interview series (13) – Eric Valentine

Eric, from record producer to manufacturing gear – how came?

The genesis of the whole thing started with my search for a new analog console that did everything I needed.  I had used vintage Neve’s for most of my professional recording life.  When the one I was using was at that point where I couldn’t trust it anymore, and it was gonna take 10s of thousands of dollars to make it healthy again, I decided to take on the crazy endeavor of building a console from scratch.  The only reason I even considered going down that road was because of a fine gentlemen named Larry Jasper.  He was a freelance studio tech in the Los Angeles area and had been keeping my tape machines and outboard gear happy.  He was just beyond brilliant.  Beyond any studio tech I had ever encountered. His encyclopedic knowledge of ALL the vintage circuitry and his effortless understanding of how audio circuitry works was truly astounding.  he reads schematics like the Sunday comics.  He was never wrong, and his brilliance  was only matched by his gentle nature and generosity of spirit.  I was sure I knew what I wanted, and I was sure Larry could design the circuitry to make it work.

We set off on the journey of building a large format automated console from scratch.  It took a couple of years, but we ended up with a console that had so many things that were unique and had never been done before.  Other people got curious, and then it seemed that maybe we could sell this stuff.  Undertone Audio was born 🙂

We made custom consoles for a handful of brave souls that trusted what Larry and I were doing.  It was Greg Wells, Fraser T Smith, Seedy Underbelly studios etc etc.  From there, it was obvious that the best market would be outboard gear.  Everyone was at the beginning of transitioning to Hybrid mixing, and we needed to make things that could work in that context.  We made a channel strip from the console.  We made a 4 channel mic preamp, and over the years, the product line has continued to expand.

For me, UTA has simply been a place for me to solve problems by creating something new instead of accepting what is available.  Each product always starts with a specific problem to solve or tool that I am looking to add to my workflow. It  has been an amazing dream factory of audio toys for me.  The process always starts with “Man, I wish I could just…” or “Wouldn’t it be cool if there was a thing that did…”

I am still dreaming and grateful for the amazing team of people at UTA that keep bringing the dreams to life.

And then, how did it come to tackle a reissue of the Fairchild 670, of all things? A sixty-year-old legacy of studio engineering, a legend, commonly referred to as the “Holy Grail” compressor. What was the reason?

That Fairchild was totally random.  It really wasn’t something I was specifically chasing.  There was another really cool studio tech guy named Garen Avetisyan.  He was making some cool tube gear in his garage and was modifying some of my tube stuff.  He was really good.  One day when he was dropping off a piece of gear of mine that he had modified, He said “Eric I made a Fairchild”.  He brought in this box that looked like a damn fairchild!!!  I was blown away.  The design he came up with was altered to work with 6BC8 tubes instead of the 6386s that were not being made at the time.  It struck me that, this is such a cool circuit it just seems crazy to have it be so unobtainable.  We built the original 10 with his design and then JJ announced that they were gonna make 6386s.  That was when it got serious.  Larry got involved and we redid the entire design so it would be manufacturable and work with the JJ tubes.  Now we are almost 7 years down the road and have over 500 of them sold.

At the time it was originally sold, it was basically pitched as the new standard for level control, especially for producing vinyl phonograph masters. Do you still see it primarily as a compressor/limiter or is it more about the mojo and color today?

I suppose the appeal of a Fairchild circuit today is primarily the “Mojo and Color” which, at the end of the day, is just varying amounts of harmonic distortion.  There are plenty of compressors available now that can generate the same compression attack and release times as a Fairchild.  Ironically, what was considered to be “the new standard” in the 60s is totally archaic by today’s standards.  We know much better now how a compressor is used for music production and what features we need.  We added all of that flexibility and control to the UnFaircihld  so it could do all of the things we expect from a modern compressor AND have all of that wonderful “mojo and color”.  That I suppose is the point of still building a Fairchild circuit.  There is nothing else out there that sounds quite like a Fairchild and it is a stunningly  beautiful sound.

And now, given the open sidechain design of the UnFairchild, it is easily possible to run it feed forward as well. How does that turn out and how would you describe the difference in sound?

Yeah, the Feed Foward  thing was something we discovered after the fact.  We had the side chain insert in there and then realized that if you split  the incoming signal, you can send it to the audio inputs  and the side chain return, and Voila! You got feed forward.  It’s a cool sound.  It essentially gives the compressor  a very extreme ratio.  It is so extreme that it will  fall off a cliff and just turn off the sound.  The resulting sound is more reminiscent of a VCA type compressor, like the really punchy grabby sound of a DBX 160.  It is really incredible on drums, and in some cases, you can almost use it like a transient designer.  It’s just really cool to get that type of aggressive compression with the color of a Fairchild circuit.

You once mentioned that compression and distortion should always go together. Do you see both as two sides of the same coin or how exactly is that to be understood?

Compression by nature, can make sounds feel more restrained and constricted.  The distortion has a more expansive quality because it adds harmonic overtones to the dynamics.  The two balance each other beautifully.  As the compression is pulling back on the volume, the distortion is adding overtones and opening up the sound.

Digital solutions in software have come a long way – do you now see them as an alternative or a complement in this area?

Plugins have definitely become indispensable  for me.  There are certain things they can do that hardware cannot.  Those are the main things I really love them for.  plugins like oeksound  soothe are total game changers and there is nothing in the analog world that comes close to what it does.  I have the luxury of a healthy collection of analog outboard gear so I tend to use those when I’m looking for analog compression or color.  The plugins are getting really good and they are amazing if you don’t have the original boxes  available to use.  The plugins are definitely close enough to get the feel of those classic boxes.

So, will we eventually see an UnFairchild as software then?

I’m working on it now 🙂

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Related Links

• UnderToneAudio
• interview series (1) – Fabien from TDR
• interview series (2) – Nico from BigTone
• interview series (3) – Tony from Klanghelm
• interview series (4) – Bob Olhsson
• interview series (5) – Dave Hill
• interview series (6) – Christopher Dion
• interview series (7) – Dave Gamble
• interview series (8) – Sascha Eversmeier
• interview series (9) – D.W. Fearn
• interview series (10) – Vladislav Goncharov
• interview series (11) – Andreas Eschenwecker
• interview series (12) – Daniel Weiss

quote of the day

Compression and distortion should always go together.
– Eric Valentine

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.

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.

Dynamic 1073/84 EQ curves?

Yes we can! The 1073 and 84 high shelving filters are featuring that classic frequency dip upfront the HF boost itself. Technically speaking they are not shelves but bell curves with a very wide Q but anyway, wouldn’t it be great if that would be program dependent in terms of expanding and compressing according to the curve shape and giving a dynamic frequency response to the program material?

Again, dynamic EQs makes this an easy task today and I just created some presets for the TDR Nova EQ which you can copy right from here (see below after the break). Instructions: Choose one of the 3 presets (one for each specific original frequency setting – 10/12/16kHz) and just tune the Threshold parameter for band IV (dip operation) and band V (boost operation) to fit to the actual mix situation.

They sound pretty much awesome! See also my Nova presets for the mixbus over here and the Pultec ones here.

Kotelnikov GE – mastering

Here is my go-to mastering preset for Kotelnikov GE. Just change the threshold and you are there.

<TDRKotelnikovGE thresholdParam=”-24.0″ peakCrestParam=”-3.0″ softKneeParam=”6.0″ ratioParam=”3.0″ attackParam=”6.0″ releasePeakParam=”20″ releaseRMSParam=”300″ makeUpParam=”0.0″ dryMixParam=”off” outGainParam=”0.0″ keyHPFrequencyParam=”60″ keyHPSlopeParam=”6.0″ keyStereoDiffParam=”80″ keyStereoBalanceParam=”Center” fdrVisibleParam=”On” fdrActiveParam=”On” fdrTypeParam=”Shelf A” fdrFrequencyParam=”50″ fdrAmountParam=”80″ yingParam=”On” yangParam=”Off” deltaParam=”Off” bypassParam=”Off” equalLoudParam=”Off” qualityParam=”Insane” modeParam=”Stereo” grDispScaleParam=”4″ grDispModeParam=”Gain Reduction”/>

released: SlickHDR

SlickHDR is a “Psychoaccoustic Dynamic Processor” which:

• balances the perceived global vs. local micro dynamics of any incoming audio.
• creates a rich in contrast, detailed and clearly perceived image which translates way better across different listening environments.
• provides a convenient workflow by simply adjusting three dynamic processors to show a roughly same load.
• offers further and detailed control about overall tone and release time behavior.

The stunning UI artwork and all renders were done by Patrick once again. Made with love in switzerland – as he said!

SlickHDR is a freeware VST audio plug-in for Windows x32 and you can download a copy right here: >>> DOWNLOAD <<<

Related Links

• SlickHDR – final teaser & release info
• that’s why they call me slick
• processing with High Dynamic Range (1)
• processing with High Dynamic Range (2)
• processing with High Dynamic Range (3)

SlickHDR – final teaser & release info

SlickHDR is a “Psychoaccoustic Dynamic Processor” which:

• balances the perceived global vs. local micro dynamics of any incoming audio.
• creates a rich in contrast, detailed and clearly perceived image which translates way better across different listening environments.
• provides a convenient workflow by simply adjusting three dynamic processors to show a roughly same load.
• offers further and detailed control about overall tone and release time behavior.

Technically speaking, SlickHDR contains a coupled network of three dynamic processors with two of them running in a “stateful saturation” configuration and one based on look-ahead processing.

Fixed amounts of the unprocessed signal are then injected into the network at several specific points and also mixed back into the networks output. Being networked, all processors are highly interacting with each other and this is utilized to cope with a wide variety of sound (sic!) to balance the perceived audio dynamic range.

The stunning UI artwork and render was done by Patrick once again. Made with love in switzerland – as he said.

SlickHDR will be available around end of January 2014 as a freeware VST audio plug-in for Windows x32.

processing with High Dynamic Range (3)

This article explores how some different HDR imaging alike techniques can be adopted right into the audio domain.

The early adopters – game developers

In the lately cross-linked article “Finding Your Way With High Dynamic Range Audio In Wwise” some good overview was given on how the HDR concept was already adopted by some game developers over the recent years. Mixing in-game audio has its very own challenge which is about mixing different arbitrary occurring audio events in real-time when the game is actually played. Opposed to that and when we do mix off-line (as in a typical song production) we do have a static output format and don’t have such issues of course.

So it comes as no surprise, that the game developer approach turned out to be a rather automatic/adaptive in-game mixing system which is capable of gating quieter sources depending on the overall volume of the entire audio plus performing some overall compression and limiting. The “off-line mixing audio engineer” can always do better and if a mix is really too difficult, even the arrangement can be fixed by hand during the mixing stage.

There is some further shortcoming and from my point of view that is the too simplistic and reduced translation from “image brightness” into “audio loudness” which might work to some extend but since the audio loudness race has been emerged we already have a clear proof how utterly bad that can sound at the end. At least, there are way more details and effects to be taken into account to perform better concerning dynamic range perception.