What I like about the Behringer 2600

What I really like about the Behringer 2600 is that it’s not just a plain copy but introduces some real useful improvements over the original concept. Most important to me is the 19″ form factor which not only reduces the originals size and fits in the rack but also remains big enough to enjoy a great user experience while cabling and tweaking things. And they got rid of those speakers! Instead it offers 2 filter revisions to choose from, two of the oscillator sections are now fully featured, the LFO is part of the main chassis now and new additional timing options for the envelopes has been added as well.

On the other hand, the Behringer 2600 sticks to CV gate voltages following original levels which limits full integration in todays modular world quite a bit. However, this is currently not a big deal to me. I only wish they would have made a true analog delay instead of the spring reverb (emulation). Offering audio in, the device also doubles as an excellent analog effect unit which seems to be a little bit underrated in this regard. Given it’s pricepoint, this feature is already something to consider if one is just looking for an outboard analog filter box or an alternative for something like a MS-20.

The ARP 2600 turns 50

And if time allows, watch this awesome documentary about the history and story behind ARP:

What loudspeakers and audio transformers do have in common

Or: WTF is “group delay”?

Imagine a group of people visiting an exhibition having a guided tour. One might expect that the group reaches the exhibitions exit as a whole but in reality there might be a part of that group just lagging behind a little bit actually (e.g. just taking their time).

Speaking in terms of frequency response within audio systems now, this sort of delay is refered to as “group delay”, measured in seconds. And if parts of the frequency range do not reach a listeners ear within the very same time this group delay is being refered to as not being constant anymore.

A flat frequency response does not tell anything about this phenomena and group delay must always be measured separately. Just for reference, delays above 1-4ms (depending on the actual frequency) can actually be perceived by human hearing.

This always turned out to be a real issue in loudspeaker design in general because certain audio events can not perceived as a single event in time anymore but are spread across a certain window of time. The root cause for this anomaly typically lies in electrical components like frequency splitters, amplifiers or filter circuits in general but also physical loudspeaker construction patterns like bass reflex ports or transmission line designs.

Especially the latter ones actually do change the group delay for the lower frequency department very prominently which can be seen as a design flaw but on the other hand lots of hifi enthusiast actually do like this low end behaviour which is able to deliver a very round and full bass experience even within a quite small speaker design. In such cases, one can measure more than 20ms group delay within the frequency content below 100Hz and I’ve seen plots from real designs featuring 70ms at 40Hz which is huge.

Such speaker designs should be avoided in mixing or mastering situation where precision and accuracy is required. It’s also one of the reasons why we can still find single driver speaker designs as primary or additional monitoring options in the studios around the world. They have a constant group delay by design and do not mess around with some frequency parts while just leaving some others intact.

As mentioned before, also several analog circuit designs are able to distort the constant group delay and we can see very typical low end group delay shifts within audio transformer coupled circuit designs. Interestingly, even mastering engineers are utilizing such devices – whether to be found in a compressor, EQ or tape machine – in their analog mastering chain.

white is the new black

SOMA LYRA-8 into KORG MS-20

out now: SlickEQ “Gentleman’s Edition”

SlickEQ_German

Key specs and features

  • Modern user interface with outstanding usability and ergonomics
  • Carefully designed 64bit “delta” multi-rate structure
  • Three semi-parametric filter bands, each with two shape options
  • Five distinct EQ models: American, British, German, Soviet and Japanese
  • Low band offers an optional phase-lag able to delay low frequencies relative to higher frequencies
  • High pass filter with optional “Bump” mode
  • Low pass filter with two different slopes (6dB/Oct and 12dB/Oct)
  • Parametric Tilt filter with optional “V” mode.
  • Six output stages: Linear, Silky, Mellow, Deep, Excited and Toasted
  • Advanced saturation algorithms by VoS (“Stateful saturation”)
  • Highly effective loudness compensated auto gain control
  • Stereo, mono and sum/difference (mid/side) processing options
  • Frequency magnitude plot
  • Tool-bar with undo/redo, A/B, advanced preset management and more

SlickEQ is a collaborative project by Variety of Sound (Herbert Goldberg) and Tokyo Dawn Labs (Vladislav Goncharov and Fabien Schivre). For more details, please refer to the official product page: http://www.tokyodawn.net/tdr-vos-slickeq-ge/

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compressor aficionados (8) – Sascha Eversmeier

Sascha, are you a musician yourself or do you have some other sort of musical background? And how did you once got started developing your very own audio DSP effects?

I started learning to play bass guitar in early 1988, when I was 16. Bass is still my main instrument, although I also play a tiny bit of 6-string, but I’d say I suck at that.

The people I played with in a band in my youth where mostly close friends I grew up with, and most of us kept on making music together when we finished school a couple of years later. I still consider that period (mid-nineties) as sort of my personal heyday, musical-wise. It’s when you think you’re doing brilliant things but the world doesn’t take notice. Anyway. Although we all started out doing Metal, we eventually did Alternative and a bit of Brit-influenced Wave Rock back then.

That was also the time when more and more affordable electronic gear came up, so apart from doing the usual rock-band lineup, we also experimented with samplers, DATs, click tracks and PCs as recording devices. While that in fact made the ‘band’ context more complex – imagine loading in a dozen disks into the E-MU on every start of the rehearsal until we equipped it with an MO drive – we soon found ourselves moving away from writing songs through jamming and more to actually “assembling” them by using a mouse pointer. In hindsight, that was really challenging. Today, the DAW world and the whole process of creating music is so much simpler and intuitive, I think.

My first “DAW” was a PC running at 233Mhz, and we used PowerTracks Pro and Micro Logic – a stripped-down version of Logic -, although the latter never clicked with me. In 1996 or 97 – can’t remember – I purchased Cubase and must have ordered right within a grace period, as I soon got a letter from Steinberg saying they now finished the long-awaited VST version and I could have it for free, if I want. WTF? I had no idea what they were talking about. But Virtual Studio Technology, that sounded like I was given the opportunity to upgrade myself to being “professional”. How flattering, you clever marketing guys. Yes, gimme the damn thing, hehe.

When VST arrived, I was blown away. I had a TSR-8 reel machine, a DA-88 and a large Allen&Heath desk within reach and was used to run the computer as a midi sequencer mainly. And now, I could do it all inside that thing. Unbelievable. Well, the biggest challenge then was finding an affordable audio card, and I bought myself one that only had S/PDif in & outputs and was developed by a German electronics magazine and sold in small amounts through a big retail store in Cologne, exclusively. 500 Deutschmarks for 16 bits on an ISA card. Wow.

The first plugin I bought was Waves Audio Track, sort of a channel strip, which was a cross-promotion offer from Steinberg back then, 1997, I guess. I can still recall its serial number by heart.

Soon, the plugin scene lifted off, and I collected everything I could, like the early mda stuff, NorthPole and other classics. As our regular band came to nothing, we gathered our stuff and ran sort of a small project studio where we recorded other bands and musicians and started using the PC as the main recording device. I upgraded the audio hardware to an Echo Darla card, but one of my mates soon brought in a Layla rack unit so that we had plenty of physical ins and outs.

You really couldn’t foresee where the audio industry would go, at least I couldn’t. I went fine with this “hybrid” setup for quite a long time, and did lots of recording and editing back then, but wasn’t even thinking of programming audio software myself at all. I had done a few semesters of EE studies, but without really committing myself much.

Then the internet came along. In 1998, I made a cut and started taking classes in Informatics. Finished in 2000, I moved far away, from West Germany, to Berlin and had my first “real” job in one of those “new economy” companies, doing web-based programming and SQL. That filled the fridge and was fun to do somehow, but wasn’t really challenging. As my classes included C, C++ and also Assembler, and I still got a copy of Microsoft’s Visual Studio, I signed up to the VST SDK one day. At first, I might have done pretty much the same thing as everybody: compile the “gain” and “delay” plugin examples and learn how it all fits together. VST was still at version 1 at that time, so there were no instruments yet, but I wasn’t interested much in those anyway, or at least I could imagine writing myself a synthesizer. What I was more interested in was how to manipulate the audio so that it could sound like a compressor or a tube device. I was really keen on dynamics processing at that time, perhaps because I always had too few of those units. I had plenty available when I was working part-time as a live-sound engineer, but back in my home studio, a cheap Alesis, dbx or Behringer was all I could afford. So why not try to program one? I basically knew how to read schematics, I knew how to solder, and I thought I knew how things should sound like, so I just started out hacking things together. Probably in the most ignorant and naive way, from today’s perspective. I had no real clue, and no serious tool set, apart from an old student’s copy of Maple and my beloved Corel 7. But there were helpful people on the internet and a growing community of people devoted to audio software, and that was perhaps the most important factor. You just weren’t alone. [Read more…]

compressor aficionados (7) – Dave Gamble

Dave, can you tell us a little about how you got into music, and your professional career as an audio effects developer so far?

Started writing trackers as a child, then wrote some code to allow me to DJ with trackers. By 14 I was writing commercial software. Had some great teachers and lecturers who helped me a lot. Did my final-year project with Focusrite. Won the project prize. Spent 4.5 years at Focusrite (I was employee 12 or 13) to add DSP to the company, during which time we acquired Novation, and grew quite a lot. We made a lot of money from audio interfaces, so that kinda took over, and I wanted to get back to the DSP (at Focusrite I did Forte suite, helped with Liquid Channel/Mix, Saffire suite, plus other non DSP projects). Left for Sonalksis, built all their shipping products (except CQ1 and DQ1), although I’d built tbk1 years before and they’d been selling it. Was fun but chaotic. Left to go freelance so I could start my own outfit, during which time I worked with Neyrinck, TAC System, Focusrite, Novation, Studio Devil, FXpansion, Brainworx/Plugin Alliance, etc. Then started dmgaudio. And here we are now. [Read more…]

what I’m currently working on – Vol. 10

Right now, I’m extending my “compressor aficionados” interview series by a couple of outstanding developers I’ve always was interested in and wanted to talk with. Beside that, I’m pretty much delved in research and development plus testing some brand new prototypes. If/when another mkII version will appear is currently not clear – but me thinks that there will be one or another surprise during Q4, though.

In recent history, I’ve constantly extended and improved my Stateful Saturation approach and within ThrillseekerVBL I’ve managed to introduce authentic analog style sounding distortion right into VST land, which is what I’ve always had in my mind and dreamed of. And there’s so much and overwhelming feedback on that – thank you very much!

And since quite a while, I’m dreaming about a brand new series of plug-ins which will combine the strength of both worlds: analog modelling on the one side but pure digital techniques on the other – incorporating techniques such as look-ahead, FIR filtering or even stuff that comes from the digital image processing domain, such as HDR (High Definition Range) imaging.

Expect an exciting announcement quite soon …

modeling the distortion in Thrillseeker VBL

It’s so important to get the non-linear modeling right if we would like to have a sort of analog feel in the digital domain. I can’t stress this ever enough since it still seems to be a common practise in todays audio plug-in design to just throw in a static waveshaper, oversample it and hope this will make everything alright. Not! Even worse, in a recently released plug-in I saw the static waveshapers curve not being continuous again and I’m not going to talk about the sound.

But what should one expect to hear if the analog modeling is just done right? Only by driving the gain of the unit but way before we notice the obvious distortions there appear different by-products caused by circuit side-effects. Depending on the actual device, circuit and components, it might be that the signal starts just getting thicker and more mid-focused, as an example. Or, the signal might appear much deeper and bigger in other cases.

Whatever it might be in particular, I do call this the “Mojo” of the device – it’s not the primary intention of the device but turns out to be a sort of an added sugar. Such effects are highly frequency, transient and gain structure dependent and this is what makes the processed signal to be much more vibrant and alive. Furthermore, the obvious harmonic distortions are not introduced abruptly but they emerge gradually.

Roland System 100m

(via)