the twisted world of guitar pedals II

Meanwhile I had the opportunity to put my hands on some Fairfield Circuitry effect pedal stuff mentioned earlier here and the “Meet Maude” analog BBD delay was right here on my desk for a deeper inspection. My actual experience was a rather mixed one.

Focusing on a rather dark and LoFi sound quality on the one hand plus a rather simplistic feature set concept wise on the other, they do not appear to be very flexible in practise and this at a rather steep price point. They appear to be very noisy featuring all kinds of artifacts even when integrated to the mixing desk via reamping. One may call this the feature itself but at the end it makes it a one-trick pony. If you need exactly that, here you have it but you get nothing beyond that. To me this trade off was too big and so I send it back.

However, I found their nifty low pass gate implementation (very prominently featured within their “Shallow Water”) that much unique and interesting that I replicated it as a low pass filter alternative in software and to have it available e.g. for filtering delay lines in my productions. The “Shallow Water” box made me almost pull the trigger but all in all I think this stuff seems to be a little bit over-hyped thanks to the interwebs. This pretty much sums it up for now, end of this affair.

Timeline & BigSky – The new dust collectors?

Going into the exact opposite direction might be a funny idea and so I grabbed some Strymon stuff which aims to be the jack of all trades at least regarding digital delay and reverb in a tiny stomp box aka desktop package. To be continued …

Further readings about BBD delays:

the twisted world of guitar pedals I

Quite recently I had a closer look into the vast amount of (guitar) effect pedals out there. Most are already DSP based which surprised me a little bit since I still ecpected more discrete analog designs after all. While looking for some neat real analog BBD delay I finally stumbled across Fairfield Circuitry’s “Meet Maude” which got me intrigued, having a rather rough look&feel at first sight but some very delicate implementation details under the hood.

Their delay modulation circuit has some randomness build in and also there is a compression circuit in the feedback loop – both designs I’ve also choosen for NastyDLA and which makes a big impact on the overall sound for granted. But the real highlight is the VCF in the delay feedback path which actually appears to be a low-pass gate – a quite unique design and soundwise also different but appealing in its very own regard.

They employed very similar concepts to their vibrato/chorus box “Shallow Water” featuring also random delay modulation and a low pass gate but this time a little bit more prominent on the face plate. On top, their JFET op-amp adds some serious grit to any kind of input signal. All in all, I did not expect such a bold but niche product to exist. If I ever will own such a thingy, there will be a much more detailed review here for sure.

The TesslaSE Remake

There were so many requests to revive the old and rusty TesslaSE which I’ve once moved already into the legacy folder. In this article I’m going to talk a little bit about the history of the plugin and its upcoming remake.

The original TesslaSE audio plugin was one of my first DSP designs aiming at a convincing analog signal path emulation and it was created already 15 years ago! In its release info it stated to “model pleasant sounding ‘electric effects’ coming from transformer coupled tube circuits in a digital controlled fashion” which basically refers to adding harmonic content and some subtle saturation as well as spatial effects to the incoming audio. In contrast to static waveshaping approaches quite common to that time, those effects were already inherently frequency dependent and managed within a mid/side matrix underneath.

(Later on, this approach emerged into a true stateful saturation framework capable of modeling not only memoryless circuits and the TesslaPro version took advantage of audio transient management as well.)

This design was also utilized to supress unwanted aliasing artifacts since flawless oversampling was still computational expensive to that time. And offering zero latency on top, TesslaSE always had a clear focus on being applied over the entire mixing stage, providing all those analog signal path subtleties here and there. All later revisions also sticked to the very same concept.

With the 2021 remake, TesslaSE mkII won’t change that as well but just polishing whats already there. The internal gainstaging has been reworked so that everything appears gain compensated to the outside and is dead-easy to operate within a slick, modernized user interface. Also the transformer/tube cicuit modeling got some updates now to appear more detailed and vibrant, while all non-linear algorithms got oversampled for additional aliasing supression.

On my very own, I really enjoy the elegant sound of the update now!

TesslaSE mkII will be released by end of November for PC/VST under a freeware license.

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.

interview series (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…]

interview series (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…]

announcing Thrillseeker VBL – Vintage Broadcast Limiter

Bringing mojo back – Thrillseeker VBL is an emulation of a “vintage broadcast limiter” following the classic Variable-Mu design principles from the early 1950’s. They were used to prevent audio overshoots by managing sudden signals changes. From today’s perspective, and compared to brickwall limiters, they are rather slow and should be seen as more of a gain structure leveler, but they still are shining when it comes to perform gain riding in a very musical fashion – they have warmth and mojo written all over.

Thrillseeker VBL is a “modded” version, which not only has the classic gain reduction controls but also grants detailed access to the amount and appearance of harmonic tube amplifier distortion occurring in the analog tube circuit. Applied in subtle doses, this dials in that analog magic we often miss when working in the digital domain, but you can also overdrive the circuit to have more obvious but still musical sounding harmonic distortion (and according side-effects) for use as a creative effect.

On top, Thrillseeker VBL offers an incredibly authentic audio transformer simulation which not only models the typical low-end harmonic distortion but also all the frequency and load dependent subtleties occurring in a transformer coupled tube circuit, and which add up to that typical mojo we know from the analog classics. This would not have been possible with plain waveshaping techniques but has been realized with my innovative Stateful Saturation approach, making it possible to model circuits having a (short) sort of memory.

Release date is not yet confirmed but most probably will be in May this year.

tasty meal preparations with Density mkIII

Since precise routing and stuff like that is not taken down into the cookbook as of now, here are some exciting tips and tricks to experiment with and maybe to obtain a different approach to cook audio with Density mkIII.

Starter

As a starter just use the default preset and dial in huge amounts of compression right with the DRIVE knob. Now mix this back to the dry signal by using the DRY:WET option to obtain a thick sounding result (New York style compression). Since the COLOR option ignores any DRY:WET settings one can dial it in afterwards to thicken the soup even further. Hmm, tasty!

Second course

Set DRY:WET back to a 100% wet signal but also pull RANGE back to the left so that there will be no gain reduction anymore. There is no compression anymore now but one can still use the MAKEUP knob to drive the gain of the non-linear circuits. Use this and experience a hot (driven) meal.

Main course

By finishing the second course, you not only have a sophisticated non-linear amplifier now where you can dial in the coloration with the COLOR knob to taste. You also can use this in M/S mode to adjust the stereo imaging in a quite unique fashion just by adjusting the amounts of saturation per channel right with the MAKEUP knobs. Omph, I’m feelin so wide now!

Dessert

Just dial in again some amounts of compression by turning RANGE clockwise, maybe full to the right but RELAX the attack times so that some transients can pass. Those will be eaten now by the non-linear amplifier as an added sugar.

Espresso, anyone?

the Dynacord VRS-23 analog delay

(click images to enlarge)

The VRS-23 was a quite successful BBD delay in the 80’s and some thousands of units were sold during that time. It’s a mono-in / stereo-out device and capable of delay times up to around 400ms. Providing also very short timings and a modulation option makes it capable of creating chorus and flanger type of effects as well. There were different revisions available and shown here is a later one with the white faceplate. [Read more…]

the Ibanez AD202 analog delay

The so-called bucked-brigade device (BBD) delay line generator is a somehow quirky and really unique technical design. Such  devices are built upon analog components entirely, but being discrete in time they are halfway digital. Their analog input voltage samples are stored and moved through a line of capacitors one step after another and hence the name comes from analogy with the term bucket brigade: a line of people passing buckets full of water. [Read more…]