perfectly. However there is one problem…

We have to put the screws from the back side. In order to do that you will need to unscrew the front metal part of the case and slightly put it at the angle in order for the screwdriver to reach the lower hole of the LCD screw.

Now remember that power inverter connector that i told you to keep? The two wires that it is providing are exactly what we need. They are 0 and +5V power supply for the LCD backlight. You might have to extend these two wires to be able to reach pin 15 and pin 16 of the LCD. Make sure you check with the multimeter which wire is 0 and which is +5V and read LCD specs sheet about which pin requires 0 and which +5V.

Or if you want to go full pimp mode, connect just the 0 wire to the LED power supply pin, and send the +5V to the potentiometer on the back of the unit. The specs is 1k LOG (usually labeled as 1kB). You want to connect it as shown in the image above (left and center pin, looking from below) to apply gradual resistance to dim the backlight. Again, these new LCD backlights are a bit on the brighter side, and this mod lets you dim the brightness and thus largely increase the contrast of the display.

Quick test on the bench shows success. Display’s “blue” color is artifact due to camera’s color balance. Display is actually white/black. And once you dim it, looks almost OLED like super sharp and contrast-y!

Akai S-950 Upgrades Part1: Power connector upgrade & cosmetics

Step one: disassemble the unit.

This is the easiest of the mods. For some reason some of the Akai S-950 use a 2 pin power connector which is not compatible with a standard IEC connector and thus our standard power supply cables.

This connector belongs to bin.

You will need a good ole file tool and a soldering iron. Simply remove the old connector. Use a file tool to widen the existing hole and insert the new connector.

You will notice a small board to which the existing connector was soldered to. Add a new wire there and solder it onto the grounding hole pin. In the image above, the new wire is the black one (the one without transparent plastic insulator!), and the location for the soldering point is shown. The length should be the same as the existing two so that you can reach the connector. Then insert the new IEC connector and solder all three wires, with the new wire that you just added going to the middle pin. That’s the ground pin connection. So that from now on your Akai not only accepts common IEC cable, but is now grounded properly.

The good thing is that the holes for the screws will fit prefectly. You just need to file the existing connector hole a bit (from all sides!).

With the unit disassembled, you can do some washing now.

And with generic grey matte spray paint, you can restore the knobs to be as new. Just make sure you sand down the existing paint first. Then 5-6 light coatings from a 30cm / 1ft distance will do fine.

A proper Akai S-950 User Manual
If there is anyone looking for a user manual for an Akai S-950 that is not missing any of its pages it can be found right here: The reason I mention this is because of the fact that all of the S-950 manuals that are online are for some reason missing / have plenty of blank pages. This one has all of the pages and no blanks! Horray.

Rack electric cable management

Introducing my approach to cable management that banishes the chaos of racks overflowing with tangled wires: a streamlined solution that prioritizes precision and elegance. Disenchanted with the sight of countless cables snaking haphazardly through racks, I embarked on a quest for simplicity and order. The answer? Custom-made cables, meticulously cut to exact lengths, ensuring a clean and organized layout from source to unit. Say farewell to the cable jungle and embrace a tidier, more efficient way to connect your components.

Filled up rack.

Yet only single electric cable on the back. How’s that possible?

Here’s how:

Cut the plywood to correct 1U size.

Drill the holes.

Paint it black.

Install it on the back side of the rack and build the wiring.

Attach EURO connectors and plug them in. Done.


Korg MonoPoly modification – analog white noise

Believe it or not, first revision of Korg Monopoly features a digital noise source with that irritating loop sound. So in response to encountering the digital noise generator with its persistently irritating sound in the initial version of Monopoly, I resolved to craft a genuine analog noise generator as an antidote. My journey commenced with the pursuit of an authentic solution, steering me towards the realm of reverse-biased transistors.

With determination propelling my endeavor forward, I delved into the depths of analog circuitry, seeking to construct a noise generator that harkened back to the essence of the original Monopoly’s design. After thorough research, I unearthed the schematic for the transistor-based White Noise generator utilized in the revised iteration of Monopoly (post the production cycle of May). I didn’t have to search far. The answer was in the service manual of the unit.

You will need a prototype board for this project. They are cheap and forgiving if you make a mistake as opposed to building a custom PCB. This drawing is a showing you how to connect the components, of course you can use any other configuration / layout that you prefer.

Once you build it, it should look something like this:

A good location on where to insert the new board into the unit. I took power from the KLM 356 board (+15V, -15V, GND):

All that was left was to adjust the trimmer to about 4V AC output and connect it to the place of the C11 capacitor on the “old” sound generator. The C11 capacitor must be removed to “disconnect” the old generator from the board. Good bye to annoying digital noise generator that loops all the time. Of course for all the details you will look into the Korg MonoPoly service manual’s schematics which can be found here.

Bill of materials:

  • 3x 10uF/16
  • 1x 1uF/50
  • 1x 4k7 ohm
  • 1x 6k8 ohm
  • 2x 10k ohm
  • 1x 100k ohm
  • 1x 1M ohm
  • 1x 1M ohm trimmer
  • 1x 2N3905 or any other PNP transistor
  • 1x LM4558 or similar
  • all resistors 1/4W

Waldorf Microwave, Pulse etc. infamous Nextel faceplate repair, redo!

Older Waldorf synths used a Nextel paint based coating that gave them a nice satin finish that was soft to the touch. They almost felt like suede. They were gorgeous when new, but unfortunately the coating breaks down over time and gets sticky to the touch. They also smudge when you touch them, while the paint can become flaky and fall off.

So I’ve asked it’s designer Axel Hartmann to tell me what is wrong with the Nextel and here is the full explanation:

“from my experience, the problem is not Nextel. It is much more the proper preparation of the basic coating. If that was done right, Nextel should be staying just as long as any other (wet paint) surface treatment. This does not only mean cleaning the basic part well; – I think, the painter must also use a special / Nextel proven ground finish (which I think, sometimes did not really happen 😉

– that ultra matte, peach like finish did really get me, back in the days -we are talking early 90ies here 😉 I have not investigated into that material in the past years, as we experienced too many problems with it. But like said before; – peeling has mostly to do with improper basic coating / preparation. Still, I found that not a series-production friendly coating, especially with low(er) product run numbers that are typical for MI products / Synthesizers.”

So, actually it wasn’t neither Axel’s or Waldorf’s fault, it just seems that one of the subcontractors messed up a bit at one point. Of course while they were new no one could have known what will happen 20 yrs later down the line. I did found some partial solutions using wipes with oil but since they contain alcohol I decided to stay away from “repairs” and do everything myself.

Paint remover – first steps
Please make sure you are while wearing gloves, eye protection and mask. Paint remover, also known as paint stripper, is a chemical product designed to remove paint and varnish from surfaces. Before applying the paint remover, it’s important to prepare the surface by cleaning it thoroughly. Remove any loose dirt, dust, or debris from the surface using a brush, vacuum, or compressed air. There are different types of paint removers available, such as solvent-based, caustic-based, and biochemical removers. The choice of remover depends on factors such as the type of paint, the material of the surface, and environmental considerations.

Follow the instructions provided by the manufacturer for the specific paint remover you’re using. Typically, you’ll apply the paint remover generously to the painted surface using a brush, roller, or spray. Ensure that the area is well-ventilated and wear appropriate protective gear, such as gloves and goggles, as some paint removers can be harsh chemicals. Allow the paint remover to sit on the surface for the recommended amount of time specified by the manufacturer. This dwell time allows the remover to penetrate the layers of paint and soften or dissolve them, making them easier to remove. After the dwell time, the paint should start to bubble or blister, indicating that it’s ready to be removed. Use a scraper or putty knife to gently scrape away the softened paint. Work in small sections, and be careful not to damage the underlying surface. Here is a picture of the unit 2 hours later as I started peeling the old paint down.

After 15 minutes of work the paint should be gone.

The surface is cleaned and treated to remove any oil, grease, dirt, or rust. This ensures proper adhesion of the powder coating and the panel is ready to be taken to a shop.

Powder coating
Powder coating is a dry finishing process used to apply a decorative and protective finish to surfaces. The powder coating material, typically a finely ground mixture of pigment and resin particles, is electrostatically charged and sprayed onto the surface to be coated. The charged particles adhere to the grounded substrate due to electrostatic attraction. The coated object is then placed in an oven where it undergoes a curing process. During curing, the powder particles melt and fuse together to form a continuous film. This results in a durable and uniform finish. After curing, the coated object is allowed to cool down. Once cooled, it is inspected for any imperfections or defects. Finally, the finished product is packaged and prepared for shipping or further processing.

Powder coating offers several advantages over conventional liquid painting methods, including increased durability, resistance to chipping, scratching, fading, and corrosion, as well as environmental benefits due to reduced volatile organic compound (VOC) emissions.

Before removing the paint I scanned the front panel to be able to build new graphics Drawing vectors over a scanned background in Inkscape involves a few key steps. Here’s a simplified guide. Open Inkscape and import your scanned background image by clicking on “File” > “Import” and selecting your image file. The background will appear on the canvas. It’s good practice to create a new layer for your vector drawings. This keeps them separate from the background image and allows for easier editing. The Pen tool is commonly used for creating vectors in Inkscape. After creating the basic shapes with the Pen tool, you can adjust the curves and lines using the Node tool. Once you’ve outlined the object, you can fill it with color and add a stroke if desired. Here are the screenshots from the process, including tests on how the final product will look like:

Before the final print it is wise to go to the print studio to make a test print to test various colors and their actual output on a printed material:

Printing and the final result
A UV printer is a type of digital printer that uses ultraviolet (UV) light to cure or dry ink as it is printed onto a surface. UV printers are versatile and can be used to print on a wide range of substrates, including paper, plastic, glass, metal, wood, ceramic, and more. They are commonly used for producing signage, labels, packaging, promotional items, personalized products, and decorative materials. The UV curing process offers several advantages, including fast drying times, reduced ink consumption, improved scratch and abrasion resistance, and the ability to print on non-porous surfaces. Here is the finished result:

And before:

Vector file graphics
Here are the vectors for Waldorf Pulse+: These are to be printed on a UV printer. I have included two files since studios prefer to have the edges as well, so they make a test print on transparent acrylic surface with the edges included, then they remove the acrylic and use the other file which contains graphics only.

New LED display is too bright? Here’s the fix!

There are a lot of replacement LED displays on the market, however some of them have not been properly configured for some specific synthesizers. For example the stock JHD732-24064C is way too bright when installed in Yamaha SY-77. Not only it will make its life shorter but the backlight is kind of on the light side of the blue, burning too fast IMO. So if you bought and installed a LED replacement LCD for your Yamaha SY-77, TG-77 or SY-99 and noticed it is a little bit too bright, don’t worry. A simple modification is required and you will be all set. I can not confirm or deny that one of the sellers who sells these replacement displays didn’t paid attention to the voltage that supplies the LED element. The fact of the matter is, the LED backlight is set to be driven by 4.7V which is way too much for a LED based element. A classic LED diode is usually set at 3.3V while displays usually go 3.4-3.5V but should not exceed 3.6V. Else the 50,000 hours rating can not be guaranteed. It will be more like 5,000 hours LOL!

It is easy to recognize these JHD732-24064C replacement displays. They come with just a strip cable and no power cable. The power is taken from the logic power supply line. And the value of the provided current limiter on the board is simply too small. If your display is as bight as this, it’s no good:

A friend offered to sell me his TG-77 and as soon as I saw the display brightness in his offer via FB I knew something isn’t right. Since I was looking for a TG-77 anyway, i bought it, but decided to inspect that display – it was way too bright. Unfortunately there is no schematic for this display board, but some visual inspection showed two jumpers on the PCB board that someone enabled. And as soon as I’ve measured 4.7V over the Anode – Cathode I knew we are onto something.

LED backlight full manual control for JHD732-24064C and Newhaven displays
In this article we will create manual control for the LED backlight. That way we will have a control over the brightness of our screen. This will ensure the long life of our LED backlight and also help relief stress from our eyes during late night hours operating on the synth. If you have a Newhaven display, go to the bottom of this article. And please note, this modification applies only to LED based LCD displays and not to the factory installed EL Foil based LCD’s that came on stock Yamaha SY-77, SY-99 and TG-77.

The two wires labeled FROM LCD and TO LED ANODE are actually the two wires that you solder onto location shown in the picture below.

On the back of the LCD unit you will notice two jumpers. Disable the one on the Anode end. That way current can no longer approach the LED and we can install our own current limiter and a potentiometer. Also inspect that the jumper for the Cathode is enabled (connected). While there, solder a wire to the SMD resistor and another wire to the IC’s pin 8. My advice is to buy 1m of ribbon cable and simply take out 2 wires out which will become our “cable”.

You want to apply a hot glue because we are talking SMD components – they don’t like having wires soldered onto them. This is to release the stress from the wire – SMD component junction/solder point which is the weakest point of this circuit now.

You will measure the length of the new “ribbon” wires until they reach the area near the power supply socket. Because that’s where the potentiometer will go.

Now you will drill a hole there.

And install a 1k logarithmic potentiometer with a 47ohm resistor on one of its connected ends.

Setting the potentiometer in zero value and measuring the Cathode to Anode voltage you should get something in this range. If you are getting 3.4V – 3.5V it is ok, but above 3.5V it is not good, so replace 47ohm resistor with a larger value (use calculator to find correct value) for this current limiter. This completes our modification. Power up the unit and set LED brightness at desired value.

Definitely try the minimal setting! The display will turn beautiful dark blue, and in late night hours you eyes will thank you. Plus it will live much longer. As a matter of fact i always have my TG-77 at this setting. It is still bright enough.

Semi manual backlight control?
Of course it is possible. If you don’t want to mess with a potentiometer but just want a fast switch for reduced backlight operation to extend the life of your backlight i.e. when you don’t work on the unit directly but use it remotely and thus not need the display. If that is the case then you can simply add a switch to the back. At the switch Position 1 you will put a 47 ohm resistor and at the switch Position 2 you will put a 1k ohm resistor. These are the values that I have found to give a good response. The rest of the modification is just the same and in the the above schematic just replace the potentiometer with a switch.

Newhaven display modification
Essentially you will do everything as described in here: Korg Wavestation A/D LCD Upgrade
Except that you replace the current limiting resistor (100 ohm) in that article with a wire that goes to the potentiometer with a 47 ohm connected onto. From potentiometer you go to the Red wire of the Newhaven display which goes to the Anode and you are set. The hole and the potentiometer go exactly as shown above.

Boss DE-200 gain mod for keyboards

To anyone who wanted to use DE-200 with synths, he most likely encountered two major problems which relate to signal levels and input impedance. Those are: output of the effect unit is very low in the volume while the input would overload at slightest increase of the input knob. The unit can be quite difficult (and frustrating) to operated under these conditions!

Here is why. This effect expects a low signal source such as the guitar at its input. Further more, it expects a guitar amplifier at the output. In typical synth based studio the setup is reversed actually. Because we want a unit to accept the full level signal of the synth (synth set at max volume) to achieve maximal s/n ratio. We also want effect unit to add a little or no amplification at the output (also known as Unity Gain). If you connect a synth with max output gain to the DE-200, you’ll soon overload the input stage, while output will be very silent forcing you to add a lot of gain at the console. This mod solves all these issues and converts DE-200 into a keyboard signal-level accepting unit with approximately -10dB level signal at the output (typical for average synth).

Gain Modification
First thing we want to do is to reduce the input impedance and gain. This will reduce the noise as well. Here is the schematic which shows a R10 1Mohm resistor which should be replaced with 10k metalfilm resistor. A better look at the schematic reveals a high pass RC filter made with R10 and C3. To retain the properties of the input filter we need to replace the C3 as well. A value of 4.7uF is adequate. This so far effectively reduces input gain stage from the current +12dB to around +6dB of gain at 10kHz. We will further reduce it across the op amp by changing R2 to 4.7k. At that point we should have input stage set at around +3dB gain which is just fine.

Important: Since you will most likely use the electrolytic cap you need to pay attention to polarity of the cap. Polarity must follow the signal, and as we all know the signal goes from the input jack to the op amp thus you should orient positive side of capacitor to the R1 and negative to the op amp. This finishes the input buffer and impedance modification.

Now we come to the Roland’s pre-emphasis part of the machine built around IC 2b where additional gain is built. Pre emphasis is basically a high pass filter with gain defined by R6, R7 and R8 resistors and corner frequency defined by C5. Our goal is to have near 0 dB of gain of the filtered signal and amplify only the non filtered part. We will achieve that by replacing resistor R8 with 110k ohm value. This effectively reduces gain from 19dB to around 13dB at 10kHz. Or globally, we reduced gain of the input stage for a total of 15dB.

Now it’s time to take care of the mixer / de-emphasis stage. Default gain is a little bit weak for average console, since it is expected a guitar player will us an amp, rather than line mixer. We will increase the gain replacing R159 with 47k and R158 with 220k ohms. It is necessary to replace C109 as well, since we’re working on a low pass filter in here made of R158, R159 and C109 (ignore the C110, it’s purpose is something different and unchanged value isn’t that critical unless you live right across FM transmitter). Same applies to C7. Both can be left at existing values.

This completes our modifications. Try DE-200 without the mod and then with the mod. Totally different units! More dynamic range, much much less noise. If you want more, please continue to read.

Crazy Ivan modification (addendum for experienced)
While calculating component values for the de-emphasis part i came with one idea to try. How would machine sound if we would have the signal partially pre-emphasized and passed thru. The rest was easy. In fact this is uber-simple yet very effective modification for all of you experimenters and le sonique bizarre aficionados. The best part is that it includes one of the components that you already need to replace during the previous gain modification.

In the image above, points A and B mark the place where existing C109 capacitor is, which you need to replace with 1nF as detailed earlier. However, while replacing it, you can solder just one of its legs, then use a wire and lead it to the switch, which can be placed on the front panel. From the other side of the PCB, in the same place you can solder another capacitor, this time 100pF value and again lead it with a wire to the switch. Then you add another wire, from the center point of the switch and lead it to point B.

There’s enough space on the front panel to do the drilling right next to the Power on/off switch. Just measure everything carefully. I strongly advise you to start drilling from the front side, with the plastic panel on. Once you get the small hole done, you can remove the plastic, and use correct width for metal drill so that switch perfectly fits in. The goal of the hole in the plastic is to make it as small as possible yet enough space to move the switch up/down. Please do not place nut on the front plastic. You will make the unit look ugly for no reason. Because there’s a metal plate right below the plastic where a nut can be placed and tightened.

In the image above you can see the Crazy Ivan ON/OFF switch added right next to the POWER switch. The goal of such modification is to make it non destructive (a click of a switch resets the unit to original sound) and of course to maintain the sleek design of the unit.

What this modification does is that it shifts the de-emphasis curve (low pass) from the 600Hz up to the 6kHz. Result are really exotic and make the unit cut through the mix like a razor. Use carefully though, since harsh delays can irritate the listener after a while. Flanges and Choruses with Crazy Ivan are interesting as well and give some extra edge to the sound!! Unit can be used for processing drum beats as well (feedback set to 0) particularly because the signal path contains some exotic elements like a compression, dynamic expansion and the emphasis. The work detailed in this article can be performed under one hour.

Boss CE-300 gain mod for keyboards

This mod lets you use CE-300 for synths as the original design is for guitar impedance / levels and connecting a keyboard will easily overdrive the circuit.

This unit represents the classic Roland chorus sound in some way similar to Dimension series, yet little bit different in modulation. It is still true analog based design with BBDs. The problem however is the signal level and input impedance. This effect expects a low signal source (guitar) at the input and an amp (gain unit) at the output. In typical synth studio the setup is however a little bit different, or somehow reverse so to say. Because we want a unit to accept the full level signal of the synth (to maximize s/n ratio) and then add little or no amplification at the output (unity gain). If you connect a synth with max output gain to the CE-300, you’ll soon overload the input stage, while output will be quite silent forcing you to add gain (and noise) at the console. This mod solves both of these issues and converts CE-300 into a keyboard signal-level unit.

First thing we want to do is to reduce the input impedance. This will reduce the noise as well. Here is the schematic which shows a R2 1Mohm resistor defining the input impedance for the op amp. Replace it with 10k metalfilm resistor. A better look at the schematic reveals a high pass RC filter made with R2 and C1. To retain the properties of the input filter we need to replace the C1 as well. A value of 4.7uF is adequate. Since you will most likely use the electrolytic cap you need to pay attention to polarity of the cap. Since the signal goes from input to the op amp you should orient positive side of capacitor to the input and negative to the op amp. This finishes the impedance modification.

Next thing is the input gain which is defined by the R3 resistor across the op amp (and the input resistor R4) that takes signal into the op amp. On default unit, this gain is too strong, thus you’ll too easy overload the unit and lose precious dynamic range. By reducing R4 to 10k we are letting more voltage / less noise hit the op amp. And by reducing the R3, we are reducing the gain. Replace both resistors with metalfilm 10k.

Now it’s time to take care of the output stage. Default gain is little bit weak for average console. We will increase the gain by increasing values of resistors R105 and R106 from 47k to 200k. Use metalfilm resistors. Replace C72 and C73 (both 47p) with a 10pF. They are located in parallel to R105 and R106 (which you will be replacing, so it’s easy to locate them). The reason for this is to preserve the output frequency response curve as original. Without this the unit might sound a bit darker (at 10kHz you might already be loosing 1dB, reaching -3dB loss at around 18kHz).

This completes our modifications. Try CE-300 without the mod and then with the mod. Totally different unit. More dynamic range, way less noise. Here are the pictures that should help you locate points where these components are located. C1, R2, R3, R4:

and R105, R106:

SCSI to CF “Blackbox” interface for samplers


If you own a lot of samplers, Blackbox is a more economical solution than filling each sampler with its own SCSI to CF (or SCSI2SD) interface. Because if you got 5 samplers, you end up in a pretty decent sum of money with minus in front of it. These interfaces don’t go for peanuts, so thinking economical comes as a useful option. This is why i went for the Blackbox solution, which is a single drive, that connects to every sampler via a standard SCSI cable (assuming the sampler has a SCSI interface). I will also assume you’ve read basic SCSI guidelines and you know each end of the SCSI chain must be terminated. If the Blackbox interface will be in between two samplers, then obviously don’t have to terminate it. I should also point out that the interface can normally be used by two samplers at the same time, with only exception that no two samplers are trying to read or write to the disk, since you will end up with corrupted data. If you need to swap the card (i.e. one sampler uses one card, the other uses another) simply power the Blackbox down, replace the card, power it up, wait 20 seconds and access the card from the second sampler. It is as simple as that.

And it is true, old samplers that feature SCSI ports can run on modern CF or SD cards. Full list of the verified / tested samplers is below. All it takes is a SCSI to IDE bridge. At one point these were relatively cheap, nowadays the price has gone up, but if you happen to have 7720U in this article i will describe how to build the ultimate SCSI blackbox for your old sampler. I should also point out that the popular alternative is a SCSI2SD which is very similar thing. In fact what i describe for the CF blackbox applies very similar to the SCSI2SD and you can build a blackbox with either of these two. In here i will describe building SCSI to CF. If you have any extra questions, please check here first and see if those have already been answered.

This interface was built using:

  • an old SCSI external case
  • ACARD 7720U bridge
  • IDE compact flash reader with 3.5″ bay
  • IDE ribbon cable
  • 3.5″ -> 5.25″ drive bay converter
  • two MOLEX connectors and some wires
  • car paint, black matte, in three coatings
  • noiseless 40mm fan
  • plexi to build a bracket between PSU and “drive” bay area for better airflow

I have tested it and works perfectly with following samplers:

  • Akai S1100
  • Akai S3000XL
  • Emulator EIIIXP
  • Emulator E4XT
  • Roland S-770
  • Ensoniq ASR-10 (power up the blackbox 20 sec in advance!)
  • Roland S-550 (System CD-5 V.1.02 required, power up the Roland, let OS loads, power up blackbox, wait 20 sec to boot, ignore “Parking”, go to Disk setup, left click “Hard Disk ID=” then right click it and the drive will show up)

So how does it work? Exactly like a normal SCSI drive, ie a HD or a CD ROM. In fact you can take all of your CD-ROM’s, save them as images and then write the one you need onto the CF card, which in this case would be a 1 GB Compact Flash card. After learning a lesson with damaged Akai CD ROM i backed up all my CD’s on the hard drive in the .img format. I still own the original disks, so this makes it fully legal. Using a software called DiskImage i can now transfer the content of a CD ROM into a (1GB) CF card. Tried it and works perfect. That way no more need for the external SCSI CD ROM connected to my sampler, which was just taking space, producing a lot of noise, and was much slower than the CF card.

A small tip for those who use Akai samplers. On the Blackbox back set the ID to 5. Because Akai’s default SCSI drive is 5. That way, each time you turn on the Akai, you won’t need to go to settings and change ID. Just press F7 (SKIP) and you’re in. And another tip, if you want to format a card with Akai sampler use the arrange command for the formatting work. It appears that “Format” command was for something else.

You can also build your own custom sample compilations and your own CD-ROMs (so to speak). CF card once formatted and filled with data on a sampler can be read from any computer with CF card reader (and appropriate software) as a disk image. It can then be stored on the computer (and ZIP-ed to reduce file size). This way you don’t need dozen CF cards. Two or three are more than enough. Simply archive your project to computer once you’ve finished it, format the card in your sampler and you’re set for the new project. Once the next project on the sampler is completed, again archive it to computer and start again. If you’re on a Windows machine best program to read and write images is Roadkill’s DiskImage. However be VERY CAREFUL with this program since it has access to all of your disks in the computer, and if you select a wrong drive as a destination to write an image, you might end up with deleted hard drive! Always double check the destination and the drive letter into which you have a card reader assigned. If you’re on a Mac then you don’t need any extra software. Just got to Utilities and use Disk Utility. From there you can read CF and SD cards and save them as images, or write ISO images onto same cards. I suggest you do some google on how to read / write ISO images on a Mac, using Disk Utility.

All connected together looks something like this:


Closer look at the IDE CF card reader, IDE cable and power supply Molex. Jumpers are set to Master:


Closer look at ACARD 7720U. Right side: power supplying the bridge, left side LED wire for the IDE drive. LED wire is not 100% necessary but highly useful, since it provides indication when the drive is active. Without it you won’t know. Jumpers are set to ID5 however please ignore it, since i don’t use any jumpers in this setup. This SCSI case has a SCSI ID selector on the back which you just connect to these three jumpers. That way you no longer have to open the box in case ID change is required:


Here you can see the 3.5->5.25 bay converter and power supply from the unit connected to CF drive from where it is divided into two lines. One stays at CF the other goes to ACARD bridge:


View from the back:


Finished product:


Moog Voyager O.S. Mods (PWM, filter poles)


After hearing Moog filter in two pole configuration, while working on one Voyager Performer, i KNEW i need to get this filter configuration possible on my Old School edition. It simply sounds stunning, particularly when using the filter as an instrument on its own. Not only the modification gives you 2 pole edition, but it also provides 1 pole and 3 pole, along with default 4 pole. Essentially it is a switch that gives you selection of a: 1 pole, 2 pole, 3 pole and 4 pole filter configuration.

And then there was another feature that i always wanted to have on Voyager, which doesn’t exist in neither Performer and Old School edition and that is PWM disable switch. Some may ask, why would you want to disable the PWM? The answer is: sub oscillator! (emulation of sub osc, to be precise). Ever wanted SH-101 style bassline out of your Voyager. You can now! Finally and at last! What this mod does is it prevents one VCO to receive PWM. Since the modification goes via switch, it means that at any point you can return the unit to original, non modified state!

One thing i beg you, PLEASE DO NOT drill the front panel on the unit when doing these mods! I have seen someone doing that to CS-15 while performing my CS-15 mods and the result was horrid ruin of something that was once a beautiful synth. I have specified exact location on the back where you can can install both of the switches, so please, place them there! The manual includes diagrams and pictures. Even if you are a novice level, you can do it – assuming you know how to desolder ICs (two ICs need to be desoldered, and IC sockets have to be placed there).

Modification manual is available here: Voyager OS Mods.pdf (right click, Save as…)

Roland LCD-1 sample CD finally available on S-550!


And it’s not a joke. I’ve found a way! For those who don’t know LCD-1 is a sample CD ROM that was built for Roland S-550 sampler. In order to use it, the sampler had to be expanded with SCSI interface card and connected to Roland CD-5 external CD ROM device. This is the only CD drive that will actually work with S-550. And there are, like 10 of those working in the world if not less. Which is probably the reason there is none information about this topic on the internet: 1)SCSI interface for S-550 is rare 2)CD-5 drive is ultra rare (Mr. Varaldo managed to found only two units in 10 years time span on eBay. Of course none of which work, which is expected after all this time).

After three days of trying various methods, some “hacking” and what a not i finally found the way to have the content of the legendary Roland CD ROM in S-550. Obviously S-550 SCSI interface is required, however no need for the impossible-to-find Roland CD-5 drive!!!! This is crucial part!

I should point out that there are some minor limitations, but it’s not that bad. Essentially you can only have 64 volumes at a time (per CF card) since that’s max Roland can make per hard drive. Now, because there are 165 volumes on the LCD-1 sample CD ROM, that means you will need three CF cards to store its content. But then, better something than nothing! In fact, you can search the whole WEB i doubt you will find anyone managed to find a better solution (or any solution at all for this sample CD ROM). Originally i got this CD from Roginator and i’ve decided to keep it since i knew i will need it one day when i get the SCSI interface for S-550 which i just did few days ago.


Let the hacking begin! First thing you need is the SCSI-CF blackbox which is detailed here. Or any sort of SCSI-CF drive that you can get hold of and that you know that works with samplers. And obviously your S-550 need to have the SCSI interface. I’ve purchased mine couple of weeks ago on eBay. These are rare, but keep in mind that some S-550 units with pre installed SCSI card can pop up from time to time. One important thing to say – you need the latest OS for the S-550 which is called “CD-5 System Ver. 1.02.”. This method will not work with system floppy that comes with SCSI interface because it is lacking some required features for this task! And you will need Chickensys Translator software.

This is one of those, don’t ask why, just DO IT as i say: Start the S-550 up, with the system floppy, let OS load, then power up your SCSI-CF interface, wait 20 sec to boot, go to Disk / Setup / COM / SCSI Check. Your drive should now get scanned and shown with ID. You might get notification “Parked”. Important: Always ignore “Parked” label and simply left click “Hard Disk ID=” then right click it and the drive will show up. If you go HD Restart, the machine or drive might hang out! Now go and format it. Job done. Power the unit down. Insert the CF card into your computer (assuming you have a CF card reader). Insert the LCD-1 CD ROM into your CD drive. Start the Translator. Select 64 volumes on the CD ROM and drag them onto the CF card (which should appear under SCSI / proprietary format drives in Translator). And that’s it!

Go back to S-550. Similar starting procedure, although once the CF card is formatted, you can turn on SCSI-CF reader 20 sec before you turn on S-550. So no need for SCSI Check in menus, simply go with the left/right click on the “Hard Disk ID=”. This is what “wakes up” the CF card, not “HD restart”, since this isn’t a HD at the first place. And there you have it. 64 volumes from the LCD-1 CD ROM. Repeat the same for cards 2 and 3 and you have the complete LCD-1 library at a press of a few buttons.


I was always curious about the content of the Roland LCD-1 CD ROM. So i did some digging through it and at address 200 (hex) i found what appears to be the top level directory. After some cleaning and reformatting of the above data, here is the complete content of LCD-1. Each of these represents one “volume” which is equivalent of one floppy. It will be very interesting to compare this list to the factory floppy disks which are available from sgroup archives (for both the S-550 and S-50).

 Acoustic Piano #1 USB501#4
 Acoustic Piano #2 RSB5501#3
 Grand Piano #3 Lower RSB5501#9
 Grand Piano #3 Upper RSB5501#10
 Pop Piano #1 USB505#7
 Pop Piano #2 RSB5501#8
 S-50 System #1 Piano S-50 sys #1
 S-330 System #1 Piano S-330sys #1
 Electric Piano #1 L501#1
 Electric Piano #2 & Clavi L501#2
 Electric Piano #2 USB502#4
 E.Piano & Vibraphone JPL552#8
 E.Piano & Vibraphone USRSB5503#7
 Pluck Pianos RSB5502#4
 Midi E.Piano & Synth Bass USB504#3
 Club E.Piano / Bass & Wave Synth RSB5501#1
 Electric Organ #1 L501#4
 Pipe Organ & Choir JPL552#7
 Pipe Organ & Choir USRSB5503#6
 Pipe Organ & Harpsichord L501#3
 Synthesizer #1 JPL501#5
 Synthesizer #1 USUSB502#6
 Synthesizer #2 USB504#8
 Synthesizer #3 USB505#2
 Midi Stack #1 USB502#7
 FM Stuff USB505#9
 FM Filtered RSB5502#8
 Harpsichord JPL552#6
 Harpsichord USRSB5504#9
 Celesta JPL552#9
 Celesta USRSB5503#10
 Strings #1 L504#20
 Full Strings Section JPL551#1
 Full Strings Section USRSB5503#9
 20th Century Orchestra USB502#3
 High Strings USB502#9
 Suspense Orchestra USB504#7
 S-50 System #2 Strings & Choir S-50 sys #2
 Cello (Solo) L551#2
 Cello (Solo) & Duo RSB5501#6
 Electric Bass #1 JPL504#18
 Electric Bass #1 USRSB5501#4
 Electric Bass #2 USB501#6
 G&L Precision Bass RSB5504#2
 WoodBass #1 & Harp #1,2 L504#19
 Mini Moog Bass 1 RSB5502#5
 Electric Guitar #1 JPL504#17
 Electric Guitar #1 USUSB503#10
 Guitar & Stick USB504#4
 Funk Guitar USB504#10
 Acoustic Guitar #1 JPL504#16
 Acoustic Guitar #1 USUSB502#2
 Acoustic Guitar #2 RSB5501#5
 Guitar, Mandolin, Harp USB502#1
 Koto #1 JPL505#21
 Koto #1 USUSB506#4
 Gagaku #2 L506#30
 Shamisen #1 (Hosozao, Tsugaru) L505#22
 Shamisen #2 (Futozao, Chuuzao) L505#23
 Shamisen #1 USUSB505#4
 Biwa #1 L506#28
 Indian Strings #1 L507#31
 Indian Strings #2 L507#32
 Western Strings #1 L508#36
 Saxophone #1 L502#7
 Saxophone #2 L502#8
 Tenor & Soprano Saxophone #1 RSB5501#2
 Alto Saxophone JPL551#4
 Alto Saxophone USRSB5503#8
 Trumpet, Trombone & Horn L502#6
 Trumpet & Trombone JPL551#5
 Trumpet & Trombone USRSB5503#5
 Multi Sampled Solo Trumpet &Horn USB501#7
 Mute Trumpet USB506#2
 Bright Solo Trumpets & Section RSB5503#2
 Brass Section #1 USB501#2
 Woodwinds #1 JPL502#9
 Woodwinds #1 USRSB5501#7
 Flute & Piccolo JPL551#3
 Flute & Piccolo USRSB5503#1
 Woodwind #2 L502#10
 Shakuhachi #1 L505#24
 Yokobue #1 L505#25
 Gagaku #1 L506#29
 Japanese Flutes #1 USB506#8
 Western Wind #1 L508#37
 Indian Wind & Thai Gong L508#39
 Andean Wind #1 L509#42
 Pan Pipes / Shakuhachi USB504#9
 Assorted Kicks #1 USB503#7
 Assorted Snares #1 USB503#8
 Assorted Cymbals #1 USB503#3
 Assorted Ride Cymbals #1 USB503#2
 Lg.Crash / Splash in Stereo RSB5502#3
 Long Ride & Bell in Stereo RSB5504#1
 Oriental Cymbals & Gongs RSB5504#10
 GT Drums USB501#3
 New Drums & 808 USB501#9
 Gate City #1 USB502#5
 Analog Percussions USB505#6
 Reverb Drums USB506#9
 EQ Drum Kits 1A RSB5502#6
 Stereo Drum Kit #1 RSB5502#9
 Stereo Drum Kit #2 RSB5502#10
 S-50 System #4 Effect Drum Parts S-50 sys #4
 Latin Percussion #1 L503#11
 Latin Percussion #2 L509#44
 Latin Percussion #3 L509#45
 Timbales & Toms USB503#6
 Conga Monga in Stereo RSB5502#1
 Exotic Hand Percussion #1 RSB5504#3
 Mark Trees & Wind Chimes RSB5504#8
 Indian Percussion #1 L507#33
 Indian Percussion #2 L507#34
 Indian Percussion #3 L507#35
 Middle Eastern Percussion L508#38
 Gamelan #1 L508#40
 Eastern Flavour #1 L509#41
 African Percussion #1 L509#43
 Ethnic-Anklungs & Wind USB501#10
 Ethnic & Gongs #1 USB503#9
 Taiko #1 L506#26
 Taiko #2 L506#27
 Japanese Percussion #1 USB505#3
 Classic Percussions L552#10
 Orchestral Percussion #1 USB504#2
 Orchestral Percussion #2 RSB5503#4
 Orchestral Percussion #3 RSB5504#7
 Oooze & Oz USB501#1
 Air Disk #1 USB503#5
 Air Disk #2 USB505#1
 Mallet Vol. 1 L503#12
 Techno Set USB504#6
 Surf's Up!(use a tone of reverb) USB505#8
 Funk Rock Set 1(Composer Series) USB506#7
 Rock Composers Tool RSB5503#3
 Designer Trap Set #1 RSB5504#4
 S-50 System #3 Brass,Bass,Marmba S-50 sys #3
 S-50 ver.2.0 Multi Patch S-50 ver.2.0
 S-550 System #1 Multi Patch #1 S-550sys #1
 S-550 System #2 Multi Patch #2 S-550sys #2
 S-330 System #1 Multi Patch S-330sys #2
 New Strings, E.Piano, Choir USB501#5
 Classical (Composer Series) USB505#10
 Big Time USB506#1
 Ethnic Soundtruck USB502#8
 Ethnic Grab Bag USB504#5
 Orchestra #1 L503#13
 Effects #1 L503#14
 Stereo Effects #1 L503#15
 Train, Ride, Air Base USB501#8
 Anarchy Time! USB502#10
 Metal & Glass FX USB503#1
 Jungle Life USB503#4
 Wild Kingdom USB504#1
 Space FX #1 USB505#5
 Drivin' USB506#3
 Comedy Props USB506#5
 Hollywood FX USB506#6
 Rainy Day USB506#10
 Jet Strings in Stereo RSB5502#2
 Stereo THX RSB5502#7
 Industrial Strength Techno Perc. RSB5504#5
 Sports Disk RSB5504#6
 S-50 System #5 Sound Effect S-50 sys #5

Roland LCD-1 contains 165 volumes. Roland Floppy library contains 139 volumes. The interesting thing is that 38 floppies that exist in Roland Floppy library aren’t present on LCD-1. These are as following:

 Synthesizer RSB501#2
 Harpsichord and Pipe Organ RSB501#5
 Woodwinds RSB501#8
 Baritone/Alto Saxophone RSB501#9
 Harp No. 1 RSB502#2
 String Variation No. 1 RSB502#5
 Balaphones RSB503#3
 Piano (New) RSB503#4
 Bass #2 RSB503#6
 Tone Wheel Action RSB503#7
 Woud & SAZ RSB504#4
 Tamboura & Sitar 1 RSB504#5
 Orchestra & Tympani # 1 RSB504#7
 Hi Strings 2 RSB504#9
 Tabla Collection 1 RSB505#9
 Saxaphones Vol. 1 RSB5501#2
 Bass Vol. 1 RSB5501#4
 Sollo Cello & Duo RSB5501#6
 16 Bit Piano 1A RSB5505#1
 16 Bit Piano 1B RSB5505#2
 1959 Strat Oddities RSB5505#3
 Grand Piano 3W RSB5505#4
 Martin Steel String RSB5505#5
 Rock Guitar Sampled RSB5505#6
 ET Perc 1 RSB5505#7
 Real Bass Slapper RSB5505#8
 Sun. Nite Sax, Solo Alto RSB5505#9
 Kingdom Pad RSB5505#10
 Pluck Guitar RSB5506#1
 Underwater FX RSB5506#2
 Cho Trumpet RSB5506#3
 Moog Brass 1 RSB5506#4
 CD Basses RSB5506#5
 Violins x 8 RSB5506#6
 Hits & Hornz RSB5506#7
 Brass Vol. 3 RSB5506#8
 Super Kit 16 RSB5506#9
 Banjo Man RSB5506#10

So if you add these to the volumes list above, you now have the COMPLETE Roland S-550 library listing!


Bam! And here they are. I loaded all 38 floppies (mentioned in post above) one by one using a PC floppy drive and Omni Flop then into the S-550. Then saved them on the S-550’s “hard disk” which is actually a CF card in my SCSI interface.

I’ve then put CF card back into the computer and ripped it as an image. Now it’s easy to compile it using Translator (since it reads .img files). I plan to put complete library (all 203 volumes) on four 128MB CF cards, since Roland’s hard disk has 80MB size limit and 64 volumes. And… that’s about it! Complete Roland S-550 library, finally available on my S-550. Can’t believe it’s done!!!

How to upload SYX files to synthesizer via MIDI or USB

Synths with MIDI interface
I will describe the procedure for the PC, though exactly the same process applies for other computers such as Mac, Linux etc. In here i will detail the use of free MIDI/SYX utility program called MIDI-OX which is used among many other things, for system exclusive (aka Sys-Ex aka SYX) transfers. First step is to obtain a MIDI cable that can be connected to your soundcard MIDI jack. If your soundcard does not provide MIDI jacks then you will have to buy a MIDI-USB interface. This will let you connect MIDI cables to your computer via USB and transfer SYX files. The cable that you purchase (ie via eBay) should look something like this:

USB Midi Cable

After you’ve purchased the cable and installed its drivers (important!) then you can proceed with the next step. And that is to download MIDI-OX software which is available here. If you’re on a Mac there also freeware available for MIDI transfers and is called Sysex Librarian by Snoize, available here. There is also a cool guide for using Snoize available here.

Ok back to PC. Once you start the MIDI-OX first step to do is to look for MIDI Devices in one of the menus and make sure that your USB cable device is selected there under MIDI Outputs. Although the picture below shows selection of MIDI Inputs, please make sure you have MIDI outputs selected with the USB cable and its name. If you have a soundcard that already features MIDI jacks, then in the below dialogue you will select your soundacrd’s MIDI outputs, and that’s it.


Once you have the device selected, close the dialogue. At this point connect MIDI Out jack from your USB MIDI Out cable (or your soundcard’s MIDI Out jack) and connect it to MIDI In jack on the back of your instrument (ie Roland JD-990). Now go back to main window and select the following command:


The program will ask you for the location of the .syx file. It is the same file that you have purchased, so you will guide program to its location on the hard disk and the transfer will begin. Wait for around 1 minute for transfer to finish, and you’re done. In case device does not receive patches, you will have to read user manual of the particular synthesizer and look for System Exclusive Enable/Disable setting. Most synths have this enabled by default, however, if any problem occurs at this stage it is most likely because System Exclusive on the synth was disabled. For example on Roland JV/1080 this setting is under system as Rx.Exc ON/OFF. Another problem might be wrong Device ID. On most units i can think of, it should be set to Unit#17 or Device ID=17. This is also set under System or MIDI settings within the synth. If you still have problems, try resetting the unit to factory defaults.


Synths with USB interface (i.e Waldorf Blofeld)
If you own a synth with USB interface then you don’t have to purchase USB-MIDI cable. Just download latest USB driver from the synth’s manufacturer’s web site and install it into your computer following the provided instructions. Once you do that, and the USB MIDI drivers are installed, connect the synthesizer to your computer via USB cable. Now start your MIDI/SYX utility of choice (MIDI OX or Snoize) and select the MIDI Outputs, which in this case should be USB Audio Device <synth name>.


One important thing for Waldorf Blofeld users: make sure the buffers in the MIDI transfer program are set to number larger than 1024 bytes. For example in MIDI OX program go to View menu. Inside you will see a command called Sysex… Click on it and the Sysex view window will open. In its header you will find Sysex menu. Inside that menu you will find a command that says Configure… Image above shows the window that should be open. Click on it and set Low Level Input and Output Buffers Size to 1048 bytes. Leave other values at default. Click Ok. Close the Sysex window as well. The rest of the procedure (transfer) is exactly the same as was described earlier and that is: Actions / Send / Sysex File…

Video Guide
If you are unsure about installation here is an example session from Windows OS on how to transfer patches onto your synthesizer.

As of the MIDI-USB cable interface, while it is cool thing for transferring system exclusive messages i do not recommend using it for MIDI sequencing. Please google for: midi jitter usb. 🙂

The ultimate Roland JV, JD, XV F.A.Q.


Super JV vs XV series
Following the JV/XP series were Roland’s XV series: 5080, 5050 and 3080. XV-5080 is mixed content 32kHz and 44.1 kHz. I got this later confirmed by Roland. (though some web pages list it as 32kHz ROM only, but this is not true). I will focus now on XV-5050 and compare it with JV-1080. Some users started complaining about the XV-5050 sounding a bit “thin”. There is some truth in that but what i can tell in reply is that 5050 sounds more hi-fi. Because of 44.1k sample content, some energy has been “lost” due to wider frequency coverage. Patches played on 1080 and 5050 side by side will sound different. This is a fact that i’ve verified myself. 5050 is more hi-fi and has that extra sheen while 1080 is more darker and is a bit more mix friendly when it comes to frequency and EQ. You will find some waveforms more hi fi sounding in XV when compared to Super JV series.

It should be worth mentioning that 5050 has some sort of permanent low shelf filter at about 30 Hz, so you’ll definitely get a less bass energy. But the high freq response is just spectacular if compared to something like a JV-1080. Especially when you start using the digital output and route it directly into DAW, it’s a no match in crystal clear sound. FAQ UPDATE according to Joe (from comments below) the 5080 seems to have the same low shelf filter going on like 5050 and they seem to sound identical. This is what i always suspected, however since 5080 can set its clock to 48 kHz when loading S series samples we can’t say they sound 100% identical, simply because 5080 can produce more high freq content in ‘S-760 mode’.

One thing that is very different on 5050 vs 1080 is the dynamics. For some reason it seems that 5050 has some sort of compressor at its output. As a result, some of the patches have less dynamics going on in them. This is most obvious on layered sounds that have a lot of phasing between oscillators going on. While the same patch on 1080 will produce more differences in volume, on 5050 it is more constant. This can be good or bad, depending what kind of sound you need. For movie/TV scores you would probably want more dynamics going on, hence the 1080. And for dance music, you would go 5050 since it delivers that straight – in your face sound – right out of the box, without need to work on dynamics. For the above reasons 1080 definitely sounds more soft and gentle.

As of XV-5080, i tested it side by side against XP-30 on the same patches and the difference was quite noticeable in what appears to be a far greater stereo field and definitely superior sonic quality of 5080 effects. I particularly remember one preset called Letter From Pat. In fact if you have both units, just load it and hear the difference for yourself. It’s day night difference in favor of 5080.


JD-990 vs. XV series
XV series contain the whole JD-990 waveform set. With XV-3080 being 32k and XV-5080 and XV-5050 with original 44.1kHz JD set. Some of the waveforms have been renamed, but they are there. It should be said that on along the Adaptive DPCM waveform compression, I always suspected (but never got it 100% confirmed) XV series have extra  (destructive?) form of compression on top, similar to mp3 and it can be spotted visually with most simple analyzer. There is no such compression on JD series. More on that in one of the chapters below.

Patch conversion JD into XV is directly not possible. However it would be possible to convert (manually) a patch from JD-990 into 5050 since Roland implemented the whole “Effects Block A” section from JD into 5050 (available as EFX called JD Mlt). Block B can be emulated with Chorus/Delay and Reverb. There is a whole article on this subject available on this website. Only difference is the filter cutoff numeration system. On JD-990 it goes from 0 to 99 while on 5050 it is 0 to 127.

There were some rumors on various forums that XV-5080 is 32kHz (thus being able to play only up to 16kHz). This however is simply not true. We will now take a look at a waveform spectra of a White Noise sample as played from JD-990 and XV-5080. What we can clearly see is that not only they are identical but they both go all the way up to 22kHz, which clearly indicates 44.1k playback.

02 01

Benefits of XV over JD is that the filter on XV has a greater dynamic range. There is no clipping issue on XV as opposed to JD when you set filter keytracking to 100%, find a resonant spot, press a chord and end up in harsh digital distortion (if resonance is above 40). Not only XV won’t distort, but even if it happens on some waveforms, there is one additional parameter called oscillator Gain that lets you reduce the volume of the waveform prior to being fed into filter. You can set it to 0dB or even -6dB. On JD it appears to be permanently set to +6dB (of XV equivalent) which is a pity. That’s the only feature i can’t regret not having on JD. Of course one thing that is very known is that there is definitely a difference in the high end of the filter. JD-990 will go a little bit higher in frequency and thus add more sweetness. The rest of the frequency range response is almost identical.


The Sound
There has been a lot of talk about difference in sound within units that should be based on the same engine. We will here list the converters used which might indicate why some minor sonic differences. There’s an old rumor that the film guys prefer the sound of 1080 against newer the XV series such as 5050. This is a bit complex matter since it involved dynamics and not just frequency, and i have explained it in a chapter above. Let’s now take a look at converters of JV and JD units (notice: XP is a JV with a keyboard)

JV-80   32k  sample rate DAC: 18-bit PCM69P
JV-90   32k  sample rate DAC: 18-bit PCM69AU-1
JV-880  32k  sample rate DAC: 18-bit PCM69AP (main out)*
JV-1080 32k  sample rate DAC: 18-bit UPD63200GS-E2
JV-2080 32k  sample rate DAC: 18-bit PCM69AU
XP-30   32k  sample rate DAC: 24-bit AK4324
XP-50   32k  sample rate DAC: 18-bit UPD63200GS-E2
XP-60   32k  sample rate DAC: 18-bit PCM69AU
XP-80   32k  sample rate DAC: 18-bit PCM69AU
JD-800  44k1 sample rate DAC: 18-bit PCM61
JD-990  44k1 sample rate DAC: 18-bit PCM61P
* uses UPD6376GS-E2 for sub out
  • JV/XP uses Adaptive DPCM, plus something that looks like a destructive form of wave compression (mp3 style)
  • JD uses Adaptive DPCM and no destructive compression (no data holes)

Some people claim they can hear the difference of JV-1080 vs. JV-2080. Unfortunately i don’t have them side by side to verify this, but if someone can, simply load the same patch, record it and send it to me or on the Gearslutz forum and we will inspect it. The rumor is that 1080 sounds “better”, whatever that means. Only thing i can confirm is that converters on the JD-990 sound way better (more stereo width) than those on JV-1080. In fact, it’s probably the best sounding synthesizer that Roland ever designed. Hearing is believing and you should really give it a try if you didn’t by now. There’s a reason why JD-990 scores for much more than 2080, although from technical standpoint, 2080 offers much more waveforms and has better mod matrix.

Some quick points: Over the years i’ve had following machines JD-990, JV-1000, JV-1080, XP-50, XP-30, XV-5080, XV-5050. From first hand experience: if you want a lot of sounds and not the quality, XP-30 is an absolute winner. It you want max quality, then go either JD-990 or XV-5080. If you care for the high sheen filter sound, go with JD-990 as it can pull out the way XV-5080 can’t. But 5080 has much more waveforms (including some from Vintage Expansion) and has far superior effects, filter dynamic range and modulation engine (it features true matrix system). On top of that it can be used as a sample player since it has a “S-760 mode” (though that limits a lot of synthesis functions).


Even the latest XV-5080 has a full backward compatibility, all the way to the JV-80. You can also load all of the patches from JV-80, JV-90 and JV-1000 into JV-1080 and JV-2080. Just like you can load JV-1080 patches into the last of the series XV-3080, XV-5080 and XV-5050. They are all full compatible with only a few minor exceptions when it comes to waveforms. Even the old JV-80 patch will sound identical if you properly convert it. Some correction in resonance is needed because old models JV-80, JV-90 and JV-1000 had a Soft and Hard resonance setting, next to the resonance amount. Because JV-80 has two resonance settings, Soft and Hard. Their equivalent on Super JV and XV is as following:

  • JV-80 Soft setting, resonance set to max = XV-5080 reso set to 44
  • JV-80 Hard setting, resonance set to max = XV-5080 reso set to 88

What applies to XV-5080 applies to all Super JV and XP series. I came with this info by testing them side by side. This also gives you idea that the filter in JV-1080 can go way beyond old JV in resonance power. This is not surprising since it is a filter from the JD series. To cut the long story short, whenever you load a JV-80 patch into Super JV or XV you will have to modify the resonance value.

Antialiasing filter in Super JV is superior to the one in JV – which, depending on what kind of sound you like, is welcome or not so feature. Mirroring in higher frequencies, particularly when using rich textures can fool the listener thinking the unit is 44kHz waveform set, though in reality it is not, it is 32kHz just like Super JV. I talk about mirroring above 16kHz which can happen during transposition, thought the waveforms are all 32 kHz. This is just an artifact that happens with low interpolation quality algorithms. So in a way, old JV can sound a bit more open than the later Super JV series, because of the weaker anti alias filter in JV.

JV-1080 contains some of the JV-80 patches. JV-2080 contains all JV-1080 patches plus a bank of additional ones. XV-5080 and 3080 contain all of the JV-2080 patches, plus a few new banks. XV-5050 contains all XV-5080 patches plus a bank of additional Fantom patches (these are located in the User area 1-128).

Destructive compression?
With the Super JV series, on top of the existing Adaptive DPCM compression it seems as if Roland added an extra compression which is destructive form of compression. This is not confirmed anywhere in documentation. But at the same time it is trivial to test that something is going on by using a JD and any JV synthesizers, plus a spectral analyzer. If we play exact same waveform on both, some parts of the spectra are simply erased on the JV/XP/XV version. Now where have we seen that before? The good ole mp3 kinda looks like it, no? Of course it is not mp3 compression, because there was no mp3 back then, but the principle is somewhat very similar. Here is one example that clearly demonstrates it:


The same waveform was chosen on JD-990 and XV-5080. Please ignore the mirror effect label on the image, it relates to interpolation and that shouldn’t concern us. If we look at the waveform from 5080 somewhere around 15kHz we can clearly see a hole. There are a lot of such waveforms in Super JV and XV series that have holes in them. Very similar how mp3 works. And as you can see there are no such holes in JD-990 which makes it clear that JD-990 does not have this missing data. JD however use some other form of compression though, but we will discuss that below.

DPCM or a companding compression and Roland
On the Gearslutz forum, in May 2010 Eric Persing (source: here) mentioned that JV-1080 uses 8-bit companding compression. We can assume the same is true for the JD series as well. Unfortunately I can not confirm nor deny this, but I believe the man’s word since he not only designed most of these waveforms but figured out how to actually put them into hardware! What is not entirely clear from his statement was the exact compression method. If it is “phone line companding” type of algorithm – this is relatively old process which goes as following: Once the waveforms are sampled at the factory, they are being dynamically compressed and converted to 8 bit. The reason why they are compressed prior to that is to preserve low level information and somehow increase the dynamic range of this 8 bit file. At that stage they are put into machine’s ROM. Once the machine boots up it will load a waveform, convert it to 16 bit and apply dynamic expansion. Essentially the same thing what a compressor and expander that you have in your rack do, although these have 0 attack / release time. Data compression dates back into days when memory was very expensive, and manufacturers were looking way to squeeze as much as possible into fixed ROM space. Companding was one of the options where for every 16 bits of input, you would use only 8 bit to store them, yet with some tricks “preserve” the data. However, from my own research, and consulting people who have reverse engineered the ROM data of these machine, it seems that Roland does not use companding compression at all. Instead what I believe happened was that Eric used this word to make it more simple for average people to understand, since after all he is constantly in talks with audio engineers, and it would take too long to explain the exact algorithm so he most likely uses this as a short phrase for compression / expansion. The unfortunate bit in here that there was actually a compression method which contained that exact name.

It seems that Roland modules, all up until recently with the 2019 Fantoms, use DPCM compression type which downscales the data into 8 bit by a process of differential pulse-code modulation. This is a signal encoder that uses the baseline of pulse-code modulation (PCM) but adds some functionalities based on the prediction of the samples of the signal in two possible ways: 1) Take the values of two consecutive samples, quantize them, calculate the difference between the first one and the next, the output is the difference. 2) Take the difference relative to the output of a local model of the decoder process and quantize it. Compression ratios on the order of 2 to 4 can be achieved this way.

The question now arises: does that make Super JV and JD series 8-bit machines? Well technically speaking no. These are not just plain 8 bit samples in the ROM but 8-bit compresses samples. It makes a difference, because prior to being played, their dynamic range is restored and expanded to 16 bit. I haven’t meet a person that doesn’t like the sound of Super JV series and they would hardly believe these originate from 8 bit samples – but in a way, they do. In this regard we can also assume when Eric Persing mentioned the “companding” compression he was referring to DPCM, since the data is actually compressed into 8 bit and then later expanded into 16 bit (realtime using dedicated DSP hardware).

Engine and sample rate
Roland JV-1080 has a waveform set which is at 32 kHz. Its DAC runs at 32 kHz. We can see that in the image below. A sine wave was played at 8 kHz, and we can clearly see a mirror effect (aliasing) at 24 kHz. From this we can gather: 24 – 8 = 16. From this, Nyquist on JV-1080 is at 16 kHz. This tells us that a DAC runs at 32 kHz. In fact, just by looking at the picture you can immediately see that the whole image above 16 kHz is “mirrored”. You will have to click on the picture below for full size. Further more, by close inspection we can see a constant carrier wave at 32 kHz which could be the bleed thru signal of the DAC itself. Because i see no other explanation for a constantly preset 32 kHz signal, than the DAC itself.


I’ve read on GS forum some people claimed JV-1080 to be 44kHz DAC, but this is simply not true. If it was, then for start, the mirror effect (aliasing) would happen at 22 kHz, not 16 kHz. Another argument given was usually “this DAC can run at 44 kHz”. Yes, that is true. But it can run at 88.2 kHz as well! Even way beyond that without any problem. Looking at chip specs table isn’t always the best source of information. A simple measurement is sometimes all it takes.

Another argument that i read was 32 kHz DAC can not produce frequencies above 16 kHz. If this was true, then the assumption of that same person (original post here) that JV-1080 runs on 44.1 kHz is wrong as well. Because we can clearly see in the image above the unit goes way over 30 kHz. So does that mean DAC runs at 60 kHz? No it does not! The problem in here is the wrong assumption to begin with. A 32 kHz DAC can in fact produce frequencies above 16 kHz. This is considered an artifact and is known as aliasing. Back then manufacturers spent a ton of resources to suppress and remove as much of these as possible. As we can see Roland went for the simpler / cheaper option with some basic LPF filter behind the DAC, far away in specs of today’s brick wall filters. In fact service manual suggest this scenario as well. As a result of all that a lot of signal is aliased.


Image above shows a DAC chip world clock input (pin 13) on JV-1080. Signal is close to 5 volts peak to peak and is running at frequency of 32,00 kHz. The story of JV’s playback and engine sample rate ends here! For those interested in how i’ve obtained the data here’s a full story: In order to verify the assumption about the data shown on spectrogram, which shows mirror at 16 kHz and to be 100% i’ve downloaded specs sheet for the UPD63200. It is a DAC chip which is used in JV-1080. Next step was to find out the pin where the World Clock is located. And that turned out to be pin 13. After that i simply opened JV-1080, and located the chip. Luckily there is a via on the PCB board which can be used to connect the probe to, rather than touching the chip pins and risking of doing the short circuit (thank you Roland). So i connected the oscilloscope probe to pin 13. The result can be seen on the image above. Clock rate of the DAC chip was measured to be exactly 32,00 kHz. Just like we estimated by observing the spectrogram data. This confirms the earlier findings and verifies that JV-1080 is indeed a 32 kHz machine.

History tree



  • JV-80 (1991) = a true masterpiece of it’s time.
  • JV-880 (1992) = rack vesion of JV-80.
  • JV-1000 (1993) = JV-80 + MC-50mkII sequencer, added new waveforms, floppy drive, 76 key.
  • JV-90 (1994 ) = JV-1000, without sequencer and floppy.
  • JV-1080 (1994) = huge step forward for Roland. This was the most popular module of 90’s. New filters, voice structures, 448 waveforms, matrix control, new features.
  • XP-50 (1995) = JV-1080 with keyboard, sequencer, floppy
  • JV-2080 (1997) = JV-1080 big LCD (better user interface), 3 EFX, 8 x expansion slots.
  • XP-80 (1996) = XP-50 with 320 x 80 dot LCD (better user interface), 4 aditional sliders, more outputs, arpeggiator, 76 key.
  • XP-60 (1998) = 61 key version of XP-80. It replaced the XP-50.
  • XP-30 (1999) = XP-60 with added patches (waveforms) from three expansion boards (session, orchestral, techno), removed sequencer. By number of factory installed waveforms, this is the most powerfull XP and JV synth!
  • JV-1010 (1999 ) = JV-1080 in half rack module, session patches (waveforms) added.
  • XV-5080 (2000) = another big step forward for Roland. 1083 waveforms, 128 polyphony, true stereo voice – each tone (T1-T4) can be set as stereo (one waveform for the left, one for the right channel), SCSI connection, sample load, up to 128 MB of RAM (SIMM), 5 effects processors: 24-bit reverbs, COSM® modeling, RSS 3D effects plus standard JV’s Chorus and Reverb/Delay.
  • XV-3080 (2000) = XV-5080 without sample playback option, without COSM effects processor, smaller display.
  • XV-88 (2000) = keyboard version of XV-3080.
  • XV-5050 (2001) = XV-5080, without sample playback option, without SR-JV80 boards slots, polyphony reduced to 64, very small display. Size reduced to 1U, added USB support (editing via PC).
  • XV-2020 (2002) = XV-5050 in half rack module but no RSS effects, no COSM efx, no SR-JV80 boards slots, sound editing only via PC.

What was before JV-80?
JV-80 is based on PCM (Pulse Code Modulation) waveform playback. First of such made by Roland was model D-50 (1987), which became very popular. Not just only in the late 80’s, but also in 90’s (because of it’s analog synthesis emulation part which is quite powerfull – 4 oscillators per patch, nice smooth 12 dB resonant filter, 6 LFO’s, pulse width modulation). Next PCM synthesizer from Roland was U-110, which was later replaced by U-220 along with keyboard version labeled U-20. It was a very limited synthesizer with no filters of any kind, no assignable LFO’s, primitive pitch and vibrato adjustments (no envelope). The U-20 was in 1990 followed by U-50 which will be in the last minute renamed to D-70 due to popularity of D-50. D-70 had upgraded U-20 engine, some new waveforms and most importantly it added a resonant multimode filter. D-70 is definitely one of the most mysterious Roland synths, often overlooked and forgotten. The reason might be a bit hard user interface which has some impractical solutions that can make your life harder rather than easier. In parallel to D-70, Roland put out MV-30 which is very similar engine with added MC-50 sequencer. Finally in 1991 the JV-80 came out and this is where the legend began.

Quality issues with JV/XP series
At one point, in the mid 90’s, Roland switched to using SMD electrolytic capacitors. This has its benefits (gear has less weight) but drawbacks too (it can be harder to service). With that being said, it was discovered, first by users and then later confirmed by Roland themselves, that the electrolytic capacitors in Roland SR-JV80 expansion cards were not of good quality and by now (2019) many of them are failing. I have determined that the same capacitors were used at least in one XP synthesizer, model XP-50. Many of these caps have failed by now. In fact I have one of these myself and had to replace all of the SMD electrolytic capacitors. First symptoms were that audio would no longer work at the output. The good side of the story is, JV-1080 and JV-2080 owners are in a safe position as these actually use thru hole electrolytic capacitors. I can not confirm their quality level, but I never heard of any of these units failing due to bad capacitors. They are safe to use and operate for many years to come, which is something that can not be said for XP-50.

Some final words on the JV-80 vs JV-1080
They sound different due to 1) different digital filters 2) different anti alias filters.

  • Super JV has a filter from JD series (or a very close version of it). JV-880 has original filter from JV-80 series (also used in JV-90 and JV-1000). Emulation of that filter is possible with Super JV though it is less precise as you have less values to choose, particularly if you’re trying to emulate the “soft” resonance option from the JV. We discussed resonance compensation values above for both the hard and soft setting in the JV-80.
  • Antialiasing filter in Super JV is superior to the one in JV – which, depending on what kind of sounds you like is – welcome – or not so welcome feature. Mirroring in higher frequencies, particularly when using rich textures can fool the listener thinking the unit is 44kHz waveform set, though in reality it is not, it is 32kHz just like Super JV. I talk about mirroring above 16kHz which can happen during transposition, thought the waveforms are all 32 kHz.

Super JV was based on a far superior RISC processor which at that time was state of the art (sort of) hence the machine can take a lot of modulations real time, without sustaining damage on evelopes and LFOs – which again is welcome or not so welcome. This depends whether you prefer jumping envelopes as “more analog” while you tweak some parameter live on a synth. Which one should you buy? Well, JV-80 was really cool synth, however on your place i would go with 1080. I tested JV-1000 against Super JV and you can practically cover all of the JV sounds, minus aliasing artefacts! So for the harsh sound factor (alias abuse), or 100% authenticity, you will go JV-80/880 route, other than that look into 1080 or even better 2080 direction.

Roland experts
When it comes to experts in the Roland synthesizers that we covered in here, first name that comes in mind is of course Eric Persing. He used to post on a Gearslutz forum as a member “spectrum” and with a little help of the search tool one can find a real gold mine of valuable infos and resources. You can use this link to find some of his posts. Another name the comes to mind, especially about the nerdy details about ROM set and the Roland compression schemes it is definitely Edward from D-Tech. I highly suggest you visit his web page to learn more in-detail about the waveform ROM of these Roland romplers we have covered. Link here

ASR-10 synthesizing industrial sounds


I love ASR-10! Here’s a Roland TR-909 cymbal turned into a factory drone. I will provide a step by step guide here on how to create one. To get this atmosphere i used a standard TR-909 ride cymbal. First task was to loop it from around 30% until 80% (loop end). One thing i like about ASR-10 is that it features several crossfade methods for making loops smooth. Those are:

  • Crossfade Loop
  • Reverse Crossfade
  • Ensemble Crossfade
  • Bowtie Crossfade Loop
  • Bidirectional X-fade
  • Make Loop Longer
  • Synthesized Loop

For this particular purpose i used Synthesized loop since it adds its own flavor, depending on what Smoothness method you use. It almost completely removed the volume differences. After that point i copied the loop addresses and set these values to sample start and end address. After that i used Truncate Wavesample to keep just the looped part. Then i normalized sample.

To add even more flavor and bring up some harmonics from the background i then applied a function called Volume Smoothing which is a sort of a dynamics compressor that further removes dynamic changes making the sound more constant. This function has Smoothness option as well and i’ve used Fine setting. It took a while for ASR-10 to do the processing.

Now it was time to expand the sound into stereo field. I loaded the last 44.1k effect that comes on OS V3.53 floppy disk which is called “Parallel EFX” and used a preset in it called “Smaller Spaces”. It adds some industrial flavor as well.

Now it was time to resample it with this effect. I pressed the same note across three octaves (D1, D2, D3) and sampled about 5 seconds of it. Trimmed it to remove the 0.5 sec of beginning and about 1 sec of ending (to remove empty space at the end). Then i’ve normalized it. I’ve set loop mode to forward and that was it. No need to crossfade loop this time since there are a lot of harmonics and no click will be heard.

After that i’ve loaded “ROM-02 44KHZ Reverb” and used its preset called “Long Reverb”. Only thing left to do was to go to filter, set it to “3pole LP / 1pole LP” and manually open it. The result is what you hear.

ASR-10 synthesizing an organ

Here is a video tutorial I’ve made some time ago. If you use software such as Chickensys Translator it is very easy to transfer samples from a computer to an ASR-10. Sometimes a single cycle sine wave is all it takes to make a good organ sound. This video demonstrates the raw power of ASR-10. Alternative solution is to sample a sine wave, normalize it and proceed as explained in the video.

There’s a very useful thread on Gearslutz forum that details SCSI with old samplers and provides some solutions such as SCSI-CF adapters that can work with the sampler and can be used as a bridge between your computer and a sampler. Here is a link to it.

Let’s face it, we don’t use these samplers because of their specs, we use them for their sound. And so far there aren’t many alternatives on the software market. Investing into something like a CF drive is not a bad idea at all. These samplers once cost thousands for dollars. You can have them for a fraction now. Their sound didn’t degraded over the years at all.

Eventide H3000 as a formant synth

Image copyright: John R. Southern

Here’s an interesting “Formant Morph” processor i’ve recently designed using the Patch Factory (aka algorithm 111). As you may know, Patch Factory algorithm as itself is not an effect, but more an effect designing mini studio which is why it unfortunately gets overlooked by many of the Eventide users, since (by default) it does not produce any special sound.

What we will build will be a primitive version of a vocal Formant filter. Proper vocal emulation requires at least three bandpass filters, while Eventide H-3000 unfortunately offers only two BPFs in the Patch Factory. Still, better something than nothing. Here is how it will sound on various settings:

External_In.mp3 – Here the effect is first applied to a saw wave. You can hear sample&hold morphing of the vowels as they shape “aaahh”, “eeehh” and the others. Then i processed the beat with the same settings. Then speaker’s voice with some variation on the settings.

Stereo_Version.mp3 – More exotic sound can be obtained by building a stereo effect using the second delay line. See the bottom of the text for this optional mod. Listen to this one on the headphones.

Stand_alone.mp3 – If you will use noise generator included with the patch factory, here are some of the effects you can achieve by doing small variation on the parameters. Much more can be created of course, only limit is your imagination.

Building the structure

First thing to do, once you load algorithm 111 is to go to Expert parameters (via Parameter button) and choose Patching. The goal in here is to patch everything to Null. Goes very fast, you don’t need to dial the value, just type 99 and press enter, this will select Null Input and press Parameter button until you browse the full circle through patch destinations (Filt1 In, Filt2 In, Delay1 In…etc.). At this point everything is disconnected. Now we will build the structure:

  • Browse via Parameter button until to get to the Scale1 In and patch a Noise Gen to it by selecting the Noise Gen as a Patch Source. IMPORTANT: If you look at the picture below, you will see that it shows a little bit different situation – instead of the noise, left input is connected, however that is how the final product should be. But at this point, you should connect the Noise Gen instead! This is for calibration purpose that we will talk about later. Scaler is also a very important element and please don’t omit it, just because you can do it without the scaler. Trick is about the correct levels which will later have to be dialed on the Scaler1, to get the correct harmonic enhancement for our formant shaper without going into distortion. This is why we patched the loud white noise source. Using it, you will calibrate the scaler later on. For now, forget about this.
  • Now it’s time to split the signal into two filters. Go to Filter1 In page and select Scaler1 for the patch source. Repeat the same with Filter 2 In.
  • Browse now to the Sum 1a In and select Bandpass1. Browse to Sum 1b In and select Bandpass2. This is the core of our effect. Two bandpass filters have just been connected into series.
  • We will spice our effect with one short delay line to get a nice flanging, which always sounds killer on Eventide machines. For that job we will need to split the signal again. First go to Delay1 In and select Sum 1 as its patch source. Then go to Sum 2 to join the delayed signal with original by doing the following: Sum2a In: Sum1 and then Sum2b In: Delay 1.
  • IMPORTANT! Prior to doing the next step please reduce the level of your amplifier or headphones to MINIMAL! Serious hearing damage might happen to you and you might damage the speakers as well!! This is not a joke! A lot of loud noise might be produced at this stage!
  • Now connect select L Output and choose Sum 2 as its source. The Sum2 is now connected to the output and we’we built the structure.
  • Its time to properly set each of its parameters. You will do that by exiting the Expert function.

Back to the Parameters now. We have to do some calculation first. Our goal is to build a formant morph that will go in between two vowles: “Eeeh” and “Oooh”. As said earlier, we only have two filters at our disposal, so we will use the upper two peak frequencies (out of three). Quick google shows us the following:

“Eeeh” 3010Hz and 2290Hz as its peaks
“Oooh” 2410Hz and 840Hz as its peaks

Since the LFO modulation is alternating between positive and negative values we need to calculate 1)the center frequency from which we will apply the LFO to sweep between the 3010Hz and 2410Hz; and 2)the LFO amount deviation needed to reach these two values:

Peak 1 morph calculation
3010 + 2410 / 2 = 2710Hz center frequency; from which
2710 – 2410 = 300Hz LFO modulation amount

Peak 2 morph calculation
2290+840 / 2 = 1565Hz center frequency; from which
1565-840 = 725Hz LFO modulation amount

Setting the Parameters

  • Go to Scale 1 and set it to 15%. (now you can increase the headphones / speaker volume).
  • Go to Parameter Cutoff1 and set it to 2710Hz. Go to Cutoff2 and set it to 1565Hz. This sets the filters right in the middle between the upper and lower points of the two vowels.
  • Go to the Q Factor1 and set it to 950, and set Q Factor2 to 950.
  • Go to the Delay 1 line and set it to 0.40ms. This is the center frequency from which our flanger will operate.
  • Now we’re done with the parameters.

Setting the Modulation

  • Push the FUNCTION button twice. This gets you into the modulation and LFO page.
  • Chose FuncGen and then select Function from the menu. Select Sample & Hold from there. Set Rate to 4Hz and Amount to 100.
  • Now select Patch from the Modulation of Parameters screen.
  • Turn the main knob until you select delay 1 and set its source to Function Generator. Press Done. Set range to -0.30. This will sweep the flanging effect from 0.10 to 0.70ms, since we previously set the center of the delay at 0.40 miliseconds.
  • Now we just need to input the LFO amount for the formant filter. Go back to Patch again and turn the knob until you select cutoff 1. Move the knob wheel and set its source to Function Generator. Press Done. Set the range to 300.
  • Repeat the same but chose cutoff 2 now and set the range to 725

At this point you should already hear sample&hold doing “formant synthesis” to the white noise source and vowels going out of your speakers (or what’s left of the vowels, given they’re made of just two bandpass filters). Only thing left to do is to replace Noise Gen with the Left Input at the Scaler1 patch. You will do that by going back to expert settings via Parameter button and press it until you see Scale1. Then just select Line Input as its Patch Source.

Now it’s time to go to your console and route Aux into the Left Input. This is where you will have to “calibrate” the Scaler1 to avoid clipping and distortion. Simply go to Parameter settings and adjust Scaler1 in between 1-100% volume, depending on your console’s or sound source’s volume. You will notice distortion as unpleasant artifacts, each time a peak hits into the Eventide. Most simple is to just set the scaler to 100% and regulate the amount of volume being sent into Eventide via Aux pot at the console. However (!) for white noise effects (stand alone) you will have to set the Scaler1 to the low volume, else the loud noise will start distorting once it enters the resonant bandpass filter. This is why i’ve said Scaler is important. In fact will will even add a soft function for its control, so that you don’t have to enter the patch settings every time you want to readjust it.

Stereo fun and Soft Function

For more stereo width and to put some exotic vibe into our effect, we will make its stereo version. First thing to do is to go to Expert settings and enter the Patching.

  • Got to delay2in and select sum 2.
  • Got to r output and select delay2 there.
  • Press Return to go back to standard parameters and navigate to the delay 2 and set it to 20ms.
  • You should already hear chorus like stereo width that has been just produced. However we will do some more fun now.
  • Go to the Functions module (press function button twice) and go to Patch.
  • Turn the knob until you select delay 2.
  • Select Function Generator as its Source and set range to 10.00. Done.
  • Last step is to add the soft function the the scaler. First go to Parameter settings and set scale 1 to 0% (at this point formant synthesizer should become silent).
  • Press Function button and select SoftFunc.
  • Press Name and rename it to Preamp. Set Sensitivity to 100 and Pol to +. Press Done.
  • Press Function button and press Patch. Select scale 1 with the knob and press Source. Select Soft Function 1 “Preamp” and press Done. Set range to 100.
  • Save the patch i.e Formant Synth.
  • This completes our job.

Tip: To match the beat you’ll need a calculator (not all the beats can be matched). For 120 bpm, (which is essentially 2Hz) you want to chose values that are either multiples or the same as the beat frequency i.e. 2.00Hz, 4.00Hz, 8.00Hz, etc. or you can halve it to 1.00Hz or 0.50Hz. Enjoy the fun! If you have any tips, feel free to add them to the Eventide H3000 Club thread.

If your sequencer slightly gets out of the beat for a small amount, you can go to the Function Generator (via Function button) and simply press the Trigger at the beat start. This should correct the problem.

Andromeda A6 sound improvement


Disable Background Tuning (i mean, do it now!)
This will provide your Andromeda with more fat sound, due to routine in Background tuning (which is enabled by default) and which is too perfect. Putting all oscillators on exactly the same frequency makes the unit a little bit sterile in sound. All the vintage analogs had imperfectly designed VCOs that would float slowly in pitch up and down on a miniature scale. With Background tuning enabled this free VCO float is killed and the Andromeda becomes more cold and DCO sounding. If you’re analog purist, you should disable the Background tuning. Also, according to some sources, Background tuning eats some CPU power reducing the overall performance. This is the procedure on how to disable and perform the necessary auto-tune after that:

Turn on Andromeda. Leave it on for about 30min so that the board reaches the “average” temperature. Disable Background tuning in the Auto Tune section. Now engage the auto tune (press Auto Tune button twice).

From now on, your Andromeda should be stable in pitch after the board reaches the “average” temperature to which you calibrated voltages of each voice and filter. Once about a month you can do the Auto Tune routine.

Set proper levels (the most crucial part of Andromeda!)
While checking my Andromeda i found one interesting feature that was maybe intentionally implemented but which at the same time might confuse new owners. It has to do with pre filter signal levels. For some reason, if the combined level of two oscillators exceeds 30 or if one of them is set above 30, waveshape takes place, waveform becomes clipped (peaks are cut off) resulting in somehow poor sound. A simple oscilloscope reveals this as well. Therefore, you should never set high VCO levels i.e. 100 unless you intentionally want this clipping (which is interesting at first, but somehow becomes annoying “plastic” sounding after a while). Some users found other stages of Andromeda to have similar behavior with excessive levels as well. We will summarize them all here:

  • The pre filter mixer will overdrive with combined values exceeding total of 30 (i.e VCO1=15 VCO2=15; or VCO1=30 VCO2=0; should work good).
  • The VCA might sometimes overdrive with default level of 100. You should therefore set levels of volume envelope at 80-90 instead. This will also improve the envelope response (see below).

Temperature tuning
If precise pitch is not critical in your track, for even more “analog” sound, you can disable Temperature tuning as well. However, disabling Temp tuning is not recommended if you do live gigs, as there’s a big chance your Andy will be way off the rest of your setup or band, to the point you won’t be able to compensate it with the Tune knob. So please keep in mind that playing this synth with disabled temperature tuning is risky (though it can be amusing if you’re hard core 1970′s analog fanatic).

Recommended download: Andromeda Tips and Tricks


The following info was found of one of the Andromeda bulletin boards.

—– Original Message —–

From: “CCJ” <coolcolj at *****>
To: “Protokol13″ <protokol13 at ****.com>; <A6 at ****.com>
Sent: Friday, December 20, 2002 4:06 AM
Subject: Re: [A6] Andromeda tips[/I]

>/ The pre filter mixer will overdrive with values about 30,
/>/ the post filter mixer will do as as well with values above 50
/>/ the VCAs are also overdriven a the default level of 100 in ENV1.
/>/ Setting ENV1′s level to 80 or below will give a clearer and brighter sound

Yes, this seems to be the case with what I found out on my side. I usually bring the sustain level on the VCA below 90, but ~80 seems to be where I get cleaner sounds. As I said in my previous message, I like staying away from what sounds like slurred attack rates by also dropping the attack level. It seems like the VCA with a setting of 100.5 in the attack stage, coupled with 100 in the sustain stage has no real ‘punch’. I get good results with a 80-something sustain, and an attack level in the 90′s… I think this might be the source of some of the confusion that people have had with sounds that aren’t snappy on the A6. It sure was something that bothered me when I first started programming on this synth, but I have no problem with percussive/bass sounds now. I think it’s more important to shape the envelopes carefully, then to default to using engine optimization automatically.

Deep FM bass on Roland JD/JV/XV series


Compatibility: JV-80 and up

Although Super JV/JD have the FXM section that is based on a frequency modulation, it is actually quite limited terms of real FM sounds – it is more oriented towards spicing the sound with a specific character (or making it more ‘wild’ as Roland manual says). For real FM sounds we must look elsewhere.

Super JV/JD is not an FM synth, but it has a nice LFO that can run pretty fast, and with a little experience in real FM programming it is not hard to recreate some basic FM sounds. Please keep in mind that we talk about really basic FM sounds created from only two operators (Yamaha DX-7 for example has 6 operators). And even those ”two operators” we will build on Super JV/JD are very primitive, compared to any real FM synth.

Deep FM Bass
We will create one of the deepest basses ever, that goes subsonic, much below 20 Hz. I first built this bass on the Yamaha SY-77 some long time ago, but since it requires only two operators, i recently came to idea to try to emulate it on the JV/JD synth. Ok, it will not sound as powerful as the real one, but it will demonstrate that it is possible to do some primitive FM on the JD, JV, XP, XV synth line.

We will be using two operators. The WG (tone generator) will be the carrier, and LFO will be the modulator. Sound will be made by the classic two point down ramp envelope applied on the modulator – that is, the modulator level starts loud and then fades away. On the real FM synth you would do that with an envelope. Unfortunately on the Super JV you can’t apply an envelope (ENV) to modulate the level of LFO, so on the first sight it appears our FM sound won’t function properly. But there is a workaround for that issue. We will build the envelope on the LFO using ramp, which Roland just calls LFO Fade In/Out function. In other words you got simple two point envelope that can be applied to LFO – i know it is primitive, but better something than nothing. With this ramp you can create dozens of bells and metallic percussion, if used the right way. Here is detailed procedure:

  • Initialize the sound
  • Enable T1, disable all other tones
  • Go to Control and set Key Assign to MONO
  • Set WG1 to Sine
  • Go to Pitch, set Coarse Tune to -12
  • Go to LFO and set it to SAW-DW
  • Jump to TVA1 and disable velocity (V-Sens=0)

You probably noticed that we used Saw wave in the LFO instead of Sine wave. We had to use the saw to add more punch to the sound, because JV is not a real FM synth, and with a sine wave LFO, the sound becomes too muddy in the low C1-C2 region. However, later you can try switching LFO to sine wave and try C2-C4 notes that will sound better with it. You will also use sine wave in the LFO for all metallic and bell sounds. Or to be precise, you should use sine wave LFO in almost all FM sounds, except those in low pitch range where you will use SAW-DW wave to compensate the lack of punch.

  • Within LFO set Depth Pitch: +50
  • Set Fade Mode: ON-OUT (this is our ramp down envelope)
  • Go to TVA and make a short sound using following parameters
  • Time: 0 20 42 42
  • Level: 127 127 0
  • Now hit A1 and C2 few times
  • For shorter and more distinctive bass set time to: 0 10 42 42

Although nature commences with reason and ends in experience it is necessary for us to do the opposite, that is to commence with experience and from this to proceed to investigate the reason. – Leonardo da Vinci

How to achieve PWM on Roland’s SuperJV / XP and XV series


Compatibility: JV-1080 and up
Audio example: PWM.mp3

The super JV series features two saw waves that have inverted amplitude to each other. A little bit of math shows us if we play them both, we will get silence at the output, but if we detune one of them, we will get Pulse Width Modulation. Basic procedure:

  • Initialize the sound.
  • Turn on T1 and T2.
  • Go to WG1, and select ‘Synth Saw 2’
  • Go to WG2, and select ‘Syn Saw 2inv’

Ok, now we got the basic setup. Next thing is to create detune. To avoid the modulation sound exactly the same each time we play the note, we will create detune by using Random Pitch.

  • Set Random Pitch on WG2 to: 1

Now we must be careful here, because random means sometimes 0 at the output, and that would result in no detune = silence. To prevent this we will add Fine Tune which must be at least +2. Why? Because there are two possible cases. In case a) random gives 0 at output, Fine Tune of +2 preserves non zero value (it will be +2). In case b) random gives -1 at the output, Fine Tune of +2 again preserves non zero value (total fine tune will be +1).

  • Set Fine Tune on WG2 to: +2

I assume most people would use PWM for the bass, therefore:

  • Set Coarse Tune of WG1 and WG2 to: -12

You will notice the sound plays very slow pulse width modulation. To give more expression we will add one controller to modify the pitch of one oscillator. Please be careful here. You can’t assign this modulator to ‘any’ Tone you desire. It must be the tone that we applied detune function. In our case, this would be the Tone 2.

  • Go to PATCH LFO&Ctrl #1 (Matrix Control)
  • Matrix Control 1 Source set to CC01: MODULATION
  • As Destination set PITCH, and put +6 to Sns (sensitivity)
  • Disable Tone1 within matrix to make it look like this: PITCH : +6 -> _ooo
  • Set TVA as necessary

Now when you move Modulation Wheel up, you will fasten the Pulse Width Modulation. If you want faster PWM by default, put higher values at WG2 ”Fine Tune”. And that’s about it!

By three methods we may learn wisdom: First, by reflection, which is noblest; Second, by imitation, which is easiest; and third by experience, which is the bitterest. – Confucius

Roland JD emulation on Super JV and XV synthesizers


Starting with model JV-1080, some waveforms from the JD-800 were transferred into JV-1080. Which meant back then that some of the patches could technically be transferred from JD into Super JV synthesizers. Unfortunately what Super JV series missed was the effects section from the JD, and thus most of the patches were a total miss. (Notice: Well this is just half of the truth, the other half is a different gain structure, filter dynamic range and 44kHz waveforms vs 32 kHz ROM, but let’s pretend for a moment we have no idea about this. If you’re curious about details, go to our Ultimate Roland JD JV FAQ article). Anyway, the process of JD “migration” continued with XV series, to the point that many of the 108 JD waveforms seem to be available in the XV synths – seems like 7 are missing – but they could be different name. This part is unfortunately unconfirmed and requires someone doing more in depth waveform tests.

Of course what would be a JD without it’s special multi effect processor. That’s why Roland implemented JD’s “Effect processor A” into XV. In other words, you got a JD synth hidden inside your XV synth, and you can finally start converting favorite JD patches. There are some differences in the filter, but more on that later. I should just state that the 44.1k referenced samples points to models XV-5080 and XV-5050. I can not guarantee that model 3080 contains 44.1k playback engine at all, neither the samples in that format – it has been reported the machine is 32k. I can however guarantee than in 5080/5050 waveforms from the JD-800 are in original 44.1k format.


Table below shows us internal memory content (waveforms) of the JD-800. Starting with ‘’001 Syn Saw 1′’, ending with ‘’108 Wind Chime’’. Position of these same waves inside XV synthesizer are marked with orange color. For example if you want to load Syn Pulse 4 that on JD is waveform number 008, on XV you will find it on number 557.

JD-800 multi effect group A
With the XV synthesizer, Roland also brought us back the famous JD-800 multi effect from its section A block (note: the JD has two effect sections). On XV series it is available as MFX number “75: JD MULTI”. Just like on the JD-800, it allows distortion, phaser, spectrum and enhancer effects to be connected in series in any desired order. It features exactly the same settings as available on JD-800. Here is a brief explanation for each one of them.

1. Distortion
The first effect in the chain is obvious – a standard distortion. This effect is useful in situations when you wish to add some drive to solos or do some nasty clipping effects depending on the sound design application. There are seven types of distortion available:

  1. MELLOW DRIVE: A soft, mellow distortion; somewhat darksounding.
  2. OVERDRIVE: The classic sound of an overdriven tube amp.
  3. CRY DRIVE: Distortion with a high-frequency boost.
  4. MELLOW DIST: Sounds like the distortion you’d get from a really big amp.
  5. LIGHT DIST: A distortion with an intense, brilliant feel.
  6. FAT DIST: Boosted lows and highs gives this one a thick, fat sound.
  7. FUZZ DIST: Like FAT DIST, but with even more distortion.

2. Phaser
In typical phaser, modulation effect is created by mixing original sound with a phase shifted one. Result is a swirling effect and is best suited for backing sounds such as strings or electric pianos. Phaser will be most effective on sounds rich with harmonics, such as saw or pulse waves. Therefore it would be better to insert the phaser after the distortion or spectrum. For the best results, you should use center frequency at around 1 kHz.

3. Spectrum
Spectrum is an effect that modifies sound by boosting or cutting specified frequency areas, resulting in different tone colors. This effect might look similar to an equalizer. However, the frequency of each band has been set at the optimal location for adding a distinctive character to the sound. Rather than correcting the sound, spectrum allows you to aggressively modify the tonal character.

Spectrum will be best heard on spectral rich sounds such as white noise. There, the change will be most evident. For most expressive result use narrow bandwidth (set it to 5) and try setting all bands to max gain (positive or negative). When using wide bandwidth settings (set to 1) sound becomes less distinctive, and it starts to sound like an ordinary EQ.

4. Enhancer
Enhancer is a sort of aural exciter type of effect. Can be effective for sharpening up the vocal types of patches, flutes, guitars, etc. It will really help the instrument (patch) stand out in the mix. Its function is to generate new overtones out of the fundamental ones. With sensitivity you can set the depth of enhancer effect. While with the mix parameter you are specifying the mixture of original sound and the newly created sound overtones.

Effects setup on XV
Image below shows us the real JD-800 effect processor routing. As you can see, effects group A is connected in both series and parallel to group B. Same thing can be done in XV. The only difference is that on XV there is no effects group B, but instead there is separate chorus and reverb/delay. Since they can be configured in series or parallel, you can think of them as “group B” with only difference that you can have either delay or reverb, but not both like on the JD.


Image below shows us typical JD-800 effects setup emulated on XV. Chorus and reverb simulate JD’s “effect group B” while MFX: 75 JD Mlt provides “group A”. In this example, group A is connected in series to group B. Inside group B we connected chorus and reverb in parallel (M+R), so that we get chorused signal out followed by reverb/delay (in this example i used Reverb 1, type: Delay).

It is possible to have delay and reverb at the same time, but you will lose chorus. If this setup is required, just set chorus to type 2: delay (200-1000ms). Now you will have both delay and reverb. Please note this emulation will sound nothing like JD Effects Block B since they contain very different algorithms while some cult ones like Flying reverbs are missing completely.

Conversion table
Before starting to build or convert you first JD patches, keep in mind that JD and XV have different filter numerating system. For example, max resonance on JD is 100 while on XV is 127. Same is with the cutoff. And exactly the same thing applies for other parameters that on JD go in range from 0-99, while on XV and Super JV they go from 0-127. For better conversion of your JD patches you will need this JD/XV conversion table.

Roland Super JV?
Ok why giving hope to Roland Super JV users by placing it in the same title? Let’s say it is for those who are constantly sending me messages “how do i get this JD pad converted into my Super JV”. Well, to be frank, you can’t! You can try it. But you will never get there. Ok? These two machines have different gain structure and dynamic range which makes JD sound a little bit “harder”. For example you can not make soft sounding bass line on a JD, it will always have this hard character to it (not soft in a way you can make it on Super JV). The reason is higher gain and most likely smaller dynamic range of the JD filter section. But on the other hand you will never achieve those legendary high frequency shimmering pads on a Super JV, simply because i can’t go that high, neither it’s filter, neither its waveforms (which are 32kHz, compared to 44k on JD). And as already mentioned these two devices have different numeration. So if you still INSIST, here i am providing the above table for those of you who want to convert their patches. And for the second time, even if you convert all the parameters correctly, you won’t achieve JD’s sound on a Super JV machine, just like JD will never achieve the sound of a Super JV (which is darker, but also a much softer sounding – hint: analog style bass patches on Super JV are simply stunning). Ideally is to have both machines. So, there you go…

Roland JD-990 Resource Centre


JD-990 Software / Patches:
Patch Organizer
Patch Converter JD-800 -> JD-990
Free Patches (61 bank total)
JD-800 Factory Patches (converted to JD-990 format)
SR-JV-04 Vintage Expansion JV patches*
SR-JV-04 Vintage Expansion JD patches

*can be used in any Super JV, XP and XV synthesizer, and in JD990 of course.

Vintage Expansion Patches “Hack”
Well not a hack, more like a work around. If you own JD-990 and a Vintage Expansion then by now you probably know you can not load its patches directly as you do regular presets. Instead you have to initialize the sound, then go to the card selection, etc. To simplify this a bit, in above SYX file titled SR-JV-04 Vintage Expansion JD patches you will find 4 SYX dumps. These originate from the Vintage Card. So now you can simply send them to your JD and have them in normal access mode, since they will occupy User RAM area. Downside is, the hack works for only 64 patches at a time. But, as they say: better something than nothing. Simply write down the patches you like / favour, repeat the same step for other three banks. Then use Patch Organizer (included on this page) to build your own compilation of 64 favourite Vintage Expansion patches.

JD-990 DAC Calibration
A noise can occur during long release times. If this is the case, you will need to calibrate the unit. Noise itself is the result of improperly trimmed DAC (digital / analog converter). Type of the noise we discuss here sounds somehow like a bitcrusher or a bad working noisegate – it’s not the hiss we talk about here. Here’s the procedure:

  1. Open the top cover of JD-990.
  2. With the JD turned off, hold down INC + DEC buttons then turn power on.
  3. This should get you into the test mode.
  4. Then press UTILTY + F3. This gets you into the MSB Adjustment page.
  5. Make sure the volume control is at maximum. (Watch out you speakers!)
  6. Press the value knob, a 440hz sine-wave is produced.
  7. Use an instrument such as an oscilloscope to observe the output waveform. If you don’t have oscilloscope, you can use headphones and listen carefully with your ears.
  8. Adjust VR1 to obtain a smooth sine wave.
  9. The VR1 is located on the main board, top right side of it (if JD is placed with its front side facing you). In fact it is the only trimmer potentiometer there.


JD-990 Factory Reset
Warning: this will delete all of your presets. Please dump them via sys-ex before doing this procedure

  1. Hold Exit and press Utility.
  2. Screen says Internal ALL = Factory Preset.
  3. Press Execute (F6).
  4. Press YES (F5).
  5. Done!



Roland JD-800 Resource Centre


JD-800 Software / Patches:
Patch Organizer
Free Patches

JD-800 DAC Calibration:
A noise can occur during long release times. If this is the case, you will need to calibrate the unit. Noise itself is the result of improperly trimmed DAC (digital / analog converter). Type of the noise we discuss here sounds somehow like a bitcrusher or a bad working noisegate – it’s not the hiss we talk about here. Here’s the procedure:

  1. To Enter the Test Mode select Multi-Mode
  2. Press and hold: Cursor Left, Cursor Right and then Exit buttons.
  3. Use an instrument such as an oscilloscope to observe the output waveform of MIX OUT L on the rear panel. If you don’t have oscilloscope, you can use headphones and listen carefully with your ears.
  4. Press [NUMBER 7] while holding down [EXIT].
  5. A sine wave with a relatively low sound volume should be output.
  6. Make it a smooth sine wave by adjusting VR1 on the Jack board.


JD-800 Factory Reset
Warning: this will delete all of your presets. Please dump them via sys-ex before doing this procedure

  1. Press data transfer.
  2. Page up to select factory preset.
  3. Press YES.
  4. Done.


JD-800 Brochures:

 photo jd800_br2.jpg  photo jd800_br3.jpg
 photo jd800_br4.jpg  photo jd800_br5.jpg
 photo sos491-1.jpg  photo sos491-2.jpg  photo sos491-3.jpg  photo sos491-4.jpg

Synthesis types

Additive (Fourier) synthesis
Amplitude (ring) modulation
FM synthesis
Granular synthesis
Linear Arithmetic (LA) synthesis
PCM sample playback synthesis
Phase Distortion synthesis
Physical modeling synthesis
Realtime convolution and modulation (RCM) synthesis
Subtractive synthesis
Vector synthesis
Wave sequencing synthesis
Wavetable synthesis

Additive (Fourier) synthesis
Every sound in the nature, no matter how complex, can be expressed as a sum of sinewave functions of various frequencies. Those can be partials or harmonics of the original fundamental frequency. Each harmonic is an integer multiple of the fundamental frequency while partial isn’t.


In the image above we have an example of fundamental frequency sinewave and it’s 2nd and 4th harmonics summed to create the final sound.

Now, lets take another example, this time in frequency domain. If we take a short snapshot of the sound of electric guitar and look at it’s spectral characteristic, we will see that it contains peaks at some frequencies, and valleys at others. Just as seen in the image below.


In next millisecond these peaks and valleys move a little bit and go to different frequencies. Now imagine you have a generator that can generate sinewaves at the same frequencies where guitar creates these peaks, and control the volume envelope of each sinewave. This generator is exactly what additive synthesizer does.

Such synthesizer has a bank of oscillators which are tuned to multiples of the base frequency (harmonics). And each oscillator has its own volume envelope. The more realistic you want additive synthesizer to be, the more oscillators you need.


The name Fourier synthesis comes from Jean Baptiste Joseph de Fourier who (among many other things) found out that every sound can be formed from summation of sine waves. The most known additive synthesizers are Kawai K-5 and later model K-5000 which has over 1000 parameters per patch, so if you like editing for hours, there’s a nice addition to your studio setup.

Amplitude (ring) modulation
In general, modulation is the process of varying a carrier signal (usually sinusoidal signal) with a modulating signal. This can be done in three ways, by modulating: phase, frequency or amplitude of the signal. A device that performs this modulation is a modulator. What we will cover in this article is amplitude (ring) modulation.

Image above shows us typical amplitude modulator. Let’s assume that we bring two sinusoidal signals at modulator’s inputs. The first one (f1) has a frequency of 1000 Hz, and second (f2) one has a frequency of 100 Hz. In mathematical terms, what amplitude modulator does is multiplies two input signals. Please keep in mind that we don’t talk about multiplying numbers (in this case 100 and 1000), we are talking about multiplying sine waves. This is completely different story, and a little bit more complicated. If you are interested into this, get a math book and read about multiplying of two sine waves. Since i don’t want to bother you with too much ”why stuff”, lets just say that at modulator’s output we will have their sum and difference: f1+f2 and f1-f2, which means 1100 Hz and 900 Hz respectively (these are the frequiencies, not plain numbers). Spectrogram below shows us result of mixing f1 and f2 inside amplitude modulator.


Time domain
Change of carrier’s amplitude in a function that depends by the level of modulating signal results in a process we call amplitude modulation. This can clearly be seen on the image below. Vertical axis shows the amplitude, while horizontal shows the time (this is a typical waveform display). Modulating signal of human voice (image 1) modulates the amplitude of the carrier (image 2) which results in modulation (image 3).


carImage 2-carrier

outImage 3-modulation

These images are showing a time frame of only few milliseconds, just to show you a brief conception of the mixing process inside amplitude modulator. Image below shows us combination of image 1 and image 3, so that you can see it in the most simple way how human voice modulates the amplitude of the carrier.


Frequency domain
Lets take a look at spectral characteristics of the same example. We took human voice which was about 3 kHz wide and mixed it with a carrier whose frequency is 10 kHz. At modulator’s output we get two side bands which contain the same information, but are mirrored against each other. The mirror itself is the carrier frequency of 10 kHz. Those two side bands have names: upper and lower side band (USB and LSB). The upper sideband is the same human voice, but transposed to 10 kHz, while to lower sideband is the inverted human voice. More on inversion later.

Capture1Image 4 – Human voice

Capture2Image 5 – Amplitude modulation at 10 kHz

If you want to have fun with transposed human voices, all that is left now is to use sharp filter to remove the lower sideband. What you have on above image marked as USB is actually a human voice transposed to 10 kHz. To have more useable ”weird voices”, i recommend lower carrier frequencies, max 3 kHz, and you can get all sorts of Donald Duck and space voices.

Spectrum inversion
Ever wondered how would this song sound if you could invert it in frequency domain (tones that were low would now be high, and those tones that were high would now be low)? Well, if you understood the process of amplitude modulation, you can do it too. Here is short example: Take a song and apply a strong 8 kHz low pass filter. This is needed to put the song inside a limited frequency band to avoid aliasing problems later. Mix this song with 8 kHz carrier inside amplitude modulator. Now you got two copies of the song. One is in the range 0-8 kHz, and the other one is in the range 8-16 kHz. The first one is inverted in frequency domain, while the second one is the same as original, but transposed to 8 kHz. Now all that is left is to apply strong 8 kHz low pass filter to remove the upper sideband, and you got frequency inverted song.

Ring or Amplitude modulation?
Both names are correct, however if you need to choose the more appropriate one, it would be ring modulator. Because when you say ring modulator is exactly known what do you mean by that: An analog circuit made of diodes, which usually has a shape of the ring and multiplies two input signals.

Amplitude modulator does that too, but not always. Reason for this is there are few different types of amplitude modulators. For example in radio transmission techniques amplitude modulator does not only have sum and difference at the output, but also a carrier signal. This is the most common amplitude modulator in the world, and the whole AM radio broadcast is based on it. If used in our example, on image above there would be a large signal present at 10 kHz with amplitude about twice bigger than any of the sidebands. But as you can see, there is nothing at 10 kHz, because we used a pure ring modulator.

If we look at the math, amplitude modulation will give us only the sum and difference of input signals. Thus amplitude modulation is correct name too, but to avoid any confusion with radio broadcast technology it is better to use term ”ring modulator”. Hint: carrier was in a way a byproduct of early amplitude modulators, but turned out useful for broadcasters, and is used to drive an AGC (automatic gain circuit) in an old type AM radios, so that the signal doesn’t fade that much in the volume during various ever changing atmospheric conditions).

FM synthesis
In FM synthesis, one (or more) oscillator is used to modulate frequency of another one. Although both oscillators are using simple waveforms (like sine wave), result can be a sound with very complex harmonic structure. Usually one oscillator we call modulator, and the other one carrier.


As seen in two examples on the image above, the complexity of result wave always depends on the output level of the modulator (marked with red). If we increase the level of carrier, we are just increasing overall sound volume. In first example (left) modulator’s level is set to 0. Resulting tone is the same as carrier. In the second example (right) we increased modulator’s output level to 10, which resulted in a tone that is totally different from both modulator and carrier tones.

Using different levels for modulation we are creating different harmonic structures at the output. However this is not enough, because each instrument have a characteristic way in which it’s sound changes during the time. This is called the envelope. For example guitar begins loud and then gradually reduces its volume and harmonic content. On the other hand Hammond organ maintains the same volume and harmonic content as long as you are holding the key. As you can see, these two instruments have different envelopes. That’s the reason why FM synthesizers (like Yamaha DX7, SY-77 etc.) have envelopes on each oscillator. A package of oscillator + envelope is usually called an Operator. Operators can be arranged in many different ways called algorithms. What we described above is the simplest algorithm consisting of one modulator and one carrier.


2 To create more complex sounds, you need more than one operator. In that case you can have an operator that is modulating another operator, which is again modulating another operator that modulates the lowest operator (which is carrier), as seen on algorithm example 1. Or you can have three operators at the same time modulating one operator, as seen on algorithm example 2.


Simple FM modulation
Some analog synthesizers do have FM, however this is just a basic FM with a lot of restrictions. To do proper FM you need more than two oscillators, and each needs to have its own volume envelope, plus many other things that are too complex to be properly implemented in analog synthesizer. It would cost too much just to sound as DX-7, which you can buy for much less. However, if you are good in FM programming and have analogue synth which features FM, you can do some nice FM sounds (bells, metals, etc.). However to achieve a true DX sound you need PM or Phase Modulation. This is what drives all of the FM synthesis types of synths that we have on the market.

FM emulation
Even if your synthesizer does not have any kind of FM, but has enough fast LFO you can create a primitive kind of frequency modulation (that is modulation, not FM synthesis). Reason for this is that technically pitch modulation is the same thing as frequency modulation (FM). So with LFO you can create frequency-modulated sound. Keep in mind that this is all you can get, and this is far away from FM synthesis. Take the LFO and set it to high speed. Route it to modulate the pitch of oscillator. Now if possible, apply envelope to modulate the output of the LFO. If not possible, then use LFO Fade function. You need the Fade Out function. Its Purpose is to reduce the output level of LFO to zero after a short time. If LFO has a delay, you can set it to hold the LFO at maximum level, and then let Fade function fades it away. We talk about very short times here for the delay (50-200ms) and fade about 300-1000 ms. Experiment here. For oscillator (wave generator) waveform, choose sine wave. For LFO also choose a sine wave. Trigger high tones on the keyboard, and adjust the amount of LFO modulation that you are applying to pitch modulation until you are satisfied with the result. With enough fast LFO and good Fade function, you should be able to create a few nice bell sounds.

Granular synhesis
In granular synthesis samples are split in small pieces of around 1 to 100 milliseconds in length these small pieces are called grains. Multiple grains may be layered on top of each other all playing at different speed and volume. You can imagine it like some kind of wavetable synthesis, but here samples are played so short that you hear them as a timbre, not as a rhythm. By varying the waveform, envelope, duration and density many different sounds can be produced, not possible by any other synthesis type. There are many PC programs which do granular synthesis. One of the more famous ones is definitely Kaivo by Madrona Labs.


Linear arithmetic (LA) synthesis
Introduced by Roland D-50 model in mid 80’s. At that time biggest problem in sample playback was limited memory. If you could build a sample player with individual samples, it would cost enormously, because chip ROM sizes on the market were ridiculously low and expensive. Some observations on human hearing showed that most important thing in defining each sound unique to other was the attack transient of a sound. That is exactly how D-50 worked. It used short sampled attack transients and analog style oscillators for the sustained part of the sound. Short samples didn’t required big memory, which reduced the cost of the synth.


Today this kind of synthesis is probably no longer needed, however it still sounds unique. We can add that analog emulation part in D-50 is awesome making it really powerful and thick sounding digital poly synth.

PCM sample playback synthesis
Once analog signal gets converted into digital through sampling (digitizing) process, the result is called a sample. Pulse Code Modulation (PCM) is the coding technique used in this process. PCM is used in all digital instruments, and digital devices like PC, mobile phones, etc. Example of PCM could be ‘.wav’ and ‘.aif’ types of files on your PC. Sampling is a very simple process. You take the instrument, connect it to  soundcard input and use recording application that will digitize it, and turn it into PCM. The core of this process is happening in the soundcard inside analog to digital converter. The better converter you have, the better results. Four parameters define the quality of A/D converter. Sampling rate, bit depth, dynamic range and signal to noise ratio. Sample is then stored in the memory (RAM / hard disk).


A device that is capable of performing functions of sampling and storing is called a sampler. If a device can play back those samples at different pitches, we call it a sample playback synthesizer. About 90% of today’s synthesizers are of this kind and they all use subtractive synthesis method. Some samplers have a lot of advanced functions previously found only on synthesizers. Among most popular of them were the Emulator E4, Roland S-760, Akai S3000, and Yamaha’s A series.

Phase distortion synthesis
Phase distortion synthesis is a synthesis method introduced in 1984 by Casio in its CZ range of synthesizers, and similar to phase modulation synthesis in the sense that both methods dynamically change the harmonic content of a carrier waveform by application of another waveform (modulator) in the time domain. Casio introduced the term ‘phase distortion’.


From programmer’s point of view, what happens here is that every waveform has a range of distortion which when set at value 0 results in a pure sine wave, and when set at max, results in a waveform selected on the front panel (ie. a saw, square, etc). Multi stage envelopes can be used to sweep back and forth between these two extreme points, resulting in a timbre change. Essentially this is how phase distortion operates. Results are pretty unique though. There are some PD demos on this site in the Store area.

Physical modeling synthesis
As the power of DSP processors advanced, it was possible to do the synthesis of sound by using a set of equations and algorithms to simulate a physical source of a sound. This method mathematically models individual instruments and their parts, for example – metal string, a body of acoustic guitar, a pluck, etc. All this can be described by mathematical means.


First physical modeling synth was Yamaha VL-1. Later came out Korg Prophecy, and the Z1. Not many more since then, aside Kaivo and some others.

Realtime convolution and modulation synthesis (RCM)
Two synthesizers in the world use this kind of synthesis and those are Yamaha SY-77 (TG-77) and SY-99. This comes as a third type of synthesis they offer next to standard subtractive synthesis (AWM) and frequency modulation synthesis (FM). The name itself sounds complicated, but in reality the process is very simple. There are actually two configurations available.


In first one, you take the whole AWM element (waveform, pitch, filter, env) and insert it as modulator input for FM operator. That is, instead of simple sine wave as modulator, you use whole tone with its own waveform, applied filter and amp. This offers even more complicated FM synthesis.


In second configuration (image above) you can take the whole FM section and feed it into AWM section. That is, the sound that was created in FM section of synth becomes a ‘waveform’ that you process in AWM section. The AWM section is standard subtractive processing line. For example, if you apply a controller to modulate FM section, you can have ‘live’ and constantly changing waveform (marked as ”=” on image above) that is altering its timbre all the time. Of course, then you can apply a filter and envelopes of the AWM section to change the sound in more complex way. I know this all sound exotic, but it requires a lot of programming to do something good and useful actually.

Subtractive synthesis
This is the most common type of synthesis and is used on all analog and digital synthesizers and samplers. It starts with a sound that is sent to filter and then to amplifier. By doing this, you are subtracting some partials that existed in original sound, and you are changing sound’s envelope. This process is in-depth described in synthesizer basics article. Link here.

Vector synthesis
Introduced in 1985 by Chris Meyer, it was totally new concept in sound shaping. When asked about how did he invented it, Chris said: ‘One engineer was asking me to explain how various instruments performed crossfades. I had finished discussing the Fairlight, and had moved on the PPG – explaining its wavetables, and the ability for it to scan a group of waves first in one direction and then back again, While I was scrawling this back and forth motion in my notebook, suddenly a little twinge went off in the back of my head, and my hand drew the next line arcing down the page.. and the concept of crossfading between waves in two dimension, not just one, was born.’


The name of this synthesizer was Prophet VS. It was able to mix four waveforms via joystick and multistage envelope. Other vector type synthesizers included Yamaha SY-22, SY-35, TG-33 and Korg Wavestaion (which is more than just a vector synth). On Yamahas with a joystick you were mixing two FM elements with two sample elements.

Wave sequencing
First introduced by Korg Wavestation, this method offers (as it name says) wave sequencing. A wavesequence is a series of waves (samples), each with its own level, duration, crossfade time (to the next wave), and transpose. Wavesequences can be stepped through automatically or via various modulation sources.


When you set crossfade to low value, you get those characteristic ‘rhythmic’ sequences, which are a trademark for Korg Wavestation. Ensoniq TS series also feature wavesequencing.

Wavetable synthesis
Best examples would be PPG Wave, and Waldorf Wave / Microwave series. Their process of sound creation is based on wave sequencing through a waveform table. It is important to note that these waveforms are single cycled – they are very short. We can imagine them like the storage of the spectral energy of a single cycle snapshot. They are called ‘waves’. These waves can then be combined into lists called ‘wavetables’.


You can apply various controllers like envelopes, LFO’s to select the entry in the wavetable you want to play. It is also possible to interpolate between subsequent waveforms to make the timbral change happen more smooth if desired. Although waveforms are short, you have so many modulation possibilities that no other sampleplayer synth can match.

Synthesizer basics

The structure
Structure of almost all synthesizers is basically the same. It starts with the tone generator whose function is to create the sound. On analog synthesizers, this generator is a simple oscillator circuit, which can generate few basic waveforms like pulse, saw or sine wave. Therefore on analog synthesizers it is more common to use term oscillator than tone generator. On digital synths used terms are: wave gen, wave generator, tone generator, and sometimes, even an oscillator (though it is not a real oscillator inside).


Next comes the filter which defines the timbre of the sound and adds / removes harmonics from the original sound created in oscillator. Filter is followed by an amplifier, in which you set up volume change of the sound. Envelopes and LFO’s are used to manipulate various settings. For example, in oscillator they can control it’s pitch. In filter they can define filter changes over the time.


This was the basic description, however each synth can have it’s own and more complicated structure. Image above shows us the structure of Roland’s XV synthesizer. Each patch can contain up to four tones each with it’s own settings (sounds). Patch also contains common data, which consists of parameters that apply to all four tones like:  patch name, overall level, octave shift, key mode (mono/poly), portamento settings, bender range and more. Patch also contains modulation control (matrix control) in which you specify which controller will change which parameter – for example mod wheel on the keyboard to change amount of cutoff and resonance of the filter or the pitch of the wave generator (WG).

The purpose of oscillator is to produce a sound that you will later process with a filter and amp. Once you press the key on the keyboard, you “activate” the oscillator (in analogue synthesizer it is actually always on). Oscillator (OSC) is the starting point of any synthesizer. It is a place where the waveform is being created. In analog synthesizer, oscilator can be digitally (DCO) or voltage controlled (VCO) and it usually produces pulse or saw wave of which pulse’s width can be controlled and even modulated (PWM). All oscillators, no matter for what application, work in the same way. If we look at their heart will find it looks something like this.


To be described in the simplest way, oscillator is an amplifier and a filter that operate in a loop. The basis of operation is the tuned resonant circuit – for example LC circuit that is made of inductor (L) and capacitor (C). In this circuit voltage and current vary sinusoidally with time and are 90 degree out of phase. There are instants when the current is zero, so the energy stored in inductor is zero, but at the same time the voltage across the capacitor is at it’s peak, while all of the circuit’s energy is stored in the electric field between capacitor’s plates. There are also instants when the voltage is zero and the current is at a peak, with no energy in the capacitor. Then, all of the circuit’s energy is stored in the inductor’s magnetic field. As you can see, the energy stored in this electrical system is swinging between two forms. Unfortunately this ”swinging” won’t go forever due to circuit losses. Conductors have some resistance, as well as capacitor and inductor. That is the reason why we need amplification. The goal of this amplification is not to add some high gain, but just to compensate the losses we have in LC circuit. You can imagine oscillator like pendulum, which due to drag and friction loose it’s movement energy, so you need to kick it from time to time.


At the output of oscillator we can have various waveforms, depending on the type of oscillator that we built. For example if our oscillator is producing a sinusoidal wave, we call it: sinewave oscillator. There are many types of oscillators, most known are shown on image above, and those are (from left to right): sine, triangle, square and saw. Square is actually a pulse wave, with 50% width.

Wave generator
Digital synthesizers have a different kind of “oscillator” in their heart. It is not a classic oscillator that we just described, but a device that uses PCM waveforms. Pulse code modulation (PCM) is a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals, then quantized to a series of symbols in a binary code. If you have some wav files on your computer, those are the PCM waveforms, the same that are in your digital synthesizer. Advantage of digital wave generator is that it is not limited to basic waveforms (sine, saw, square, etc.) but can contain any kind of data that was previously digitized. You can have real piano samples, guitars, drums, etc. Of course if you want you can have basic waveforms too. Most digital synths have at least a sine and a saw wave sample inside their waveform memory – so you can create some analog sounding tones too.

Now you might ask, why one needs analog synth, when the same type of waves can be created with digital PCM wave generator. The answer was actually just said is in the previous sentience, it is a word PCM generator. Each time you trigger this type of generator, it will produce exactly the same sounding waveform. Result is uniform tone that stays the same. In contrast, analog synthesizers have imperfect oscillators, which result in various types of fluctuations that are different every time you hit the key. In a sense they sound unpredicted and different on each other key. But that is just the beginning of the story. Once you engage pulse width modulation, there is no digital PCM synth that can enter this area. Then comes the analog filter, which again adds its own character. In short, this is one of the reason why 30 year old synthesizers usually cost more than the latest “state of the art” PCM digital synthesizer with 9000 patches.


It is important to understand that we talk about two different worlds here. It would be pretty naive to blame some digital synth and call it poor because it can’t do some good analog sounds. If someone asks why a PCM synth can’t make a super thundering Moog bass sound, the simplest answer is – it was never designed to so.

The most dramatic change of your sound is taking place in the filter circuit. The richer the harmonic content on the waveform, the more the change. Example of rich harmonic waveforms would be square and saw wave. If your synthesizer does not happen to have resonant filter, you are actually missing one really important and charming aspect of a synthesizer.

If we look at frequency domain of a sound, for example square wave which is rich in harmonics and play a low note at say 20 Hz, we can see that these harmonic components are spread in the whole audio range (20 Hz to 20 kHz). What filter will do is to isolate some parts of this range in order to accentuate it’s other frequencies.

Types of filters
The most know filter is the low pass filter – LPF. It reduces the volume of all frequencies above the cutoff frequency. You can specify this cutoff frequency in filter settings. Once you cut out high frequency range, the sound will become more mellow.

Next type of filter widely known is a high pass filter – HPF. It is doing exactly opposite thing from LPF. It cuts the part of the spectrum which is below the cutoff frequency. It can be useful for percussive sounds (nice analog sounding hi-hats can be made with it). High pass filter can be resonant too, but actually there are not so many synthesizers that feature it. Korg MS-20 one of the rare analogue synths that features analog resonant high pass filter.

An one of the most specific sounding filters is probably the band pass filter – BPF. This filter leaves only the region in the vicinity of the cutoff frequency, and cuts the rest. With resonance setting you are actually shaping the width of this filter. The more resonance, the more narrow this filter will be. Image below shows frequency response of three basic filter types we just described.

Once you activate resonance on a low pass filter, new harmonics will pop up in the lower range, creating new sonic components that didn’t exist in original sound. High levels of resonance can produce self oscillation. Roland’s Juno manual has a very interesting and specific explanation of resonance (i find it really fun to read it today, but actually it is a good explanation):

“This control emphasizes the cutoff point set by cutoff frequency knob. As you raise the knob, certain harmonics are emphasized and the created sound will become more unusual, more electronic in the nature. If you alter the cutoff frequency while the resonance knob is set to a high level, you can create a type of sound that is attainable only from a synthesizer.” – Roland Juno 106 Manual

Filter poles / slopes
Sometimes you will read that a filter is 4-pole type. This is just another term for 24 dB filter slope, which is the most common filter type in the world of analogue synthesizers. Number of poles defines the sharpness of the filter. The more poles filter has, the sharper it’s frequency response will be. This also affects the resonance, since sharper filter results in more powerful sounding resonance. Roland TB-303 uses the not so common, 18dB filter slope. Now you might wonder, what this 18 dB means anyway. It is a unit which tells you how much a filter will block per octave. Which in this case is 18dB. If you put a filter cutoff point to 440 Hz, one octave above at 880 Hz signal will be attenuated for 18 dB (which is about 63 times).


If a filter is 4 pole, the signal will be attenuated 24 dB, which for 440 Hz cutoff point means that the signal on 880 Hz will be about 255 times weaker. Image above shows attenuation curves of four filter types: 6dB, 12 dB, 18dB and 24 dB.

Which filter should you use? The choice is up to you. If you prefer nice smooth filter sweep sounds, you should use 12 dB filter with a good resonance value. For thundering bass sounds or high resonance zaps you should use 24 dB filter (full resonance for zaps). While 6 dB filter is generally not used, it comes hand for sample playback, to gently remove high end from the harsh sounding samples, while not totally distorting the phase of the sample.

Amplifier and envelope
Last stage of sound manipulation that is taking place in the synthesizer is happening at the amplifier section. Its purpose is to control volume changes of the sound. On analog synthesizers amplifier is usually called VCA (if it is voltage controlled) or DCA (if it is digitally controlled). On digital synthesizers amplifier is usually called AMP or in case of Roland it is called TVA, which stands for time variant amplifier. Main part of the amplifier is the ADSR envelope.

ADSR envelope
This stands for: Attack, Decay, Sustain, Release and it represents four points. Once you hit the key, you are at the attack point. With Attack you are setting amount of time that it will take for sound to evolve from its starting level, to the point where you press the key. This is followed by Decay in which sound can evolve to another level that you set. This is followed by Sustain point. As long as you are holding the key pressed, you are at sustain point, and the level you set at sustain point will be the level of the sound during the time the key is held. Once you release the key, the sound will go off. To prevent the sound from going away to soon you can set the Release point. Release sets the amount of time it will take for sound to get to zero level, after you released the key.

There are envelopes with more than four points that we described, but they all work in the same way. Roland’s typical envelope consists of two decay levels. Some Yamaha’s SY series synths such as SY-77 and SY-99 have loop points that you can set to the envelope which is pretty cool feature.

LFO and control
The purpose of LFO is to alter various sound settings in back/forth cyclic manner. Usually LFO can apply change to oscillator’s pitch, filter cutoff frequency and amp level. If you apply LFO to the pitch, you will get vibrato, if you apply it to filter, you get sweeping sound (such as wah-wah), if you apply it to amp level, you get tremolo.

As it’s name implies, LFO is an Low Frequency Oscillator. Usually it consists of a few basic oscillator wave types like sine, saw or square. You can imagine LFO like a helping device, which is helping in way that you don’t have to manually change the pitch or level of the sound. Instead you program LFO to do that. However if you do wish to change some setting manually, then you need to specify them in controller section.

Audio example 1: Tone’s level modulation. Click here to hear original non-modulated sound. LFO was set to modulate amplifier (tone level). LFO’s waveform was sine wave. Result can be heard here.


The two images above show amplitude as a function of time (waveform display). As you can see original sound had constant level (amplitude). Once we applied LFO, level started to modulate from minimum to maximum value. Shape of the sine wave that was modulating original tone can be clearly seen on second image.

Audio example 2: Tone’s frequency modulation. Click here to hear original non-modulated sound. LFO was set to modulate oscillator’s pitch (frequency). LFO’s waveform was sine wave. Result can be heard here.

Image above shows frequency as a function of time (spectral display). It can be clearly seen how applied LFO modulates the pitch (frequency) of the oscillator. Shape of the sine wave that was modulating original tone can be clearly seen. Original tone was fixed frequency at 440 Hz. Once we applied the LFO, tone started to vary the pitch for about 50 Hz above and 50 Hz below original frequency.

Most of digital synths offer you to apply various controllers like mod wheel, aftertouch or velocity to some of the sound’s parameters like filter’s cutoff point or resonance, sound level, etc. Some synths call this feature modulation matrix. It goes something like this. First you specify the source of the controller – for example modulation wheel on the keyboard. Then you specify its destination, for example filter cutoff frequency. Now you set the amount, and you are ready to modulate the filter with modulation wheel. If filter settings on the sound are on the maximum open position, then you need to apply negative value to the controller, so that when you start to move the wheel, filter gets closed, for the amount you specified. Some better modulation matrix systems will allow you to apply almost any synth feature as a source to modulate it’s destination – for example LFO1 to modulate speed of LFO2 which can result in very complex and unpredicted results (this is in case the synth has two LFO’s). This is the area that requires a lot of experimenting, but results are always rewarding.

Abbreviations and common terms

Circuit bending is changing, removing or adding new electronic components within synthesizer to achieve different performance that is unavailable in original version. Usually cheap synths are being circuit bent to sound more wild, unpredicted, strange, or all together. Circuit bending results with warranty void, and can damage your synth permanently.

Legato is a function that should only work in monophonic mode. When Legato is on, pressing one key when another is already pressed causes the currently playing note’s pitch to change to that of the newly pressed key while continuing to sound. This can be effective when you wish to simulate performance techniques such as a guitarist’s hammering on and pulling off strings.

Modulation wheel affects the sound as specified by the control parameters (control matrix). On many synths it is set to vibrato by default.

Portamento is a function that causes the sound’s pitch to change smoothly from one note to the next note played. Portamento is common on guitar, violin and other string instruments. However, portamento is not possible on a fixed pitch instrument like the piano. On a synthesizer, parameter called ”portamento time” or ”portamento speed” defines the speed at which an oscillator moves to a new note you pressed on the keyboard. When the Key Assign Mode is mono, this can be effective in simulating performance techniques such as a violinist’s glissando.

Pitch bender (pitch wheel) bends pitch of the played note up or down, and is spring-loaded to return to center position.