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

interview series (10) – Vladislav Goncharov

Vlad, what was your very first DSP plugin development, how did it once started and what was your motivation behind?

My first plugin was simple a audio clipper. But I decided to not release it. So my first public released plugin was Molot compressor. I was a professional software engineer but with zero DSP knowledge (my education was about databases, computer networks and stuff like that). I played a guitar as a hobby, recorded demos at home and one day I found that such thing as audio plugins exist. I was amazed by their amount and also by the fact that there are free plugins too. And I realised that one day I can build something like this myself. I just had to open a DSP book, read a chapter or two and it was enough to start. So my main motivation was curiosity, actually.

Was that Molot compressor concept inspired by some existing devices or a rather plain DSP text book approach?

That days there was a rumour that it’s impossible to make good sounding digital compressor because of aliasing and stuff. I tried to make digital implementation as fluid as possible, without hard yes/no logic believing this is how perfect digital compressor should sound. And the way I implemented the algorithm made the compressor to sound unlike anything I heard before. I didn’t had any existing devices in my head to match and I didn’t watch textbook implementations too. The sound was just how I made it. I did 8 versions of the algorithm trying to make it as usable as possible from user point perspective (for example “harder” knee should sound “harder”, I removed dual-band implementation because it was hard to operate) and the last version of the project was named “comp8”.

Did you maintained that specific sound within Molot when you relaunched it under the TDR joint venture later on? And while we are at it: When and how did that cooperation with Fabien started?

TDR Molot development was started with the same core sound implementation as original Molot had. But next I tried to rework every aspect of the DSP to make it sound better but keep the original feel at the same time. It was very hard but I think I succeeded. I’m very proud of how I integrated feedback mode into TDR Molot for example. About Fabien: He wrote me to discuss faults in my implementation he thought I had (I’m not sure it was Molot or Limiter 6), we also discussed TDR Feedback Compressor he released that days, we argued against each other but what’s strange the next day we both changed our minds and agreed with our opposite opinions. It was like “You were right yesterday. No, I think you were right”. Next there was “KVR Developer Challenge” and Fabien suggested to collaborate and create a product for this competition. That was 2012.

And the Feedback Compressor was the basis for Kotelnikov later on, right?

No, Kotelnikov is 100% different from Feedback Compressor. Fabien tried to make the sound of feedback compressor more controllable and found that the best way to achieve this is just to change the topology to feedforward one. It’s better to say, Feedback Compressor led to Kotelnikov. Also the interesting fact, early version of Kotelnikov had also additional feedback mode but I asked Fabien to remove it because it was the most boring compressor sound I ever heard. I mean if you add more control into feedback circuit, it just ruins the sound.

Must have been a challenge to obtain such a smooth sound from a feed-forward topology. In general, what do you think makes a dynamic processor stand out these days especially but not limited to mastering?

I think, it’s an intelligent control over reactions. For example Kotelnikov has some hidden mechanisms working under the hood, users don’t have access to them but they help to achieve good sound. I don’t think it’s good idea to expose all internal parameters to the user. There must be hidden helpers just doing their job.

I so much agree on that! Do you see any new and specific demand concerning limiting and maximizing purposes? I’m just wondering how the loudness race will continue and if we ever going to see a retro trend towards a more relaxed sound again …

I think even in perfect loudness normalized world most of the music is still consumed in noisy environments. The processing allowing the quietest details to be heard and cut through background noise, to retain the feel of punch and density even at low volumes is in demand these days. Loudness maximizers can do all this stuff but in this context they act like old broadcast processors. In my opinion, the loudness war will continue but it’s not for overall mix loudness anymore but how loud and clear each tiny detail of the mix should be.

Can we have a brief glimpse on what you are currently focused on, DSP development wise?

You may take a look at Tokyo Dawn Labs Facebook posts. We shared a couple of screenshots some time ago. That’s our main project to be released someday. But also we work on a couple of dynamic processors in parallel. We set high mark on the quality of our products so we have to keep it that high and that’s why the development is so slow. We develop for months and months until the product is good enough to be released. That’s why we usually don’t have estimation dates of release.

Related Links

• Tokyo Dawn Labs

interview series (9) – D.W. Fearn

Doug, when and how did you arrived in the music business?

I have had an interest in electronics ever since I was a kid growing up in the 1950s and 1960s. I built a crystal radio  receiver when I was 8 and my first audio amplifier (tubes, of course) when I was 10. I passed the test for an amateur radio license when I was 12 and that experience of communicating using Morse code was excellent training for  learning to hear. I built a lot of my own radio equipment, and experimented with my own designs.

The high school I attended had an FM broadcast station. Most of the sports and musical events were broadcast, and I learned about recording orchestras, marching bands, choirs, and plays. Friends asked me to record their bands, which was my first experience working with non-classical music.

Another major factor was that my father was a French horn player in the Philadelphia Orchestra. As a kid, I would attend concerts, rehearsals, and sometimes recording sessions and broadcasts. I learned a lot about acoustics by walking around the Academy of Music in Philadelphia during rehearsals.

It would seem logical that my musical exposure and my interest in electronics would combine to make the career in pro audio I have had for over 40 years now.

I was a studio owner for many years before starting the D.W. Fearn manufacturing business, which started in 1993.

the sound of Gravity

In this exclusive SoundWorks Collection profile we talk with Director Alfonso Cuarón and Re-recording Mixer Skip Lievsay about the sound teams work to create a dramatic sound scape to a dark and vast outer space environment.

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a must see

And don’t miss to see the whole DTS “Sound Magician” Series.

the visual art of Brian Eno: Light and Time

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