sound affairs & audio effect design
A public beta version of epicCLOUDS is available for testing. As already announced, the beta testing will be supported by the new VoS Discord server to gather feedback and discuss issues as best as possible. Feel free to join, its open for everyone!
Click here to join the VoS Discord server. A Discord login is required. Once on my server, you must agree the welcome message just shown to you by ticking it with a “thumbs up” to get full rights for posting etc.
I just asked this to ChatGPT and here is the answer:
As an AI language model, I don’t have access to real-time information about upcoming releases from specific companies or developers. However, it’s worth noting that “Variety Of Sound” is a well-known developer of high-quality audio plugins, with a particular focus on analog-style signal processing.
In the past, “Variety Of Sound” has released a range of plugins, including EQs, compressors, tape saturation emulators, and more. These plugins are known for their attention to detail, their intuitive interfaces, and their ability to add warmth and character to digital audio signals.
While I don’t have information about any specific upcoming releases from “Variety Of Sound”, it’s possible that they may continue to expand their range of analog-style plugins, or explore new areas of audio processing. As with any developer, it’s likely that they will release new products based on their own interests and expertise, as well as feedback and requests from their user base.
So, what about “feedback and requests” from your side?
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:
The mkII update is available for Windows VST in 32 and 64bit as freeware. Download your copy here.
Related Links:
When Cubase 3.0 came out in 1996 and introduced VST for the first time with all its new and fascinating possibilities, that was the point where I decided to get more involved in music production and set up a small (home) recording studio. VST was the basis for all this and how I imagined a modern (computer based) studio production. What a revolution that was. Watching this video today brings up a lot of nostalgic feelings …
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.
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.
Short review from Bedroom Producers Blog – Variety Of Sound Releases FREE FlavourMTC Passive Equalizer Plugin:
Variety of Sound is one of the legendary old-school freeware VST plugin developers, providing top-tier audio software that easily rivals the quality of paid plugins. Their incredible FerricTDS mkII is still my favorite tape saturation plugin.
If you’re looking for mixing and mastering software with a taste of analog hardware, I highly recommend checking out Variety of Sound’s plugins. It is one of the best freeware plugin collections for Windows-based music producers.
Thread over there at gearslutz gearspace – A completely new plugin after a long streak of redesigned classics:
This sounds like analogue gear. Incredible.
… it’s a big fat lump of ‘ashish dissolved in a bucket filled with two parts honey and one part roasted sesame on a warm Himalayan afternoon. Dip yer finger, don’t forget to lick it and soon not only sesame opens …
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.
First of all, congrats on your Technical Grammy Award this year! Daniel, you’ve once started DSP developments during the early days of digital audio. What was the challenge to that time?
Thank you very much, Herbert.
Yes, I started doing digital audio back in 1979 when I joined Studer-Revox. In that year Studer started their digital audio lab with a group of newly employed engineers. At that time there were no DSPs or CPUs with enough power to do audio signal processing. We used multiplier and adder chips from the 74 chip series and/or those large multiplier chips they used in military applications. The “distributed arithmetic” technique we applied. Very efficient, but compared to today’s processors very inflexible.
The main challenges regarding audio applications were:
Later the TMS32010 DSP from TI became available – a very compromised DSP, hardly useable for pro audio.
And a bit later I was able to use the DSP32 from AT&T, a floating point DSP which changed a lot for digital audio processing.
What makes such a converter design special in regards to audio and was the DSP math as we know it today already in place or was that also something rather emerging to that time?
The A/D and D/A converters back then had the problem that they either were not fast enough to do audio sampling frequencies (like 44.1 kHz) and/or their resolution was not high enough, i.e. not 14 Bits or higher.
There were some A/D and D/A modules available which were able to do digital audio conversion, but those were very expensive. One of the first (I think) audio specific D/A converters was the Philips TDA1540 which is a 14 bit converter but which has a linearity better than 14 bit. So we were able to enhance the TDA1540 by adding an 8 bit converter chip to generate two more bits for a total of about 16bits conversion quality.
The DSP math was the same as it is today – mathematics is still the same, right? And digital signal processing is applied mathematics using the binary numbering system. The implementation of adders and multipliers to some extent differed to today’s approaches, though. The “distributed arithmetic” I mentioned for instance worked with storage registers, shift registers, a lookup table in ROM and an adder / storage register to implement a complete FIR filter. The multiplication was done via the ROM content with the audio data being the addresses of the ROM and the output of the ROM being the result after the multiplication.
An explanation is given here: http://www.ee.iitm.ac.in/vlsi/_media/iep2010/da.pdf
Other variants to do DSP used standard multiplier and adder chips which have been cascaded for higher word-lengths. But the speed of those chips was rather compromised when comparing to today’s processors.
Was there still a need to workaround such word-length and sample rate issues when you designed and manufactured the very first digital audio equipment under your own brand? The DS1 compressor already introduced 96kHz internal processing right from the start, as far as I remember. What were the main reasons for 96kHz processing?
When I started at Studer the sampling frequencies have been all over the place. No standards yet. So we did a universal Sampling Frequency Converter (Studer SFC16) which also had custom built interfaces as those haven’t been standardized either. No AES/EBU for instance.
Later when I started Weiss Engineering the 44.1 and 48 kHz standards had already been established. We then also added 88.2 / 96kHz capabilities to the modular bw102 system, which was what we had before the EQ1, DS1 units. It somehow became fashionable to do high sampling frequencies. There are some advantages to that, such as a higher tolerance to non-linearly treated signals or less severe analog filtering in converters.
The mentioned devices were critically acclaimed not only by mastering engineers over the years. What makes them so special? Is it the transparency or some other distinct design principle? And how to achieve that?
There seems to be a special sound with our devices. I don’t know what exactly the reason is for that. Generally we try to do the units technically as good as possible. I.e. low noise, low distortion, etc.
It seems that this approach helps when it comes to sound quality….
And maybe our algorithms are a bit special. People sometimes think that digital audio is a no brainer – there is that cookbook algorithm I implement and that is it. But in fact digital offers as many variants as analog does. Digital is just a different representation of the signal.
Since distortion is such a delicate matter within the design of a dyncamic processor: Can you share some insights about managing distortion in such a (digital) device?
The dynamic processor is a level controller where the level is set by a signal which is generated out of the audio signal. So it is an amplitude modulator which means that sidebands are generated. The frequency and amplitude of the sidebands depend on the controlling signal and the audio signal. Thus in a worst case it can happen that a sideband frequency lies above half the sampling frequency (the Nyquist frequency) and thus gets mirrored at the Nyquist frequency. This is a bad form of distortion as it is not harmonically related to the audio signal.
This problem can be solved to some extent by rising the sampling frequency (e.g. doubling it) before the dynamic processing is applied, such that the Nyquist frequency is also doubled.
Another problem in dynamics processors is the peak detection. In high frequency peaks the actual peak can be positioned between two consecutive samples and thus get undetected because the processor only sees the actual samples. This problem can be solved to some extent by upsampling the sidechain (where the peak detection takes place) to e.g. 2 or 4 times the audio sampling frequency. This then allows to have kind of a “true peak” measurement.
Your recent move from DSP hardware right into the software plugin domain should not have been that much of a thing. Or was it?
Porting a digital unit to a plug-in version is somewhat simpler compared to the emulation of an analog unit.
But the porting of our EQ1 and DS1 units was still fairly demanding, though. The software of five DSPs and a host processor had to be ported to the computer platform. The Softube company did that for us.
Of course we tried to achieve a 1:1 porting, such that the hardware and the plugin would null perfectly. This is almost the case. There are differences in the floating point format between DSPs and computer, so it is not possible to get absolutely the same – unless one would use fixed point arithmetic; which we do not like to use for the applications at hand.
The plugin versions in addition have more features because the processing power of a computer CPU is much higher than the five (old) DSPs the hardware uses. E.g. the sampling frequency can go up to 192kHz (hardware: 96kHz) and the dynamics EQ can be dynamic in all seven bands (hardware: 4 bands maximum).
Looking into the future of dynamic processing: Do you see anything new on the horizon or just the continuation of recent trends?
We at Weiss Engineering haven’t looked into the dynamics processing world recently. Probably one could do some more intelligent approaches than the current dynamics processors use. Like e.g. look at a whole track and decide on that overview what to do with the levels over time. Also machine learning could help – I guess some people are working in that direction regarding dynamics processing.
From your point of view: Will the loudness race ever come to an end and can we expect a return of more fidelity back into the consumer audio formats?
The streaming platforms help in getting the loudness race to a more bearable level. Playlists across a whole streaming platform should have tracks in them with a similar loudness level for similar genres. If one track sticks out it does not help. Some platforms luckily take measures in that direction.
Daniel, do you use any analog audio equipment at all?
We may have a reputation in digital audio, but we do analog as well. A/D and D/A converters are mostly analog and our A1 preamp has an analog signal path. Plus more analog projects are in the pipeline…
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Currently on my desk, awaiting further analysis: The Manultec Orca Bay EQ
Rebuilding my studio and restarting blogging activities one year ago was pretty much fun so far. Best hobby ever! To get things started in Jan/Feb this year, I did a short summary about the recent trends in audio and I might revise and update that in January again. Quite some audio gear caught my attention over the year and some found its way into the Blog or even in my humble new studio setup, e.g. the unique SOMA Lyra-8 and the Korg MS-20 remake as well as the Behringer Clone of the ARP 2600.
I also went into more detail on how to get the most out of the SPL Tube Vitalizer or the renaissance of the Baxandall EQs just to name the two topics and also had a more realistic look at the Pultec style equalizer designs which might be something I will continue to dig into a little bit further in 2022. As of lately I’m also intrigued by some analog effect pedal designs out there, namely the Fairfield Circuitry stuff. And as always, I’m highly interested in everything psychoacoustic related.
By end of August I started re-releasing my very own plugins and also did mkII versions for FerricTDS, ThrillseekerXTC and TesslaSE. I will continue that route and on top of my list is to have the whole Thrillseeker plugin series complete and available again. Some are asking me if I will develop brand new audio plugins as well. While I’m doing that already but just for my very own, at this point in time it remains unclear if some of that stuff will ever gonna make it into a public release. But you never know, the TesslaSE remake was also not planned at all.
Something I will continue for sure is that special developer interview series I did over the years. This year I already had the chance to talk to Vladislav Goncharov from Tokyo Dawn Labs and Andreas Eschenwecker from Vertigo Sound which gave some detailed insights about creating analog and digital audio devices, especially dynamic processors. To be published in January, the very next interview has also been done already and this time it will be with this years Technical Grammy Award winner, Daniel Weiss.
I’m looking forward to 2022!
Stay tuned
Herbert
Variety Of Sound 
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