ThrillseekerLA mkII released

ThrillseekerLA mkII – bringing mojo back

ThrillseekerLA is an optical stereo compressor optimized for gentle mix bus coloring. It combines smoothest optical compression with vibrant coloration options that deliver a unique box tone in their own right, including thrilling bass and elegant top end void of any harshness in the mids. Its compression not only glues things together effortlessly but also enhances the stereo image by increasing depth and dimension.

10 years after – new in version 2:

  • Technical redesign with advanced opto cell emulation
  • Simplified gainstaging including automatic output gain compensation
  • Streamlined coloring options: Interstage, Tube and Loudness
  • New compress/limit option and reworked sidechain filtering

The mkII update is available for Windows VST in 32 and 64bit as freeware. Download your copy here.

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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.

now that we’ve reinvented 8-bit audio – what’s next?

Gone are the days when choosing the right noise shaping for dithering was a headache and we were still dreaming of Hires Audio as the glorious upcoming consumer format. In the end, it was more important to us to make everything brutally loud. As if it could otherwise be overheard in all the streaming mush: Radio and advertising have led the way, after all, and in the realm of asocial media, the cry for attention is naturally even louder. A pleasant sound, transient-rich and detailed? Forget it! The pitiful remainder of dynamics could now be packaged in 8-bit, lossless. That’s how it looks. The other day I saw a report about audiophiles who still and tirelessly spare neither expense nor effort to optimize the sound at home down to the smallest detail, which seemed unintentionally comical in this context. But it seems just as anachronistic today to spend such an immense technical effort for an inferior production performance. How can this be justified? Surely any stock EQ, a decent multiband compressor and limiter should suffice. In the attached video, the evolution of EDM is outlined in fast forward from the 80s to 2020. Regardless of the qualities of the codecs, this shows very impressively the decline in technical production quality over the years and the side effects of the increase in loudness. Now that we have successfully ruined audio quality, the question remains: What’s next?

the history of Cubase

When Cubase 3.0 came out in 1996 and introduced VST for the first time with all its new and fascinating possibilities, that was the point where I decided to get more involved in music production and set up a small (home) recording studio. VST was the basis for all this and how I imagined a modern (computer based) studio production. What a revolution that was. Watching this video today brings up a lot of nostalgic feelings …

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.

I just had to have this

  • Hardcover book with almost 300 pages
  • Covering most nerdy vintage studio classics from AKG, AMS, Dynacord, EMT, Lexicon, MXR, Quantec, Roland, ….
  • Phaser, delays, reverbs, pitch shifter, vocoder, exciter, multi fx – 68 devices presented in total

  • Great pictures (color and b&w) as well as insightful stories and statements from artists and manufacturers
  • Very fun to read or just to browse – inspirational in all regards
  • Available at Thomann for 69 smackers

next level saturation experience & still missing VoS plugins

The magic is where the transient happens.

Since a year or so I’m not just updating my audio plugin catalog but also focusing on bringing the original Stateful Saturation approach to the next level. That concept was already introduced 2010, embracing the fact that most analog circuit saturation affairs are not static but a frequency and load dependent matter which can be best modeled by describing a system state – hence the name Stateful Saturation.

The updated 2022 revision is now in place and got further refined regarding the handling of audio transient states 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. This was profoundly influenced by working with audio exciters over the recent years but also by deep diving further into the field of psychoacoustics.

This important update was also the reason why I actually did hold back some of the plugin updates, namely TesslaPRO and the Thrillseeker compressors since they heavily rely on that framework. Meanwhile, TesslaPRO has been rewritten based on the framework update already and will be released early September. ThrillseekerLA and VBL are in the making and scheduled for Q4.

how I listen to audio today

Developing audio effect plugins involves quite a lot of testing. While this appears to be an easy task as long as its all about measurable criteria, it gets way more tricky beyond that. Then there is no way around (extensive) listening tests which must be structured and follow some systematic approach to avoid ending up in fluffy “wine tasting” categories.

I’ve spend quite some time with such listening tests over the years and some of the insights and principles are distilled in this brief article. They are not only useful for checking mix qualities or judging device capabilities in general but also give some  essential hints about developing our hearing.

No matter what specific audio assessment task one is up to, its always about judging the dynamic response of the audio (dynamics) vs its distribution across the frequency spectrum in particular (tonality). Both dimensions can be tested best by utilizing transient rich program material like mixes containing several acoustic instruments – e.g. guitars, percussion and so on – but which has sustaining elements and room information as well.

Drums are also a good starting point but they do not offer enough variety to cover both aspects we are talking about and to spot modulation artifacts (IMD) easily, just as an example. A rough but decent mix should do the job. On my very own, I do prefer raw mixes which are not yet processed that much to minimize the influence of flaws already burned into the audio content but more on that later.

Having such content in place allows to focus the hearing and to hear along a) the instrument transients – instrument by instrument – and b) the changes and impact within particular frequency ranges. Lets have a look into both aspects in more detail.

a) The transient information is crucial for our hearing because it is used not only to identify intruments but also to perform stereo localization. They basically impact how we can separate between different sources and how they are positioned in the stereo field. So lets say if something “lacks definition” it might be just caused by not having enough transient information available and not necessarily about flaws in equalizing. Transients tend to mask other audio events for a very short period of time and when a transient decays and the signal sustains, it unveils its pitch information to our hearing.

b) For the sustaining signal phases it is more relevant to focus on frequency ranges since our hearing is organized in bands of the entire spectrum and is not able to distinguish different affairs within the very same band. For most comparision tasks its already sufficient to consciously distinguish between the low, low-mid, high-mid and high frequency ranges and only drilling down further if necessary, e.g. to identify specific resonances. Assigning specific attributes to according ranges is the key to improve our conscious hearing abilities. As an example, one might spot something “boxy sounding” just reflecting in the mid frequency range at first sight. But focusing on the very low frequency range might also expose effects contributing to the overall impression of “boxyness”. This reveals further and previously unseen strategies to properly manage such kinds of effects.

Overall, I can not recommend highly enough to educate the hearing in both dimensions to enable a more detailed listening experience and to get more confident in assessing certain audio qualities. Most kinds of compression/distortion/saturation effects are presenting a good learning challenge since they can impact both audio dimensions very deeply. On the other hand, using already mixed material to assess the qualities of e.g. a new audio device turns out to be a very delicate matter.

Lets say an additional HF boost applied now sounds unpleasant and harsh: Is this the flaw of the added effect or was it already there but now just pulled out of that mix? During all the listening tests I’ve did so far, a lot of tainted mixes unveiled such flaws not visible at first sight. In case of the given example you might find root causes like too much mid frequency distortion (coming from compression IMD or saturation artifacts) mirroring in the HF or just inferior de-essing attempts. The most recent trend to grind each and every frequency resonance is also prone to unwanted side-effects but that’s another story.

Further psychoacoustic related hearing effects needs to be taken into account when we perform A/B testing. While comparing content at equal loudness is a well known subject (nonetheless ignored by lots of reviewers out there) it is also crucial to switch forth and back sources instantaneously and not with a break. This is due to the fact that our hearing system is not able to memorize a full audio profile much longer than a second. Then there is the “confirmation bias” effect which basically is all about that we always tend to be biased concerning the test result: Just having that button pressed or knowing the brand name has already to be seen as an influence in this regard. The only solution for this is utilizing blind testing.

Most of the time I listen through nearfield speakers and rarely by cans. I’m sticking to my speakers since more than 15 years now and it was very important for me to get used to them over time. Before that I’ve “upgraded” speakers several times unnecessarily. Having said that, using a coaxial speaker design is key for nearfield listening environments. After ditching digital room correction here in my studio the signal path is now fully analog right after the converter. The converter itself is high-end but today I think proper room acoustics right from the start would have been a better investment.