released: SlickEQ

TDR SlickEQ main flat

TDR VOS SlickEQ is a mixing/mastering equalizer designed for ease of use, musical flexibility and impeccable sound.

Three (and a half) filter-bands arranged in a classic Low/Mid/High semi parametric layout offer fast and intuitive access to four distinct EQ modes, each representing a set of distinct EQ curves and behaviors. An elaborate auto gain option automatically compensates for changes of perceived loudness during EQ operation. Optionally, SlickEQ allows to exclusively process either the stereo sum or stereo difference (i.e. “stereo width”) without additional sum/difference encoding.

In order to warm up the material with additional harmonic content, SlickEQ offers a switchable EQ non-linearity and an output stage with 3 different saturation models. These options are meant to offer subtle and interesting textures, rather than obvious distortion. The effect is made to add the typical “mojo” often associated with classy audio gear.
An advanced 64bit multirate processing scheme practically eliminates typical problems of digital EQ implementations such as frequency-warping, quantization distortion and aliasing.

Beside the primary controls, the plug-in comes with an array of additional helpers: Advanced preset management, undo/redo, quick A/B comparison, copy & paste, an online help, editable labels, mouse-wheel support and much more.

SlickEQ is a collaborative project by Variety Of Sound (Herbert Goldberg) and Tokyo Dawn Labs (Vladislav Goncharov and Fabien Schivre).

Key specs and features

  • Intuitive, yet flexible semi parametric EQ layout
  • Full featured, modern user interface with outstanding usability and ergonomics
  • Carefully designed 64bit “delta” multi-rate structure
  • Three EQ bands with additional 18dB/Oct high-pass filter
  • Four distinct EQ models: “American”, “British”, “German” and “Soviet” with optional non-linearity
  • Four output stages: “Linear”, “Silky”, “Mellow” and “Deep”
  • Advanced saturation algorithms by VoS (“stateful saturation”)
  • Highly effective and musically pleasing loudness compensated auto gain control
  • Oversampled signal path including stateful saturation algorithms
  • Stereo and sum/difference processing options
  • Tool-bar with undo/redo, A/B, advanced preset management and more

Availability

TDR VOS SlickEQ is a freeware audio plug-in available for Windows and Mac in VST and Audio Units format (both 64-bit and 32-bit). VST3 and AAX formats will follow later.

All downloads are available via the Tokyo Dawn Labs website.

Related Links

SlickEQ – some more release info

Just a couple of days ago we introduced the upcoming release of SlickEQ and lots of questions raised already. So, here is what Fabien already committed about it in a public forum:

  • Win/Mac, AU/VST2/VST3 (+AAX planned and in process), x32/x64
  • No linux builds planned, sorry.
  • The name is “TDR VOS Slick EQ” and it will be available for free.
  • Release is a matter of days. Maybe a week or two.

As of today I just want to add: With the introduction of TDR VOS SlickEQ, quite a number of amazing and previously unheard DSP algorithms will see the light of day – including (but not limited to) several Stateful Saturation algorithms running within an audio signal path entirely upsampled to a constant high sample rate for maximum precision.

Expect smoothness, best-in-class.

Related links:

RescueMK2 2.1 update available

RescueMK2 [Read more…]

working ITB at higher sampling rates

Recently, I’ve moved from 44.1kHz up to 96kHz sampling rate for my current production. I would have loved to do this step earlier but it wasn’t possible with the older DAW generation in my case. With the newer stuff I was easily able to run a 44.1kHz based production with tons of headroom (resource wise – talking about CPU plus memory and disk space) and so I switched to 96kHz SR and still there is some room left.

I know there is a lot of confusion and misinformation floating around about this topic and so this small article is about to give some theoretical insights from a developer perspective as well as some hands-on tips for all those who are considering at what SR actually to work at. The title already suggests working ITB (In The Box) and I’ll exclude SR topics related to recording, AD/DA converters or other external digital devices. [Read more…]

the side effects of intermodulation in audio processors

typical IM distortion in a digital compressor

The general and most obvious effect of intermodulation components in audio signals is distortion of course – hence the concept of “intermodulation distortion” (aka “IM distortion” or simply “IMD”). IM distortion and harmonic distortion are two pairs of shoes and must be defined individually as already shown in the short essay about “myth and facts about aliasing” but more on this later on.

The existence of intermodulation components can affect the performance of an audio production in various ways. In the best case, IMD components are a desired artistic effect e. g. to obtain heavily crushed audio effect signals but in the worst and rather common case, they are one of the contributing factors which deteriorate the overall audio quality and might ruin a production. [Read more…]

preFIX 1.0 – out now!

preFIX – getting those alignments done

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preFIX – final teaser and release info

preFix

preFIX - gate and expander section with detailed sidechain fitering options

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towards stateful saturation

the static waveshaper y = tanh(x)

Still today, most developers are sticking to static waveshaping algorithms when it comes down to digital saturation implementations. This wasn’t very convincing to me from the very beginning and in fact it was one of the motivations why I’ve started my own audio effect developments – to come a little bit closer to what I thought what saturation and non-linearity in general is all about.

And so the Rescue audio plug-in was born in summer 2007 and was already an approach to relate audio transient events to the signal saturation itself. Not that much later TesslaSE appeared which was a different exercise leaning towards a frequency dependent non-linearity implementation coupled in a feedback structure. I still really love this plug-in and how it sounds and prefer it over much more sophisticated designs even today in quite some cases. Following, the pre-amp stage in BootEQmkII then focused on “transformer style” low-end weirdness and did feature oversampling on the non-linear sections of the device. A really great combination with the EQ – smooth and very musical sounding. The TesslaPRO thingy sums up all this and puts it into one neat little device with an easy to use “few knob” interface. Don’t let you fool by this simplistic (but so beautiful) design: It already features everything which makes a saturator to stand out from the crowd today: transient awareness, frequency dependency, dedicated low-end treatments. Sound-wise this results in a way smoother saturation experience and a better stereo imaging en passant.

With FerricTDS not only the notion of  subtle frequency dependent compression got extended to a core saturator algorithm. Since revision 1.5 I’ve ditched the oversampling based core and included a version which premiered the notion of memory into the non-linearity which transforms it from a stateless into a stateful algorithm. One could basically see this as a system which reacts different on the very same actual input signal depending on the recent history of events (on a very microscopical level). The input stage algorithms which I’ve included in NastyVCS and NastyDLA (both are actually the same) are a cpu and feature wise stripped down version of that to have the basic sound of it already as an option when mixing the tracks and its according fx.

Quite recently, I’ve started to look into implicit stateful models where memory is not applied from the outside of the algorithm but the algorithm itself contains a sort of memory. As an example, I’ve implemented a stateful version of the well-known tanh() function so that it is aware of recently occurred events but provides the very same harmonic structure compared to the original. Given some analyzer plots it even shows the very same transfer curve but in fact it does not limit strictly anymore but allows some minor overshots of some peak signals. Interestingly, the sound appears a little bit brighter (without letting you see that through the analyzer plot) and the low-end appears not to be that hard “brickwalled” but a little bit smoother. Lets be assured that I’m going to follow this path and then lets see where this will lead to in 2011.

applying saturation through the sidechain

This short article gives a brief introduction on applying waveshaping algorithms not directly in the audio path but via a transformed equation through the sidechain and a VCA instead.

Typical saturation curves

Typical saturation curves

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