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Akai S3000 vs S3000XL the ultimate test – which is better?

Image source: eBay

The Akai S3000 and S3000XL samplers, iconic fixtures in the music production landscape of the 1990s, revolutionized the way musicians and producers approached sampling and sound design. Renowned for their robust build, intuitive interface, and unparalleled sound quality, these samplers became staples in studios worldwide. The S3000, introduced in the early 90s, quickly garnered acclaim for its versatility and expansive feature set. The subsequent release of the S3000XL built on this foundation, offering enhanced functionality and greater ease of use. Together, these models not only defined an era of digital sampling but also influenced a generation of artists, shaping the sounds of genres ranging from hip-hop to electronic music.

Simple Test
However, despite their shared legacy, various myths have emerged online, with debates over whether one model is superior to the other in terms of sound quality. To address these claims, the most straightforward approach is to conduct direct comparisons. Audio enthusiast Tech44 from the Gearspace forum undertook this task, leveraging his high-level expertise and high-quality audio equipment to ensure accurate results. Tech44’s method was meticulous: instead of merely playing loops through each sampler, he loaded individual samples into both the S3000 and S3000XL and then sequenced them back via MIDI. This approach provided a real-world application scenario, allowing for a true comparison of their performance and sound reproduction capabilities. This article delves into the history, features, and lasting impact of the Akai S3000 and S3000XL, and presents Tech44’s findings to determine if there is a definitive answer to the debate over which sampler truly reigns supreme. A simple A vs B test was created from his recordings by splicing the two samplers back to back. For those curious or who want to analyze the data themselves here is an audio recording of one sampler followed by the other:

S3000 vs S3000XL.wav (24bit, 44kHz audio file in uncompressed wav format)

On the Gearspace forum in “All Akai’s are sexy thread” we posted this audio file above. There people were required to tell which part of the recording is S3000 and which is S3000XL. Out of 750 downloads a total of 0 correct answers were given. Let that sink in. 🙂 At this point we could simply end the article, because regardless of your listening environment I think it is clear you will be unable to tell which is which. However, let’s dig a little bit deeper, since we want the final truth.

Advanced Analytics
“But wait, they must sound different I’ve read it on the internet…”. Alright, folks. Since on the Gearspace forum we already determined that us humans can’t tell the difference between the S3000 and the S3000XL as nobody was able to pinpoint exactly which sampler was at certain point of the recording. Time to call in the big guns: computers! Inspired by someone’s astute observation that the S3000XL sounds like it was made out of Legos (it sounds plastic), I present to you our thrilling new saga.

Finding the plastic: Volume 1 The Quest Begins
First stop: the low end! If I were plastic, that’s where I’d hide in between 20-350 Hz. Let’s compare the S3000XL on the left and the S3000 on the right. Click the image for full size. Hmmm… the spectrograms look identical. Well, that’s a bummer.

Wait! If it sounds plastic, it must be in the highs. Let’s crank it up to -120dB, and go to high frequency, and I mean extremely high in 17000-22000 Hz where only dogs and bats dare to listen. Click the image for full size. Aaand… nothing. Spectrograms still look the same. S3000XL on the left and the S3000 on the right.

You know what. Now when I think of the plastic and frequencies involved in the sound. Of course, plastic would be in the midrange. How did I miss that? Let’s check it out. Left: S3000XL, right: S3000. But no, they look identical in the 1-5kHz range.

Finding the plastic: Volume 2 The Plot Thickens
And then it hit me, like in that movie, “The Usual Suspects.” When you can’t find the tiny details, zoom out and look at the big picture! So, let’s zoom out and search for plastic artifacts. Nope. Spectrograms are still twinsies. Left: S3000XL, right: S3000.

Hey plastic! Where are you hiding?
Just as I was about to throw in the towel, a lightbulb moment! If it’s not in the signal and in the things we hear, then it must be hiding deep in the noise. Right! Let’s amplify the range a gazillion times and inspect the background noise. Aha! The plastic has to be there. Somewhere! Elsewhere? Nowhere! Left: S3000XL, right: S3000. Spoiler alert: still nada.

The verdict
Despite rigorous tests showing no difference in sound between the Akai S3000 and S3000XL samplers, many users continue to believe that one sounds better than the other. This phenomenon can be attributed to cognitive bias, specifically confirmation bias and the placebo effect.

This occurs when people favor information that confirms their preexisting beliefs or values. If a user believes that older equipment, such as the S3000, must sound better due to its higher original cost or its vintage status, they are likely to interpret their listening experiences in a way that supports this belief. They might ignore or downplay evidence that contradicts their preconceived notions, such as blind tests showing no discernible difference in sound quality. And this takes a LOT of effort especially once the evidence starts pouring in.

When we say rigorous tests, we mean EXTREME rigorous in 24 bit domain which offers 140 dB of headroom for analytics where human ear is not even capable(!) to analyse anything to begin with. This is why he have computers. And they say: no difference whatsoever. And it makes sense as we will touch this subject in the chapters below as they are crucial in understanding the whole picture. One thing to remember from all of this:

Do not believe the myths coming from random nobodies on the internet. Instead, prioritize relying on empirically supported data that is open to personal analysis. Should you lack the necessary expertise to conduct such analysis, it remains prudent to refer to authoritative sources. This particular article presents the results of rigorous testing and includes access to the raw 24-bit data, enabling independent verification and analysis. This article was not written for those individuals with cognitive bias because they already made their mind and any attempt into changing it is a waste of time. The more evidence you present them, the more faster they change the goal posts. Instead of ending the article here, and especially if you are interested in buying any of these units, or it was in fact the reason you visited this article at the first place, we will include a few chapters that could help your decision and even include a few FX and resonant filter demos.

So, which model is for you?
One important consideration is RAM. If you plan to upgrade S3000 expect to pay $150 (May 2024) for 8 MB expansion. If you plan to upgrade S3000XL, expect to pay $15 for 16 MB expansion. This extreme difference (1:10 ratio) is due to the fact that S3000 uses proprietary RAM while S3000XL uses 72 pin SIMM which is cheap as the postage cost itself.

It should be noted that the S3000 comes with a built-in effects processor, while the S3000XL does not. Instead it requires an additional effects card, the EB16FX, which has become increasingly difficult to find these days. The S3000’s onboard effects include Echo, Chorus, and Pitch Shift. In contrast, the EB16FX board for the S3000XL offers a more extensive array of effects: Echo, Chorus, Reverb, Pitch Shift, Distortion, EQ, Ring Modulation, Flange, and Phaser. This means that an S3000XL equipped with the EB16FX board provides a much larger palette of effects. So if you are after a known artist for whom you know used S3000 and its internal FX, then go with the S3000. If you just want that S3000 sound for basic playback, then it doesn’t matter, you can pick either of the two. If you have EB16FX board in your MPC and looking to add a hardware sampler, then S3000XL would definitely be a better choice.

Not only does the EB16FX board expand the variety of available effects, but it also enhances the flexibility of their application. The EB16FX features two parallel multi effects blocks, each capable of handling any of the effects, plus two additional effects blocks dedicated to reverb. This configuration allows for a total of four parallel effects blocks, offering a significant advantage in terms of creative possibilities.

While this article does not delve into a comparison of the quality of these effects or the differences between the S3000 and S3000XL in this regard, we acknowledge that this is an area of interest for many users. For those curious about the sound of the EB16FX board, we will provide demo recordings in the addendum. These demos aim to showcase the capabilities of the EB16FX board, offering listeners a chance to hear the effects in action and form their own opinions.

Image source: mpchunter.com

Moore’s Law
There is a common concept that older equipment inherently sounds “better”, maybe it does, but when comparing the Akai S3000 and S3000XL samplers this is a flawed starting point. Because both models featured absolute state-of-the-art converters at the time of their release, representing the pinnacle of audio technology. Converter is not just one single IC but the entire circuitry dedicated for recreating digital signal in analogue domain. It is unfortunate that people focus on a single chip, rather than entire board that is delivering the actual waveform.

When the S3000i was introduced, it came with an internal drive and maximum memory capacity, and it could cost close to $10,000 with all expansions —a staggering price that reflected its cutting-edge capabilities and premium components. The period when these samplers were released was marked by significant breakthroughs in semiconductor technology, with Moore’s Law driving rapid advancements. This meant that just a few years later, the S3000XL was introduced at a much more affordable price point. The key difference was not in sound quality but in technological advancements and cost efficiencies. Unlike the S3000, the S3000XL no longer required expensive proprietary Akai memory. Instead, it used more affordable 72-pin SIMMs, which had become widely available and inexpensive due to the booming computer market.

Additionally, the S3000XL benefited from more powerful CPU resources, offering better performance for a fraction of the cost of the S3000. This drastic reduction in price, due to cheaper memory and improved semiconductor technology, often leads to confusion when people compare the original release prices of these two models. It’s important to note that despite the lower cost, the S3000XL was still a significant investment, unaffordable for an average school kid at the time.

Understanding this context is crucial for appreciating why the S3000XL could be sold at a lower price without compromising on quality. The advancements in technology allowed for more efficient production and reduced costs, making high-quality sampling more accessible to a broader range of users.

The history
When the Akai S3000 was released, it represented the ultimate goal for every serious music producer. Owning an S3000 signaled that you had truly “made it” in the industry. Its cutting-edge technology and superior sound quality set it apart from competitors, making it a coveted piece of equipment. Producers with an S3000 knew they had a significant advantage over those still using older models like the Roland W30, Akai S950, Roland S-550, or even the Akai S1000. There was simply no comparison to the superiority of the S3000 when considering the context of that era. Among other things it finally gave the Akai S series a resonant filter and we will actually demonstrate some of it on the later S3000XL model in the demos below (in the addendum of the article).

The prestige of the S3000 was such that people would go to great lengths to acquire one, including taking out loans. This sampler was more than just a piece of equipment; it was a serious investment in one’s music career. Owning an S3000 meant you were a serious player in the game, with tools that could outperform nearly anything else available at the time. Its powerful features and unparalleled capabilities made it indispensable for professional music production, and having one in your studio was a clear indication of your commitment to quality and innovation in music creation. A few years later it was replaced by S3000XL with the same impressive specs. First units of S3000XL were assembled in Japan, while the later were assembled in China. They all feature exact the same internal boards that were made in Japan. In the two images below we can see the inside of the unit that was assembled in Japan and below it the inside of the unit that was assembled in China. They both contain the same PCB’s which say made in Japan.

S3000XL assembled in Japan

S3000XL assembled in China (click to zoom)

 

ADDENDUM:

Akai EB16FX board – the demos with S3000XL
First a small disclaimer. Don’t let it fool you, but EB16FX can be a bit frustrating at times, since all of the effects are routed through the main output. Also while containing some very useful effects, you will find yourself using it at only 50% of its potential. Because the board is split into 4 effects groups that run all the time in parallel. So if you use effect A it will take the whole headroom and you are forced to mix it internally with effect B. As a result you start using only effect A, because both effects at once can turn into mud, while all 4 effects can create a lot of mess. With that being said it is possible to use them all at once. But in most situations the half of DSP chip will be twiddling thumbs – doing nothing.

In my opinion, what they should have done was to use ALL of the DSP power and gave us one multi effect plus one reverb. As a result the board would have twice more fidelity, more quality, dynamic range, bandwidth etc. If it only had one proper effect rather than 4 at a time and employ full DSP power for that effect, it would be excellent addition. Because algorithms are quite good. There is: distortion, EQ, ring mod, chorus, flange, phaser, pitch shift, pan mod, delay and reverb.

The board can look absurd at times, because in effects A and B you already have reverb option, while C and D are reverbs only. So you end up having a possible scenario where you were supposed to Mix 4 different reverbs, all internally in digital domain and all streaming that through one set of stereo outs. This is an early 90’s 16 bit device, and not your DAW where you can mix things with near infinite bit depth. But it is what it is. Let’s hear some of the effects in a few demos. Sorry these were made under 5 minutes, so don’t expect some hit music in here and pardon the mp3 audio quality. Sources of the four demos below: All internal synthesis (internal waveforms only), no external samples used (except for drums). Internal effects only (EB16 effects board).

Analogue Pad – demonstration of a wide EB16 chorus.
Ring Modulator Arpeggio – excellent ring modulator for Techno!
Bass and Arpeggio – demonstration of envelopes and resonant filter performance.
Saw Line Flanged/Delay – another one, but with cool flanged delay.

Extreme low (filter test):
Filter open/close – loop processing. I hope you have either large speakers or good headphones that go well below 30Hz. This is extreme low end, particularly towards the end of recording, yet full of power, earth shaking. Absolutely fantastic sounding filter. Hint: try this on other samplers. Only a few can do this kind of low end!

Well that kinda wraps it all up. I hope that you enjoyed the article. If you would like to add some extra infos that you think would be beneficial for the article feel free to comment below, we will gladly share it with the rest of the world! Or if you just want to share your anecdote, again feel free to comment!

Finding the tightest hardware MIDI sequencer among a dozen (measurement tests)

Hardware MIDI sequencers have a rich history rooted in the evolution of electronic music technology. MIDI, or Musical Instrument Digital Interface, was introduced in the early 1980s as a standardized protocol for electronic musical instruments to communicate with each other.

The first hardware MIDI sequencers emerged around the mid-1980s. Devices like the Roland MC-500 and Yamaha QX1 were among the pioneering standalone sequencers. These early models allowed musicians to record, edit, and playback sequences of MIDI data, enabling them to control multiple synthesizers and drum machines in synchrony.

Throughout the late 1980s and 1990s, the market saw significant advancements in MIDI sequencing technology. Companies like Roland, Yamaha, Korg, and others introduced sequencers with improved features such as more tracks, better editing capabilities, and enhanced integration with other MIDI devices.

In the late ’90s and early 2000s, hardware MIDI sequencers experienced a shift with the emergence of computer-based DAWs (Digital Audio Workstations). These software applications offered more comprehensive recording, editing, and mixing capabilities, challenging the dominance of standalone hardware sequencers.

However, hardware sequencers persisted, appealing to musicians seeking tactile interfaces and dedicated performance tools. Companies continued to innovate, releasing units like the Elektron Machinedrum and Octatrack, Akai MPC series, and newer versions of the Roland MC series, offering unique sequencing approaches, sampling capabilities, and real-time performance features.

Fast forward to the present day, hardware MIDI sequencers remain relevant in the music production landscape. They often integrate modern features such as touchscreen interfaces, advanced MIDI capabilities, CV/Gate outputs for analog gear, and innovative sequencing methods, catering to the preferences of various musicians, producers, and live performers.

The evolution of hardware MIDI sequencers showcases a journey from the early days of MIDI technology to the present, where they continue to carve out a niche by combining the hands-on approach of hardware with the power and flexibility of modern electronic music production. But which is the tightest midi sequencer? Let’s build some custom cables and run some measurement tests to find out!

In this article the following devices will be tested

Hardware sequencers:
Akai MPC2500
Kawai Q-80EX
Roland MC-500MkII
Yamaha QX3
Yamaha RS7000

Hardware keys synths and samplers featuring a sequencer:
Ensoniq ASR-10
Ensoniq ESQ-1
Ensoniq TS-10
E-MU Emulator 4 Ultra
Korg 01/W
Kurzweil K-2600 RS
Roland XP-50

Computers featuring sequencer software:
Amiga 500
Atari 1040 ST
Mac running OSX and Windows 10 using RME UCXII and MPC Renaissance

MIDI Jitter
MIDI jitter refers to variations or deviations in the timing or regularity of digital signals. In the context of MIDI jitter can disrupt the accurate reproduction of the original signal due to timing inconsistencies. In data transmission, it can affect the timing of bits being sent across a channel. Jitter can arise from multiple sources. It might occur due to imperfections in the clock signal, signal interference, signal reflections, noise in the transmission medium and limitations in the precision of the components or the timing of the system.

In case of computers we need a perfect and dedicated USB bus that will not be interrupted by other protocols and services, which at some systems can be a difficult task to do (i.e. Windows 10 based computers). If the same USB port is being interrupted for some random reason every few seconds it will definitely have less “space to breathe”. Which brings us to the last parameter and these are drivers. So even if we are limited to i.e. the Win10 system a set of good drivers can improve things a bit. Looking at the graph with the results some might notice that Akai Renaissance is missing in the graph. The reason for that is, it would simply not fit, or if it would fit, the rest of the graph would be hard to read. This is a clear example of good vs bad drivers, in this case the RME vs Akai.

Measurements
All of the measurements were performed on a free software called MLA – MIDI Latency Analyser v2.1.1. It’s a piece of software to help us determine the effects that hardware, driver and software changes have upon MIDI latency and jitter. The program is also handy for identifying the ideal number of samples of offset to apply to MIDI tracks to compensate for round-trip latency when recording them to audio tracks. I am not the author of this software not related in any way to it, therefore I can not provide any sort of technical assistance regarding this software. A special hardware cable is required to be built before using it. If you decide to join the research, all of the details can be found on this address: http://tinyurl.com/midijitter

Offtopic: Some extra scores and some explanations
Yes, many of the keyboards feature hardware sequencers. This is why I included many of the hardware keyboards / synths into the measurement. I believe Ensoniq ESQ-1 was one of the first ones with a decent sequencer. It’s a pity it does not have some more features like setting the fixed velocity onto the recorded data or modifying the gate times. This is also the reason an additional (Features) column was included in the results table. Point is, if just because some sequencer scored high, does not mean you should try to grab it immediately, then send me 50 emails cursing me why didn’t I tell you the sequencer has nothing useful inside. This is the reason a scoring column called Features was added and it works on the following principle (how the scores are added). Please note the Features score column DOES NOT in any way relate to the MIDI jitter measurement results. If you’re curious here’s how the Features scoring column works: 

20% score = bare minimum Record, Play, Transpose, Quantize, Copy, Paste
+ 20% for advanced MIDI editing, change velocity, gate, note editing ranges
+ 20% for step recording
+ 20% for microscope edit
+ 20% for XoX style edit

So a sequencer that scored 100% has all of the above. Again this is just to make your potential shopping list easier, has nothing to do with the MIDI jitter results. And yes some of the rack synths and samplers have sequencers too! So they are included as well.

Atari vs Amiga – the final battle, which is better for MIDI?

To answer another potential question: Why including computers in the article titled hardware sequencers? As a reference point. Nothing more. Sort of to see where you stand if you run any of the computer + audio interface combos mentioned in this test. We will also have a privilege to see the battle of two 16 bit legends, the Atari 1040 and Amiga 500. For this test Amiga was running the M.E.D. software tracker, while Atari was running the Cubase 3.0.

Results
Before looking at the graph and the table one thing to keep in mind, the smaller the number, the better the result. Ideal number in this case is 0, but we didn’t test Expert Sleepers in here, so above 0 it is. First and top of the table we have the incredible Ensoniq TS10’s. These results and numbers are ridiculous, I agree. I repeated the test several times thinking I did something wrong. Even re-soldering the cables. The number are correct, TS10 has incredibly precise sequencer. On the second place Emulator 4 stored pretty impressive, but unfortunately has a very limited sequencer, so beware. As expected the Atari as a rock solid MIDI standard still stands well, so nothing special required to be said about it. Roland XP-50’s powerful 32bit RISC processor clearly shows up, with results even slightly better than Atari if we include the Max jitter (the max amount a note will deviate from the mean value). Another interesting “battle” of the grooveboxes, Yamaha RS7000 vs MPC2500. Yamaha came out far superior. Or did it? Check out the next chapter titled Individual MIDI hardware outputs vs Jitter as the things are not always as simple as 1+1.

Continuing with the graph we see the regular Mac computer running OSX Sierra connected to a RME UCXII. The results are essentially identical to the Atari. Something that can not be said for the same computer running Windows 10. While the average jitter results are fine, in the sub 1ms range, for some reason a few of the MIDI notes will jump as far as 3ms to the front or back. Don’t worry an average listener won’t hear it, in fact no one will, however if you layer percussive sounds on top of each other then a transient jumping back and forth 3ms (6ms in total) can be very annoying at times. Now keep in mind these are RME drivers (probably the best in the world!). But to see how bad things can go with Windows 10, see the entry in the table that says Akai Renaissance (hint: it’s on the bottom). This is an example why Mac dominated the DAW all of these years, at least for people who run external gear. With MPC Renaissance having plus minus 8, that’s 16 mili-seconds combined, that’s something even a non musical person can hear, say you lay down a pattern of 1/16th hats, this kind of deviation is way too easy not to miss. So yeah, Windows 10 and external MIDI gear, not my first recommendation, or if you have to, go RME interface. I should point out MPC Renaissance was tested only as a MIDI output interface, not as a software per se, and was running Reaper DAW for the test.

I know, you can’t see a thing. Please click on the graph to enlarge it.

Continuing with the graph we see the Korg 01/W which has an excellent sequencer (actually I tested the 01/RW), packed with features almost as much as Roland XP-50 (the later is slightly superior as it has RPS realtime phrase feature which speeds up things quite a bit). 01/W is closely followed by Yamaha QX3, a super complicated sequencer, at least for me who never worked on it before, so it can be very confusing. It looks cool though and is super tight. Next surprise was ASR-10 – it’s literally on a level of QX-3 and QX is a dedicated hardware MIDI sequencer just for that. I was quite surprised as I remember having some reserved thoughts for its song mode so I tested it as well. And I was right, after measuring ASR’s sequencer in song mode, the performance unfortunately drops. I didn’t want to include the data in the table, because most of the people use it in regular pattern mode. For those interested, in song mode ASR-10 is 0.347ms average jitter and 2.7ms max jitter putting it just slightly shy of MPC2500.

Next on the graph we have the Amiga along with Atari, a cult 16 bit machine that was most of the time used for “tracker” music but had a MIDI option using the serial port interface. The results are solid, but I never expected them to be stellar as Amiga has a set of many chips inside that require a lot of coordination – it was designed as a multimedia system. And this is where the sub 1 millisecond range ends and we are entering past 1ms area starting with Akai MPC2500 and Kurzweil K2600. I was actually surprised to see Kurzweil in here, I was expecting it near the top as Kurzweil is known for its “best of everything” approach. Followed by Ensoniq ESQ-1, and Kawai Q-80EX. Last but not least of the hardware sequencers listed in here came the Roland MC500 MkII. Espen Kraft has a cool video on YT check it out. It will throw away a note or two as far as 2.5ms, but for the 80’s soundtrack scores, it will do just fine. There is something magical about those tactile switches and the fact everything is there at a press of a finger, although it can go deep in microscope edit, hence good marks on the scoring table. The graph ends here, and is missing the MPC Renaissance for the reason already mentioned.

Individual MIDI hardware outputs vs Jitter
Akai MPC2500 has 4 MIDI outputs while Ensoniq TS10 has single MIDI output. So if you want to run 4 external devices with the MPC2500 you will still get results that are shown in the table, which is something that can not be said for a TS10 despite being far superior. Speaking about MIDI chaining, first of all, each additional device in the MIDI chain will add 1ms of delay, some might add even more and some might add totally useless data to their MIDI thru port. For example if you have a Yamaha TG-33 never place it as the first device in the chain, it will make the rest of your day pretty miserable. To avoid MIDI chaining problems you will need a MIDI patchbay. But then keep in mind the second part, which is that all of the output data still has to pass through one single MIDI port of our main sequencer on TS10. While 31.25 kilobits-per-second (Kbps) seems enough for a couple of MIDI notes, the moment you start sending control CC messages for several external moduiles you will soon reach the bottleneck of your MIDI interface. This is why a MIDI device with 4 hardware outputs, will in many cases or always be superior to a single MIDI port connected to a patchbay. My point: don’t dismiss the MPS2500 because of its position in the table, or think that TS10 will solve all your sequencing needs just because it is first on the list. Increasing the number devices chained to the single MIDI output will increase the MIDI jitter related issues, as the data will be more and more packed where there will be no more space left, and jitter will literally take over at one point.

The comment section welcomes any extra infos, anecdotes and stories related to this subject. So feel free to comment!

Demos of a few dozen hardware Reverbs

Far from any scientific or “professional” test, this is just a quick bunch of demos when a reverb is pushed a bit harder, say into the 10 second decay time and only around* -6dB below main track. Don’t use it as a reference because results will vary depending on the recording levels at – take it with a grain of salt. All tracks encoded to FLAC (lossless format). Feel free to share if you find it useful.

Regarding the Akais, both of my units are expanded with their respective FX boards (it is not the same FX board as they are many years apart) and they can be used as a regular effects processors, while Kurzweil has a sampling board which again turns it into an external effects processor (it can do way way more than just a reverb, think of it as Eventide’s little brother). I no longer remember why I recorded two reverbs from PCM-70. I guess I wanted to display it’s less chorusy side of things and more closer to the rest of the bunch.

*yeah, some reverb tails might be a bit off. I actually mixed everything on an analogue mixer during the period of a few days, so probably some are louder than others. Sorry about that!

Bonus:

Roland JD-990 revolutionary concept two decades later

It’s been 22 years since its release and the unit is still going strong. Even the latest in Roland’s series, Integra is in fact based on the DNA of the legendary JD-990. But let’s get back in the the 90s for now and see how it all happened. First of all, it should be noted that when model 990 came out, market (at least the Roland’s portion of it) was already dominated by the JD-800, D-70 and the quite popular JV-80 which came year earlier in 1992. Being there at that time i can tell you we considered JD-800 as a Rolls Royce, way out of our reach, but JV-80 was definitely second best, and it sounded incredibly good (for that time) with nice juicy digital resonant filters, for all of us who were into electronic music back then. Just a year later 990 came out. But for some reason it went under the radar for the most of us. It was mostly taken by studios and the pro’s. Little did we know how powerful this synth was (internet was very limited back then, there was no such thing as Gearslutz) while reading some brochure isn’t going to tell you much about it neither.

In fact it wasn’t until JV-1080 which came year later that everyone turned their heads into the direction of Roland with a big wow(!) all over their face. For start it was one of the first synths with 64 polyphony and we were all ready to sell our kidneys to obtain one. The JD-990 came out in kinda unlucky moment, jammed in between JD-800 release in 1991 and JV-1080 released in 1994 it somehow remained unnoticed by many. While those who have heard of it assumed it was just a module version of the JD-800. And because in 1994 we already had 1080 on the market, there was nothing to think about – go for the JV-1080! (myself included)

But there was another group of people, like film composers, producers, you know people who can recognize certain things such as good audio quality right away. Many of them kept their JD-990 despite JV-1080 being the star of the show with all the spotlights focused when it came out. The reason was simple: JD-990 was soundwise class above 1080 and those who had it didn’t want to let it go that easy. Technical differences between 990 and 1080 are explained here. Unfortunately JD-990 wasn’t perfect and had some drawbacks compared to JV-1080 which is why some studios and composers decided to use both (such as for example Vangelis did). Drawbacks are obvious smaller waveform ROM content compared to 1080 ROM, and nasty digital distortion at higher resonance values. This is why in model 1080 Roland implemented variable waveform gain to avoid such scenarios happen ever again (unless you intentionally want them).


Image copyright: Cloudschatze

It is a know fact that Super JV’s architecture is based on the JD-990. On top of that it comes with the same type of aggressive filter (as opposed to other JV series like JV-80, JV-90 and JV-1000 which had mellower filter resonance) and it came with the now legendary “structures”. When the XV series came out, Roland continued the DNA line of the Super JV. The only thing “different” was that its SYX patch data was based on JV line of synths. Hence why XV can normally load Super JV patches via sysex, yet is unable to load the JD-990 patches. Then the Fantom came out, which again is based on the same concept. And now we have the Integra, which continues where XV series left, maintaining full compatibility and architecture of the Super JV. All of these synths are based on a revolutionary concept of a JD-990 synthesizer. And this is what makes it so special, along with its wide lush sound.

While a lot of extras were added with XV and Intergra series, the core structure is based on what was originally designed in a JD-990 synthesizer. Some may ask, ok but where is the JD-800 in all this? Well, the truth is, it has nothing much to to do with JD-990. It can be described as a D-70 with sliders on it with only difference of having a new effects processor and tones which are now part of a patch, and no longer separate thing. But when it comes to 990 these two synths have practically “nothing in common” except the first 108 waveforms and the mentioned JD effects processor which consists of two blocks. I will now list all the additions that 990 gave over the model 800:

  • Wave ROM was expanded to 6MB (vs 4MB on JD-800) with 195 PCM waveforms (vs 108 on JD-800).
  • Pan inside each Tone was added. On the JD-800 you can NOT pan individual tone for wide stereo sounds.
  • Matrix modulation was added on the JD-990. Let’s explain this – on JD-800 you can’t: increase cutoff point of Tone 1, decrease cutoff of Tone 2, increase resonance of Tone 3, decrease pitch of Tone 4 – all at the same time by moving the modulation wheel.
  • Multiple sources for the same destination added. On JD-990 you can for example use two different LFOs for the same destination – i.e. pitch, filter, TVA. This can create complex modulations. On JD-800 you can only use one source for the same destination.
  • SR-JV expansion card support. Next to standard JD-800 series card slots, there is additional card slot to use a 8MB expansion boards from SR-JV series. (i.e. Vintage Expansion, Orchestral, SFX, etc.)
  • Roland JV-80 patch import.
  • The LFO section has additional waveforms: sine, trapezoid and chaos.
  • Osc Sync function was added. It lets you synchronise two oscillators – a feature found in many analogue synthesisers.
  • FXM was added (Frequency Cross Modulation) – again found in some analogue synthesisers. It has 8 positions (labelled Color) that actually control the frequency of the modulating signal, and a depth setting 0 – 100 that controls the amplitude of the modulating signal.
  • Ring Modulation, for creating all kind of metallic percussion and strange effects.
  • This synth features 6 types of ”structures” which among many other things let you stack two filters in series, for building complex filter textures.
  • 24 dB filter (using structures), next to standard 12 dB which is always available.
  • Outputs increased to 8 total.
  • Polyphonic portamento.
  • Tempo MIDI sync delay.
  • Analog Feel. Which adds a random modulation to the sound to recreate an analogue synth’s “drift”. Not just on the pitch, but varying amplitude also.

Some people asked me about my opinion on JD-800 so I will give it just a brief info on why I initially didn’t became an owner but later I did. I’ve played JD-800, really liked the UI, thought of buying it, but got distracted by the fact there’s no even basic modulation matrix i.e. you can’t set modulation wheel on JD-800 to open the filter and apply some resonance. In fact there’s no even mod wheel on JD-800. Also, all the modulation buses on JD-800 are pre routed. You can’t use two or more sources to modulate the same destination (for complex modular style modulations). On top of that JD-800 is monaural per patch, there’s no pan feature per individual tone – so you can forget wide pads evolving through stereo field, which is a signature of many JD-990 sounds. However as time went by I eventually picked JD-800 as a sort of experiment. The idea was to record a sequence then to use sliders to modify some parameters of  the Tone 1 and record it into the DAW. Then to overdub the sequence but now to modify parameters of Tone 2. Important thing to note here: all of these parameters from JD-800’s front panel are being sent and recorded as MIDI into a DAW. These can later be fine tuned using editing tools on the computer and then finally data thinned before being sent back. This process can be repeated almost indefinitely with MIDI’s own maximal bandwidth (31kbps) being the only bottle neck in here. Using this method you can create very expressive soundscapes where you literally control every parameter in real time. In fact you can overdub every single parameter of every single tone realtime, although at this stage I think the MIDI notes would become quite sluggish due to before mentioned data bottleneck. This fun experiment made me started to like the JD-800. The second thing that I liked  was the fact, with that nice UI, it kinda makes you want to tweak it all the time. All in all I am now a happy JD-800 owner.

Ensoniq TS-10 wavetable and wavesequencing monster

9403

If i would have to choose one ROM-pler to hit the category mysterious, it would definitely be TS-10. First of all i never understood why such high second hand market price (particularly in States). You would think it is because from the impressive synthesis capabilities of having both the wavetable and wavesequencing synthesis in one machine. But i am 99% sure that is not the reason. Even today (writing this in dec/2015) and good condition TS-10 unit can set you back over $1000 USD. Which is in a way funny because in Europe you can obtain it for around 400 notes or ever less if you look long enough. Unfortunately i don’t know the secret connection of the TS-10 and US, if someone does, feel free to add a comment. Personally I suspect the secret is: 1)polyphonic aftertouch; 2)session gig players who got used to it; 3) excellent build quality; 4) excellent sequencer (again gig players territory)

With TS series, Ensoniq continued their line of transwave synths, this time introducing the sample playback in the synth engine. The first thing user would check when exploring waveform content are the transwaves. And unfortunately all those good transwaves from SD-1 are gone. In fact, this synth has a weakest set of transwaves, of all Ensoniq’s transwave series. There is total of just 8 of them. But the worst thing is, they all sound almost the same. So, on the first sight it appears this is no good synth for transwave fun, right? Well…. wrong! We got some good news.

Sample playback in TS series is not just ‘basic playback’, but it also features transwave synthesis. If you load a transwave into TS-10, you can change its properties from the basic waveform into the transwave. Now all that is left is to route a controller (LFO, env, mod wheel, etc.) on to it and your transwave is ready for fun. And we got some more good news.

Since transwave synthesis requires extreme playback precision the same can be applied for basic samples (non transwaves). You can for example use extreme short loop points, and route sample end position to mod wheel. As you move the mod wheel, new harmonics are being generated. This works best on short, white noise samples. Or instead of mod wheel you can use random LFO for some really unique effects.

Another good feature this synth has, is that you can shift the loop point and ‘browse’ through various regions of your sample. This works best on complex samples, made from small snippets, vocals for example (connected in series) merged into one large sample. Route LFO or mod wheel and you got some of the craziest vocals at the output. Believe it or not, but even some high-end professional samplers do not have this kind of loop shift feature. Now you might ask – is this all we can do with it? What would happened if we would have one sample made of 64 or 128 small short (pure waveform) samples, connected in series and then we would apply a loop shift feature onto it? Ever heard of synths such as PPG or Waldorf Microwave? Well, that is exactly what they do! Welcome to the…

Wavetable synthesis. Although not from the default state available on TS series, is possible, once you build a wavetable. Technically speaking, TS-10/12 does feature wavetable synthesis, but unfortunately there is no Ensoniq software for creating custom wavetables so one would need to make it ‘manually’ with standard waveform editing software. Considering there are total of 128 waveforms, this can be a big work. Also, every cycle must begin and end at zero amplitude. This ensures smooth playback of each individual frame, since any amplitude difference between start and end point at such short loops alters the harmonic content or totally shifts it into wrong pitch. However, once you build it, the result can be quite impressive. In fact it is possible to gain much higher quality (longer cycle waves, more hi-fi sounding) than on a standard wavetable synthesizers. This is because a single wavetable on TS can be as big as RAM size in it. For example 1 MB wavetable contains a frame with a size of 8 kB. In the days of PPG, 8 kB was the size of the whole waveform ROM!

ts10

Some might ask how come this synth has Wavetable synthesis, yet its specs or manual don’t say anything about it – they only mention Transwaves. Well, transwaves are similar to wavetables, in many aspects identical, exept there is no interpolation calculation between to adjacent frames (waves). Single transwave is made out of 128 individual single cycle waveforms and no calculation occurs in between. In other words, what you put into is what comes out (aka garbage in – garbage out). You can’t smooth it out or change in any other way. This is what makes it different from a wavetable, along with the way the data is stored and calculated on wavetable synths. Typical stock transwave is usually made out of two major waveform frames, the first and last cycle in the transwave. However, if you have editing skills and a desire you can build any transwave you imagine, which puts this machine in the vicinity Waldorf wavetable synths and their cool wavetable banks. Unfortunately Ensoniq never provided anything remotely interesting as Waldorf’s wavetables which is probably the reason why wavetable synthesis never took off on the TS series. Kinda pity. Even the Waveboy disks and their custom wavetables aren’t much impressive (i bought them all and regretted). Still if you have patience, once you build a set of good custom transwaves, you’re in the business! And just when you though, this synth has so many cool features, we come to another chapter…

Wavesequencing – just like on the famous Korg Wavestation. Although called Hyperwave, it is basically the same thing. Offering the same methods and similar settings it has one additional and quite useful feature called crossfade volume point. As you might know, a volume loss naturally occurs in the center of a linear crossfade point and with this feature you can completely compensate it. That’s why TS produces constant volume wavesequences, making them completely undetectable – almost sounding like some kind of a morph. Of course, you can always set it to 0 dB to achieve the classic Wavestation-like wavesequence with volume loss.

Custom Transwaves (detailed procedure)
Lets now go back co custom transwaves mentioned at the beginning of the article in case you decide to put them into the TS you might encounter some problems. For example: if you want to add another layer or duplicate existing one. Once you load the sample, you can’t – for some reason. So you must do it prior to loading.Recently i found a way to transfer multi layer samples to the synth. Lets say you build few transwaves in the PC and you want to put them in TS-10 via EPS disk software. No problem, you save the sample, load it to TS and start to program it. But there is a problem. You want to add another Layer (to place the same sample there, but with different parameters for thicker sound) – TS-10 wont let you do that. So i found some really old prehistoric program called Ensoniq MIDI Disk Tools. This program requires Win98 OS, but can run on Virtual PC (Microsoft’s PC emulator for WinXP and Win7).It is a Demo version, but for some reason it will do exactly what you need (in fact, this program is for something completely different). With it, you can make a copy of existing layer and create another one (this is just a copy, so total size won’t increase!), or you can put another transwave in another layer (useful for Ensoniq Fizmo type of sounds). The trick is that this program operates directly on file. So it doesn’t matter if this is demo version, for what you need this program, will be already done even before you click exit.

Here is a procedure on how to create custom transwaves (works on ASR-10 too). This requires commercial program called Awave Studio, but if you are musician you probably already have this program as it can do 1000 other things when it comes to sample conversion:

  • First, to create a transwave use Tranzilon – nice and simple prog.
  • Then convert .wav to ensoniq .efe file via Awave Studio program.
  • Then create ASR-10 floppy with EPSDisk.exe and save this .efe file it to disk. Done!
  • In case you want multi layer, then before you use EPSDisk start that Ensoniq MIDI Disk Tools program (described above) and add 3-4 layers (or just use copy if you want the same wave, so the waveform stays in layer 1 and you don’t get unnecessary large file size). Remember, this program operates directly on file, there is no undo. So make a copy of whatever you do.

Modulation
We need something to modulate all those transwaves, wavetables, etc. right? When it comes to modulations, TS offers one good feature called: modulation mixer. This is very similar to Kurzweil equation FUN’s where you can combine two controllers, apply scale and shape to one of them and get new controller at the output. With modulation mixer you can create really incredible modulators, some of them possible only on complex modular systems. Here is more info about it (from the manual):

Here are the available shapes:

Some examples (but possibilities are endless):

A couple of my patches
Originally i had idea to build a larger demo, but instead decided to build a soundset for TS-10 first, then do the proper youtube demo. However, not to leave you empty handed i found a couple of wavesequencing atmospheric demos that i did for the legendary deepsynthesis.net web side (also known as Sealed’s Deep Synthesis for those of who still remember it!!). These are no ordinary sounds but mostly long evolving textures, demonstrating the Hyperwave function.






Regarding the soundset
I planned other demos but decided to put them in the Youtube video once the soundset is completed. Please don’t ask me when that will happen, though. It will be available on this same website. If it isn’t available then it means it hasn’t been made – plain and simple. 😉

And here is an excellent demo by thekyotoconnection that i found on a YouTube with TS-10 doing Hyperwaves and Pads. There is even a link below the video (on youtube) that provides access to the patches in that demo.

Which OS version to go for?
Stock version is in 90% cases 2.20. There has been some talk about OS3.10 being the better (latest) version, however from what I have heard one can not just swap the old EPROM chips. Some modification to the motherboard is needed. As soon as I find out what needs to be done I will publish it in here. Currently I have OS2.2. I don’t remember any specific bugs here and there, on this version of the system. Feel free to discuss below.

UPDATE: I have bought OS3.1 from here. Will have it installed next week. Fingers crossed everything goes well. I am also curious to check the sequencer MIDI jitter response after the upgrade.

Yamaha A4000/A5000 Review

a5000

When its older brother Yamaha A3000 came out in the mid 90’s it was one of the more affordable samplers on the market that featured a full synthesis engine (sampler + synth). A lot of us lusted after one, since the big E-MUs and Akais were prohibitively expensive back then, this was the second best. Still they were not cheap. In fact by the time i could afford a Yamaha A series sampler it was already year of 2002. And by the time models A4000 and A5000 were released. So i purchased one – my first real hardware sampler. It was like living a dream “wow now i finally have a sampler!”. By that time i sold my beloved XP-50 (which was my only synth) and somehow missed it. What surprised me by Yamaha was that i could easy cover all of the sounds that i liked on the good old Super JV series. The filters were full and juicy, not thin like their earlier incarnations in SY series. The machine offered a lot of different filter types as well as all the modulation routings you would require for one serious “synth style” patch. As i’ve metioned, at that time i really missed my XP-50 so first thing to do was to dig into machine and go for the basic synth patches consisting of nothing more than a raw saw or square or sine wave. Coincidently most of the patches that i’ve made there were synth style and this is exactly what i will show on this page in a form of synth demos.

You will notice that the machine sounds quite nice, yet for some reason it is quite cheap on second hand market. Well better to clear some things right away!! Read this as a warning if you plan to purchase one of these devices. Unfortunately all of the A series Yamaha samplers suffer from the same problem, rotary encoders located below main display. After some time they start to produce weird results outputting wrong values or going into wrong direction – ie you scroll down, value goes up. I should point out that i bought my A4000 brand new from the store. And after one year in smoke free studio one encoder started acting. Quick browse to the web to find out other people have the same problem. On of the cause for this might be the fan in the back the sucks the air out of the unit, which means air will be sucked in from the front side – that is from the place where encoders are, and the dust will literally be pushed into the encoder. Even if you open it up, which isn’t an easy task to clean it up, the problem might return. You might want to google Yamaha A4000 encoders and see of the solutions that people came with. Or you can buy a new set of encoders. Unfortunately this might set you back around 50 EUR which is frustrating given the unit itself can be found for less than 100 EUR.

Second warning i have to give out to potential buyer. If you plan to use this as a live unit, simply forget it. The loading times on this unit are extremely slow. Even if you load it before the concert, if power shots down for some reason you have to go thru it all again and lose a lot of precious time. Not to mention the frustration if you have several different things to load, you would have to wait several minutes in between.

Back to the features. Although called a sampler, Yamaha A-4000 offers amazing synthesis capabilities and is maybe one of the most underrated samplers. when it comes to synthesis power. For start, it gives you: 2 LFO’s, 3 EG’s, 16 different filter types with complete control over the parameters. Single patch has no layering tone limit. Many samplers had layer tone limits of one patch to contain up to four tones (waveforms) such as Akais and some Rolands. Yamaha is not limited in that way. You can stack as many tones/waveforms as you want for a single patch (well as much as polyphony allows you to). This way you can create very rich and complex sounds.

It also has some unique synthesis functions like ”expand detune / dephase” and LFO with fully programmable waveforms. Once you create new sound, you can resample it to create even more complex one. There are also 96 onboard effects inside of three (six in case of A5000) independent effect blocks, which can be connected in series, parallel or individual. Combine this capability with the resampling function and you’ll have virtually endless effect processing power at your fingertips. You can even use the A4000/A5000 as a stand alone effects processor by assigning effects to the stereo analog inputs. Here are .mp3 sound examples of Yamaha A-4000 which will demonstrate it’s synthesis capabilities. In most examples, used sample will be the saw waveform. The size of this sample is only 0.7 kB or 712 bytes to be exact!

2osc_saw_lfo.mp3 (375kB) – First we will start with standard LFO modulated filter sweep. In this case used filter was 18 dB with medium resonance applied. Sound is made of two saw oscillators octave transposed and a touch of reverb.
2osc_saw_lfo-Note.mp3 (171 kB) – Same sound, single note.
AnalogRAW.mp3 (591kB) – Analog sound made of two saw waves and one square wave filtered through band pass filter (BPF) with max width. Each osc uses random pan, square uses expand detune function.
2saw_eg_lfo.mp3 (795kB) – Another LFO sweep with two oscillators, filtered one through LPF the other through HPF, both resonant. To demonstrate effects unit, a little bit of chorus and hall efx were added.
2saw_osc+dist+delay.mp3 (273kB) – Dual saw sound, processed through TWah+OD and T-XDly efx.
2sqr_lpf18(dephase)_hpf24.mp3 (478kB) – Introducing dephase function. This will make your sound extra wide without need for efx. Two oscillators were used – both square waves. One filtered through 18 dB low pass filter (LPF), the other one trough 24 dB high pass filter (HPF). I recommend headhones for this one.
AditiveRAW.mp3 (127kB) – Remember stacking as many tones as you want on a single patch? Well this feature gives you a chance to build a primitive, but functioning additive synthesizer. Following sound was created by only using sine waveform. There are totally 8 sine waves, each controlled by it’s own LFO, but all together inside a single patch (this makes overall controlling of the patch much easier). Each sine wave’s pitch is transposed so they make standard music harmonics (second harmonic is octave up, third harmonic is 7 notes above second, fourth is 5 notes above third, etc…).
AditiveChordRAW.mp3 (115kB) – Same sound as above but a chord.
AditiveSynthesis.mp3 (310kB) – Organic type of sound created by aditive synthesis.
sqr_lfo_port.mp3 (146kB) – A little bit of fun with square wave and LFO.
expand_detune+7_singleOSC.mp3 (555kB) – The cool ”expand detune” function. According to manual this feature sets up a tuning differential (discord) between left and right channels. However, when you put the (stereo) width to 0 you get this crazy sounding PWM type of effect. Believe or not this sound example is a single saw oscilator . That is one sample, one timbre, one poly, no efx, no tricks or anything, just single saw waveform + expand detune function.
SineWave+reverb.mp3 (72kB) – Using sine wave oscillator and LFO to create simple organ. Little bit of reverb was added. Note: To create a sine wave and other ”analog” waveforms like pulse, saw, square or white noise, there are many good programs available of which some are freeware.
sqr_detune.mp3 (185kB) – Single square wave with 18 dB filter, full resonance and expand detune function set to +3.
AnalogPad.mp3 (555kB) – Following pad was created using one pulse wave, two saw waves and one sine wave. That’s all – the rest is A-4000.
SawPad.mp3 (422kB) – Another pad, this time made of three saw waves.
Resample.mp3 (137kB) – This sound started as a sine wave. Using resample, efx and a lot of programming i turned into this organ sound.
Resample2.mp3 (67kB) – Another organ sound that started as sine wave, then resampled.
transwave-mult_timbr.mp3 (2152kB) – Testing multi timbral part of A-4000. Simple drum set and a ”transwave” type of sound that i programmed.

Korg Trident MkI Demo

trident

After many years of search, and one unsuccessful purchase of a busted MkII (that i’ve never managed to repair) i’ve finally found a good condition MkI at a very good price. There was nothing to think about but to pull the trigger. Always liked that “cosmic” string sound of this machine. And in the meantime i’ve became a fan for its brass section as well, because there is something magical and retro about that brass section – it screams 70’s. Although the synth was manufactured in 1980, circuits inside were designed in the 70’s and that’s about how they sound! As of its synth section, the thyristor based VCO core can hardly disappoint you and in combination with powerful and liquid SSM filters, it just brings smile on your face each time you hit a note. Same design will be later used for the famous Korg Polysix, though many corners will be cut to make Polysix affordable for average (read: starving) musician. Trident was no doubt the flagship model, you know that big thing that dominates the center of the studio with its ability to cover a large sonic territory thanks to independent synth, string ensemble and brass sections. I just wish it had the arpeggiator and unison that later came with the Polysix. To anyone who played Polsix, already knows it is SO EASY to lose several hours just by dialing some nice resonant patch with a long release and then hitting a six note arpeggio while gently tweaking knobs. Don’t do it – you’ve been warned.

Back to Trident. While i must admit the price was good, it actually requires some minor work. Resonance does not work on the brass section (i suspect dead SSM2044), which is why brass in the demo will play only non resonant sweeps, so i apologize for that part. Also most of the push buttons are busted and these are first to be replaced after i fix the resonance issue. Luckily PCB boards inside are separated per unit, so i hope it really shouldn’t be hard to fix the brass section.

On the back of the unit there are CV inputs for synth and brass filter sections. Which is precisely what i did on one part of the demo. I’ve connected a Korg MS-20, built a simple Sample an Hold patch there and then routed that voltage into the Trident. Also used MS-20’s LFO to do the ramp down type of repeating note effects. In case you wonder how i’ve achieved that effect.

Overall the sound of the synth section is really nice, filter can be opened really high in the spectrum (not as some other analogs where it reaches certain range deep within human hearing) so you can achieve some razor sharp synth tones if required. The sound of Trident/Polysix VCO is hard to describe. It is not silky like Jupiter 8 or aggressive like Prophet 5. It is just something different on its own. I’d say somewhere in the middle between the mentioned two polys, at least for the PWM types of sounds. Bass is nice too, again unison would come as a killer feature and i’m not excluding the possibility to design it myself. I really want to hear this thing in unison because i know it will be brutal for basslines. Just like the Polysix, Trident’s VCOs don’t push a lot of power in the extreme low which is actually ideal type of VCO for unison types of sounds (you don’t end up in most of the headroom eaten by extreme low). For more info please check Robert L’s Trident review while i will proceed with the demo now:

Dynacord DRP-20 review

drp1x

German classic from the late 80’s (manufactured in 1989). For those on the other side of the great lake where this company is not so much known, we should mention that Dynacord has been a long time manufacturer for pro audio equipment. Old analogue classics like TAM-21 (flanger/chorus) and VRS-23 Vertical Reverberation System come first in mind (both from the late 70’s – still can be found relatively cheap).

Visually, what makes DRP-20 stand out in every rack is obviously its white face plate, which really looks cool. However, there is also the black face plate version around which has balanced jacks and is less noisy at the output stage (noiseless to be exact). Speaking about ease of use, i gave 5/10 because, it’s one of those devices, where you browse through the menu and tweak with the big knob. Can be time consuming and is not as “hands on” as some other devices, i.e. VRS-23 with one knob per function. Assigning patch names can be real pain in the a**, hence low mark on ease of use. I must admit everything is logically laid out in the unit, and once you get used to it, it’s point and shoot.

DRP-20 is a classic reverb processor made with a true stereo engine, which, on some algorithms can be split to “dual” mode where each line features its own “processor”. The unit features dedicated input and output level knobs with a HI/LO gain button for easy switching from a line level for synths to high gain setting for something like a guitar connected directly into the unit. On top of that, there’s a dedicated “mono input” and “mono output” jack on the front panel which guitarists might appreciate. For the classic studio setup, there are 2 inputs and 2 outputs on the back of the unit.

It is very hard to describe the sound of DRP-20 unless you actually try the unit, in your own setup, with your own gear, etc. I won’t bother describing the “sound” of reverb but rather focus on discussing the unit’s weaker and stronger points. First thing i should point out is that the short room reverb algorithms are definitely not this unit’s specialty. Much cheaper reverbs like Midiverb I and II are far better for the small rooms (ie. drums / percussion work). Luckily there are some really nice multi-taps for small room simulations to compensate for this somehow (intentional or not), but more on that later. Where DRP-20 really shines are the vast spatial reverbs. They sound so lush. And the tails on those are just magnificent! In fact i remember reading many years ago comments from ppl mistaking DRP-20 for Lexicon 480 and Quantec QRS (though this is all highly subjective, still somehow funny anecdote). One thing i know 100% sure, huge reverbs on this unit sound really really good! For some reason there are no much of them in the presets area, so you’ll have to make your own, using the buttons and the dial. Luckily there are more than enough parameters for the reverb, and every major aspect can be precisely set.

Speaking of other good stuff inside, the unit has some nice delay algorithms. One of my favorite is a VCO Delay, where LFO can be used to modulate the time. Since each line can be set independently in time and modulation amount, you can produce everything from wide choruses and flangers to old school delays with modulation (for some 60’s style Sci-Fi movie soundtrack). One very useful feature in the unit is the IN LOOP enable / disable function. Basically, when engaged, all the patches in the unit will have their “original signal amount” set to 0. In other words, all effects will be set to fully wet. This saves a lot of work to someone who’s moving from serial to loop connection or vice-versa, otherwise all the patches would manually have to be reprogrammed.

In Specs
The unit features 32 bit signal processing based around the NEC’s DSP chip. Converters are 16 bit, both the A/D and D/A section. The unit features MIDI for external control of all parameters in real time. Parameters in the unit can be set to either “value” of 0-100% or in classic dB scale which is very useful for both the beginners and the professionals – kudos to Dynacord for that. There are 128 preset locations and another 128 user locations for program storage.

DRP-20 has a total of 26 algorithms or effect structures as they call it. Each structure has its own range of parameters. Structures range from Echos, Reverbs, Plates, Rooms, Echos+Reverbs, Multitaps, Gated Reverbs to the Flanger and Chorus effect. There are also five dual channel algorithms which basically feature echo line on one channel and room or plate on the other channel.

Speaking of Multitaps, last algorithm features 2×11 taps which can produce some really nice room simulations! Those 11 delays per channel are grouped with another channel to form a stereo cluster which can be drawn using cluster time function. It is possible to select one from 9 different clusters for each channel, for some really exotic stereo room effects. Further more, cluster time can be independently set for each channel (L & R) along with the independent feedback amount for each. With some careful programming, a collection of really cool room reverbs can easily be built.

Full list of algorithms:
Two Channel Echo
VCO-Echo Stereo
Plate Reverb
Room Reverb
Echo + Plate
Echo + Room
VCO-Echo + Plate
VCO-Echo + Room
Echo + Live Reverb
L = Echo / R = Plate
L = Echo / R = Room
L = VCO / R = Plate
L = VCO / R = Room
L = Echo / R = Live
Freeze Automatic
Freeze Manual
Plate Reverb & Gate
Room Reverb & Gate
Gated Reverb
Echo + Gated Reverb
Multitap 2×3
Multitap 2×6 Syn
Multitap Presets
Stereo Flanger
Stereo Chorus