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.

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.

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.

BootEQ mkIII released

BootEQ mkIII – a musical sounding Preamp/EQ

BootEQ mkIII is a musical sounding mixing EQ and pre-amplifier simulation. With its
four parametric and independent EQ bands it offers special selected and musical
sounding asymmetric and proportional EQ curves capable of reproducing several
‘classic’ EQ curves and tones accordingly.

It provides further audio coloration capabilities utilizing pre-amplifier harmonic distortion as well as tube and transformer-style signal saturation. Within its mkIII incarnation, the Preamp itself contains an opto-style compression circuit providing a very distinct and consistent harmonic distortion profile over a wide range of input levels, all based now on a true stateful saturation model.

Also the EQ curve slopes has been revised, plugin calibration takes place for better gain-staging and metering and the plugin offers zero latency processing now.

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

TesslaSE mkII released

TesslaSE mkII – All the analog goodness in subtle doses

TesslaSE never meant to be a distortion box but rather focused on bringing all those subtle saturation and widening (side-) effects from the analog right into the digital domain. It sligthly colors the sound, polishes transients and creates depth and dimension in the stereo field. All the analog goodness in subtle doses. It’s a mixing effect intended to be used here and there where the mix demands it. It offers a low CPU profile and (almost) zero latency.

With it’s 2021 remake, TesslaSE mkII sticks to exactly that by 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 to appear more detailed and vibrant, while all non-linear algorithms got oversampled for additional aliasing supression.

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

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.

ThrillseekerXTC mkII released

ThrillseekerXTC – bringing mojo back

ThrillseekerXTC mkII is a psychoacoustic audio exciter based on a parallel dynamic equalizer circuit. It takes our hearing sensitivity into account especially regarding the perception of audio transients, tonality and loudness.

The mkII version now introduces:
• Plugin operating level calibration for better gainstaging and output volume compensated processing.
• A reworked DRIVE/MOJO stage featuring full bandwidth signal saturation and a strong
focus on perceived depth and dimension. It provides all those subtle qualities we typically associate with the high-end analog outboard gear.
• Special attention has been taken to the mid frequency range by introducing signal compression which improves mid-range coherence and presence.
• Relevant parts of the plugin are running at higher internal sampling frequencies to minimize aliasing artifacts.

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

FerricTDS mkII released

FerricTDS mkII – the updated award winning Tape Dynamics Simulator

New in version 2:

• Introducing operating level calibration for better gainstaging and output volume compensated processing
• Metering ballistics revised and aligned accordingly
• Updated tape compression algorithms increasing punch, adding 2nd order harmonic processing, less IMD
• Updated limiter algorithm featuring ADC style converter clipping
• All non-linearities are running at higher sampling frequencies internally
• Adding a sophisticated analog signal path emulation

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

tips&tricks with SlickEQ

Note: Some of the tips rely on features from the GE version.

Mixing against HP/LP combo

A good generic practice when EQing several tracks in a mix is too start by dialing in HP/LP combinations by an  appropriate level and then do further EQing/mixing against those settings. Also using the tilt filter is a good idea to apply very first and rough tonal corrections and then working out the details afterwards with the three EQs.

Preserving low-end energy when high-pass filtering

A cool trick to preserve some low-end energy when high-pass filtering is applied is to boost the low-end while using the EQ-SAT feature. As you can see in the routing diagram the HPF comes after the main EQs and EQ-SAT. This way, harmonic overtones are generated based on the fundamentals before the HPF is applied.

Decoupling the low-end

The low-end EQ features a “Phi” option switch which allows to decouple the low-end by an allpass filter network. The crossover can be freely adjusted with the normal frequency control in this band while the gain control does not have any effect in this mode. This may work great for that mellow bass drums just as an example but in other cases it might loose some definition as a trade-off.

Compare different settings

SlickEQ contains two effect settings slots, A and B. Use them in combination with the automatic output gain control to AB test different settings. Within the plugin you can move settings between A and B but also copy&paste is there to freely copy settings between different plug-in instances. Also, undo/redo comes in handy here.

Adjusting precise values

The gain/frequency displays can also be used to enter specific values and also shortcuts are accepted, e.g. “5k” can be entered to set a value to 5000. And did you know that SlickEQ has mouse-wheel support?

 

 

out now: SlickEQ “Gentleman’s Edition”

Key specs and features

• Modern user interface with outstanding usability and ergonomics
• Carefully designed 64bit “delta” multi-rate structure
• Three semi-parametric filter bands, each with two shape options
• Five distinct EQ models: American, British, German, Soviet and Japanese
• Low band offers an optional phase-lag able to delay low frequencies relative to higher frequencies
• High pass filter with optional “Bump” mode
• Low pass filter with two different slopes (6dB/Oct and 12dB/Oct)
• Parametric Tilt filter with optional “V” mode.
• Six output stages: Linear, Silky, Mellow, Deep, Excited and Toasted
• Advanced saturation algorithms by VoS (“Stateful saturation”)
• Highly effective loudness compensated auto gain control
• Stereo, mono and sum/difference (mid/side) processing options
• Frequency magnitude plot
• Tool-bar with undo/redo, A/B, advanced preset management and more

SlickEQ is a collaborative project by Variety of Sound (Herbert Goldberg) and Tokyo Dawn Labs (Vladislav Goncharov and Fabien Schivre). For more details, please refer to the official product page: http://www.tokyodawn.net/tdr-vos-slickeq-ge/

Related

• announcing SlickEQ “Gentleman’s Edition”