quote of the day

Whatever you now find weird, ugly, uncomfortable and nasty about a new medium will surely become its signature. CD distortion, the jitteriness of digital video, the crap sound of 8-bit – all of these will be cherished and emulated as soon as they can be avoided. It’s the sound of failure: so much modern art is the sound of things going out of control, of a medium pushing to its limits and breaking apart.
― Brian Eno, 1996

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

200ms

1/5 of a second – that’s not much time, is it? And yet this is exactly the crucial time in which our hearing not only captures but also already processes very essential information after an audible event has been detected. Likewise, this is about hearing-related limiting factors that come to light precisely in this short period of time. Lets have a look at some examples:

Up to 20ms – Pre masking effects (also called backward masking): Signals can be masked during this time by a subsequent, stronger signal.

Up to ~40ms – The well known Haas effect: The time region from which two signals can no longer be distinguished as separate audio events.

Up to ~50ms – Audio transient events: The typical acoustic instrument onset / percussive audio events, before something is perceived as a steady state signal.

~10-60ms – Pitch detection: The time span needed to convey a clear sense of pitch.

~40-200ms – Sound localization: Events in this time range can give the auditory system important clues about the location of the event, e.g. by early reflections.

Up to 200ms duration – Post masking effects (also called forward masking): The time span a masking effect can last after the mask itself disappears. The effect shows a linear decay (over a logarithmic time scale) for exactly this time.

Up to 200ms – Masking, Perceived loudness: Signals shorter than 200ms can be masked much more easily by a steady noise signal or, in other words, for shorter signals the perceived loudness decreases (by ~10dB/decade below 200ms).

Above 200ms –  Perceived loudness: Above this duration rather steady state conditions appear concerning perceived loudness as if the hearing is a sort of averager with an integration time of 200ms.

The effects are of course immense and relevant for very different areas, especially in mixing and mastering, but also already during recording. It also makes clear why certain phenomena cannot be shown with simple audio measurement devices. We must rely on our hearing and, if necessary, train it accordingly, there is no way around it.

lets talk about multi-channel production

Multi-channel production has been pushed strongly again for some time, and not only by Dolby and Apple. But what does “multi-channel” actually mean in your music production? Is it already relevant for recording and mixing or rather a downstream production step? Does it play a role at all or is it irrelevant for you as a music producer?

the world of sound localization according to psychoacoustics

Sound localization refers to the ability of the human auditory system to determine the location of a sound source in space. This is done by analyzing the differences in the arrival time, intensity, and spectral content of the sound waves that reach the two ears. The human ear is able to localize sounds both horizontally (azimuth) and vertically (elevation) in the auditory space.

The brain processes the incoming sound signals from both ears to calculate the interaural time difference (ITD) and interaural level difference (ILD), which are used to determine the location of the sound source. Interaural time difference refers to the difference in the time it takes for a sound wave to reach each ear, while interaural level difference refers to the difference in the level of the sound wave that reaches each ear.

The auditory system uses both ITD and ILD as complementary cues that work together to allow for accurate sound localization in the horizontal plane, aka stereo field. For example, sounds coming from straight ahead might have similar ITDs at both ears but different ILDs, while sounds coming from the side might have similar ILDs at both ears but different ITDs.

It’s also worth noting that the relative importance of ITD and ILD can vary depending on the frequency of the sound. At low frequencies, ITD is the dominant cue for sound localization, while at high frequencies, ILD becomes more important. Research has suggested that the crossover frequency between ILD and ITD cues for human sound localization is around 1.5 kHz to 2.5 kHz, with ITD cues being more useful below this frequency range and ILD cues being more useful above this range.

In addition to ITD and ILD, the auditory system also uses spectral cues, such as the shape of the outer ear and the filtering effects of the head and torso, to determine the location of sounds in the vertical plane and also to identify backside audio events.

The temporal characteristics of an audio event, such as its onset and duration, can have an impact on sound localization as well. Generally speaking, sounds with a more distinct onset, such as a drum hit, are easier to localize than sounds with a more sustained signal, such as white noise. This is because the onset of a sound provides a more salient cue for the auditory system to use in determining the location of the sound source, especially in regards to ITD.

In the case of a drum hit, the sharp onset creates a more pronounced difference in the arrival time and intensity of the sound at the two ears, which makes it easier for the auditory system to use ITD and ILD cues to locate the sound source. In contrast, with a more sustained signal like white noise, the auditory system may have to rely more on spectral cues and reverberation in the environment to determine the location of the sound source.

If a tree falls in the forest and no one there to hear it, does it make a sound?

The question of whether a tree falling in the forest makes a sound if no one is there to hear it is a classic philosophical conundrum. It raises the question of whether sound exists independently of our perception of it.

One view is that sound is a physical phenomenon that requires the vibration of particles in a medium (such as air or water) to transmit sound waves. According to this view, if a tree falls in the forest and there is no one or nothing there to detect the sound waves, the tree would indeed make a sound, but there would be no one or nothing there to hear it.

On the other hand, some philosophers argue that sound is a purely subjective experience that requires the presence of a conscious observer. According to this view, if there is no one or thing present to perceive the sound waves, then there can be no sound.

Ultimately, of course, the answer to this question depends on how you define ‘sound’ and whether you see it as a purely subjective or objective phenomenon, but as music makers we are well advised to understand ‘sound’ on all levels and in all its facets.

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.

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.

Related Links:

• • the beauty of opto-electrical compression – volume 2

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.

interview series (11) – Andreas Eschenwecker

Andy, your Vertigo VSC compressor has already become a modern classic. What has been driven you to create such a device?

I really like VCA compressors. VCA technology gives you a lot of freedom in design and development and the user gets a very flexible tool at the end. I was very unhappy with all VCA compressors on the market around 2000. Those were not very flexible for different applications. These units were working good in one certain setting only. Changing threshold or other parameters was fiddley and so on. But the main point starting the VSC project was the new IC VCA based compressors sounded one dimensional and boxy.

Does this mean your design goal was to have a more transparent sounding device or does the VSC also adds a certain sound but just in a different/better way?

Transparency without sounding clean and artificial. The discrete Vertigo VCAs deliver up to 0,6% THD. Distortion can deliver depth without sounding muddy.

Does this design favour certain harmonics or – the other way around – supresses some unwanted distortions?

The VSC adds a different distortion spectrum depending when increasing input level or adding make-up. The most interesting fact is that most of the distortion and artifacts are created in the release phase of the compressor. The distortion is not created on signal peaks. It’s becoming obvious when the compressor sets back from gainreduction to zero gainreduction. Similar to a reverb swoosh… after the peak that was leveled.

Where does your inspiration comes from for such technical designs?

With my former company I repaired and did measurements on many common classic and sometimes ultra-rare compressors. Some sounded pretty good but were unreliable – some were very intuitive in a studio situation, some not…
At this time I slowly developed an idea what kind of compressor I would like to use in daily use.

From your point of view: To which extend did the compressor design principles changed over the years?

The designs changed a lot. Less discrete parts, less opto compressors (because a lot of essential parts are no longer produced), tube compressors suffer from poor new tube manufacturing and some designers nowadays go more for RMS detection and feed forward topology. With modern components there was no need for a feedback SC arrangement anymore. I think RMS is very common now because of its easy use at the first glance. For most applications I prefer Peak detection.

Having also a VSC software version available: Was it difficult to transfer all that analog experience into the digital domain? What was the challenge?

In my opinion the challenge is to sort out where to focus on. What influence has the input transformer or the output stage? Yes some of course. Indeed most of the work was going into emulating the detection circuit.

Which advantages did you experienced with the digital implementation or do you consider analog to be superior in general?

I am more an analog guy. So I still prefer the hardware. What I like about the digital emulations is that some functions are easy to implement in digital and would cost a fortune in production of the analog unit.

Any plans for the future you might want to share?

At the moment I struggle with component delays. 2021/22 is not the right time for new analog developments. I guess some new digital products come first.

Related Links

• Vertigo Sound