why the Thrillseeker compressors complement each other so well

Audio compressors use either a “feed forward” or “feedback” design to control the gain of an audio signal. In a feed forward compressor, the input signal is used directly to control the gain of the output signal. Essentially, the compressor compares the input signal to a threshold and reduces the gain of the output signal if the input signal exceeds the threshold. In a feedback compressor, the output signal is fed back into the compressor and used to control the gain of the input signal. So, the compressor compares the output signal to a threshold and reduces the gain of the input signal if the output signal exceeds the threshold. Both feed forward and feedback compressors can be effective at controlling the dynamic range of an audio signal, but they operate in slightly different ways and do have different characteristics in terms of their sound and response.

However, the specific sound of a device depends largely on other features of the circuit design and its components. For example, an optoelectric compressor uses a photoresistor or photodiode to detect and control the degree of gain reduction of the signal. But the make-up amplifier afterwards may contribute the most to the sound, depending on its design (tube or solid state). A variable gain tube compressor, on the other hand, uses a vacuum tube to control the gain of the compressor. The vacuum tube is used to amplify the signal, and the gain of the compressor is controlled by changing the bias voltage of the tube. This alone provides a very typical, distinctive sound that is very rich in harmonic overtones.

Both opto-electrical and variable-mu tube compressors are commonly used in audio production to control the dynamic range of a signal, but they operate in different ways and can produce different tonal characteristics. Opto-electrical compressors are known for their fast attack times and smooth release characteristics, while variable-mu tube compressors are known for their warm and smooth sound.

the beauty of opto-electrical compression – volume 2

When I was looking for a sophisticated stereo compressor for the outboard studio rack a year ago, I was surprised to see how many of the more interesting models now use opto-electric compression technology. Whether transparent or coloring, tube or solid-state amplifiers, transformer or transformerless, even two-channel layouts in mid/side encoding: far advanced compared to all the classic mono replicas.

Optical compressors are usually characterized by their distinct program-dependent compression behavior, mainly based on a physical memory effect in the detector itself. Other subtle nuances are found across the frequency spectrum that affect timing and curve characteristics, creating a complexity that cannot be reduced to simple two-stage controlled release curves, and which is the beauty of opto-electrical compression in its entirety.

Significant audio signal colorations, however, are shaped not by the gain reduction circuitry but by the make-up gain amplifier, whether it is tube or solid-state. Here, the audio transformer also plays an important role in polishing the transients and creating a cohesive sound.

ThrillseekerLA was designed from the beginning in 2012 as a modern stereo compressor with exciting sound coloring possibilities. It is a compressor with authentic opto-electric control behavior in feed-forward circuit topology.

The upcoming mkII update is a technical redesign dedicated solely to improving the sound. It delivers a unique box tone with thrilling bass and elegant top end void of any harshness in the mids. The compression not only glues everything together effortlessly, but also enhances the stereo image by adding depth and dimension.

The release is scheduled for mid-December.

bringing mojo back – volume 2

ThrillseekerVBL is an emulation of a vintage broadcast limiter design that follows the classic Variable-Mu design principles from the early 1950s. These tube-based devices were initially used to prevent audio overloads in broadcast transmission by managing sudden level changes in the audio signal. From today’s perspective, and compared to digital dynamic processors, they appear to be rather slow and can be considered more of a gain structure leveler. However, they still shine when it comes to gain riding in a very musical way – they’ve written warmth and mojo all over it.

ThrillseekerVBL is a modded version that not only features basic gain control, but also gives detailed access to both compression behavior and the characteristic of tube circuit saturation effects. Used in subtle doses, this provides the analog magic we so often miss when working in the digital domain while overdriving the circuit achieves much more drastic musical textures as a creative effect.

ThrillseekerVBL offers an incredibly authentic audio transformer simulation that models not only the typical low-frequency harmonic distortion, but also all the frequency- and load-dependent subtleties that occur in a transformer-coupled tube circuit and that contribute to the typical mojo we know and love from the analog classics.

new in version 2

Conceptually, the mkII version has been refined in that the peak limiting itself is no longer the main task but versatile and musically expressive gain control as well as a thrilling saturation experience. The saturation is now an integral part of the compression and is perfectly suited for processing transient-rich material. Both compression and saturation can be individually activated and controlled.

The circuit-related frequency loss in the highs has been almost eliminated and the brilliance control – originally intended just for compensation – can now also perform exciter-like tasks. The bias control has been extended to shape the harmonic spectrum in much greater detail by allowing the contribution of second order harmonics as well as the adjustment of the saturation behavior in the transient area of the signals. The transformer circuit has also been technically revised not only to resolve all the subtleties realistically but also to reproduce an overall tighter sound image.

ThrillseekerVBL has become a real tonebox, able to reproduce a wide range of tonalities. It provides access to the attack and release behavior and all compression controls can also affect the saturation of the signal, even when the compression function is turned off. This allows specific textures of signal saturation to be realized. As with the good old outboard devices, the desired sound colorations can be achieved just by controlling the working range. And if too much of a good thing is used, the DRY/WET control simply shifts down a gear.

To further improve the user experience some additional UI elements have been added giving more visual feedback. Although oversampling has been added, the actual cpu load was significantly reduced thanks to efficient algorithms and assembler code optimizations.

ThrillseekerVBL mkII will be released October 14th for Windows VST in 32 and 64bit as freeware.

sidechain linking techniques

How an audio compressor responds to stereo content depends largely on how the channel linking is implemented in the sidechain. This has a major influence on how the spatial representation of a stereo signal is preserved or even enhanced. The task of the compressor designer is to decide which technical design is most suitable for a given overall concept and to what extent the user can control the linkage when using the device.

In analog compressor designs, in addition to unlinked “dual mono” operation, one usually finds simple techniques such as summing both stereo channels (corresponding to the center of the stereo signal) or the extraction of the maximum levels of both channels using a comparator circuit implementing the mathematical term max(L,R).

More sophisticated designs improve this by making the linking itself frequency dependent, e.g. by linking the channels only within a certain frequency range. It is also common to adjust the amount of coupling from 0 to 100%, and the API 2500 hardware compressor serves as a good example of such frequency dependent implementation. For the low and mid frequency range, simple summing often works slightly better in terms of good stereo imaging, while for the mid to high frequency range, decoupling to some degree often proves to be a better choice.

The channel coupling can also be considered as RMS (or vector) summing, which can be easily realized by sqrt(L^2+R^2). As an added sugar, this also elegantly solves the rectification problem and results in very consistent gain reduction across the actual level distributions that occur between two channels.

If, on the other hand, one wants to focus attention on correlated and uncorrelated signal components individually (both of which together make up a true stereo signal), then a mid/side decomposition in the sidechain is the ticket: A straight forward max(mid(L,R), side(L,R)) on the already rectified channels L and R is able to respond to any kind of correlated signal not only in a very balanced way but also to enhance its spatial representation.

More advanced techniques usually combine the methods already described.

TesslaPRO mkIII released

the magic is where the transient happens

The Tessla audio plugin series once started as a reminiscence to classic transformer based circuit designs of the 50s and 60s but without just being a clone stuck in the past. The PRO version has been made for mixing and mastering engineers working in the digital domain but always missing that extra vibe delivered by some highend analog devices.

TesslaPRO brings back the subtle artifacts from the analog right into the digital domain. It sligthly colors the sound, polishes transients and creates depth and dimension in the stereo field to get that cohesive sound we’re after. All the analog goodness in subtle doses: It’s a mixing effect intended to be used here and there, wherever the mix demands it.

The mkIII version is a technical redesign, further refined to capture all those sonic details while reducing audible distortions at the same time. It further blurs the line between compression and saturation and also takes aural perception based effects into account.

Available for Windows VST in 32 and 64bit as freeware. Download your copy here.

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 Ibanez AD202 analog delay
• the Dynacord VRS-23 analog delay

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.