Audio analyzers currently in use here

During tracking, mixing and mixdown I’m utilizing different analyzers whether thats freeware or commercial, hard- or software. Each of them doing a decent job in its very own area:

VU Meter

Always in good use during tracking and mixing mainly for checking channel levels and gainstaging all kinds of plugins. I also love to have a VU right on the mixbus to get a quick visual indication about Peak vs RMS dynamic behaviour.

TBProAudio mvMeter2 is freeware and actually meters not only VU but also RMS, EBU LU as well as PPM. It is also resizeable (VST3 version) and supports different skins.

Spectrum Analyzer I

To me, the Voxengo SPAN is an all-time classic analyzer and ever so reliable. I’ve always used it to have a quick indication about an instruments frequency coverage or the overall frequency balance on the mixbus. There is always one running at the very end of the summing bus in the post-fader section.

Voxengo SPAN is also freeware and highly customizable regarding the analyzer FFT resolution, slope smoothing and ballistics.

Spectrum Analyzer II

Another spectrum analyzer I’m using is Voxengo TEOTE which actually is not only an analyzer but a full spectrum dynamic processor. However, let alone the analyzer itself (fully working in demo mode!) is an excellent assistant when it comes to assess the overall frequency balance. The analyzer does this in regards to a full spectrum noise profile which is adjustable with a Tilt EQ, basically. Very handy for judging deviations (over time) from an ideal frequency response.

Voxengo TEOTE demo version available on their website.

Loudness Metering

I’m leaving all EBU R128 related business to the TC Electronic Clarity M. Since it is a hardware based monitoring solution it always is active here on my desktop no matter what and also serves for double-checking equal RMS levels (for A/B comparisions) and a quick look at the frequency balance from time to time. The hardware is connected via USB (could be SPDIF as well) and is driven by a small remote plugin sitting at the very end of the summing bus in my setup here. It also offers a vector scope and provides audio correlation information. It supports a vast variety of professional metering standards.

Courtesy of Music Tribe IP Ltd.

Image Courtesy of Music Tribe IP Ltd.

 

 

 

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

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.

sustaining trends in audio land, 2021 edition

Now, after spending some time on digging a little bit more deeper into the current offerings and market situation in audio production I just wanted to briefly outline some of my personal summaries regarding sustaining trends but maybe outline also some new things I do see on the horizon.

The mobile audio evolution

To me this indeed looks like an ongoing trend for years now which simply does not stop. On the one hand we can see the whole software and especially the App market continuing and increasing in all areas and platforms: notebooks, tablets, smartphones and their respective eco systems accordingly. Where Ableton once started in providing an almost complete mobile music production approach in literally just a bag, Bitwig and others followed and now Apps are everywhere allowing any kind of recording and music or media production on the go. Apples recent move with the M1 SOC (System on Chip) approach fits perfectly into this trend by increasing the mobility even further in terms of power, size and efficiency. Others will follow this path for sure. Also we can see traditional music gear manufacturers going more and more into compact and battery powered solutions as well, such as the Korg Volca series or the Roland boutique thingies, just to name the two.

The retro cult continues

Companies like Behringer will continue to spit out analog HW clones like there is no tomorrow. Whether thats synthesizer reissues or blatant plain copies of vintage mixing outboard or modeled software – you’ll find everything and in almost all shades of quality and price. And I think this is a really good thing to have such a variety to choose from and also this will lead to some serious price drops in the overpriced used gear market in that area.

Modular madness

I don’t think this is part of the overall retro trend but a niche on its very own. In any case the modular synthesis thing is still gaining more and more momentum. There is a sheer amount of hardware options to choose from and meanwhile also quite a lot of audio interfaces and controller solutions are offering not only Midi but also CV support. Even in software land one can put his/her virtual hands on something modular. All in all, this looks and sounds like real fun and a great opportunity to spend a lot of time on (and money).

Look mom no computer

All those neat outboard DAW-less setups shown on YT: Some hardware samplers and grooveboxes here, some fancy retro synths there and fx stomp boxes all over the place. Well, “Look mom no computer” is of course absolutely wrong here because half of that stuff has tiny little digital displays and computers underneath you have to tinker with. Personally, I would prefer some neat “one knob, one job” analog interfaces plus a real full-blown DAW any day. However, definately a sustaining trend and a good thing.

Loudness war, quo vadis?

While it seems that LUFS finally made it and in fact has been successfully settled as a standard in the broadcast domain – in music production in general it has not. Todays audio mastering target levels are still insane and even some “engineers” continue to present converter clipping as the holy loudness grail to their YT followers. That really hurts. At least some of the big streaming sevices restricted target loudness levels to -14 or -16LUFS which gives a little hope.

ITB production finally took over

Now that even the last renowned mixing engineer has finally surrendered to the dark side in the box – at least for the recording and mixing part – the question remains, why this has taken so long. Was it for quality concerns? The time-to-market pressure to finally have total recall in all regards? Simple ignorance or fear? We might not be sure about the final answer but we do know that today almost everybody can run some media production tasks in a decent quality on his very own while having a low entrance barrier. And this is what I really would call the “game changer” of the last decade. Now, your skills are the limit.

Game of DAWs

There is really no trend in particular here other than the fact that we have the very same players on the board since a decade ago. Maybe Bitwig will aim for the crown from Ableton? It’s whole inherent synthesis and modulation integration make this comprehensive sequencer an instrument on its very own and also it runs natively on Linux. All the other contenders improved step by step here and there but quite comparable. Maybe having build in mixing scenes and more convincing analog style summing is a thing which sticks a little bit out. So, on my own I wasn’t that much impressed about this very last episodes and now I’m looking forward to an upcoming but much more entertaining season, hopefully.

The pandemic impact

As we all know, the Covid impact on everything live performance related was and still is a sheer desaster. How this will evolve in the future is hard to predict but it is clear that there won’t be any back to normal any time soon if ever. That means this area must transform into the digital/virtual domain as well and most of the suppliers in exact this kind of areas are already the winners of the current situation.

Stay healthy!

 

out now: SlickEQ “Gentleman’s Edition”

SlickEQ_German

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

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

interview series (4) – Bob Olhsson

Bob, you are a professional recording and mastering engineer for over thirty years and already legend. What is more important: The ability to hear or the ability to use the right device according to the context and to apply the right adjustment?

You need to BOTH be able to hear AND find the right device and settings! Probably the most important thing to understand is that just raising the volume a tenth of a dB. will always sound better. All comparisons must be checked with the average level compensated. Otherwise, it’s easy to wind up with something that’s louder but far worse sounding.

Another challenge is mastering for what the recording needs and not what your monitors want. Our goal is for it to sound great everywhere. The best way I’ve found to tackle this problem is sticking to broad strokes unless I’m removing distractions. If a half dB. sounds different but not better, I’ll generally leave it alone. [Read more…]

what Psychoacoustics is about

If you once listened to MPEG 1 Audio Layer 3 (aka mp3) compressed audio files you’ve already heard some encoding algorithms which are highly based on psychoacoustic principles. But what is that voodoo stuff all about in general? The science of psychoacoustics basically investigates the impact of a physical audio signals property on to the subjective perceived signal (hearing) and whats going on under the hood of that hearing, meanwhile. So it can basically be seen as an input/output model where a certain acoustic stimuli comes in and some sort of perception comes out. On top of that, hearing has not just been seen as a black box but a lot of modelling has already been done to get a better understanding on how hearing is actually performed in our bio-mechanical apparatus and our brain as well. [Read more…]

loudness wars – episode IV

Yes, a new hope. While some of the recently established  metering systems did not successfully managed the loudness race problems in general there seems to be a new hope concerning those issues and this comes from the broadcasters standardization efforts. Started in 2006 the ITU recommendation BS.1770­‐1 defined already some replacement for the common QPPM metering and instead was oriented towards loudness metering. [Read more…]

about audio signal coloration

In this comprehensive article some deeper explorations and explanations on this topic are given and at the end a brief but handy definition about audio signal coloration is proposed.  Some tips on mixing can be obtained here as well and – by the way – some myth about equalizing audio in the digital domain gets busted.

Digital image spectrum

Digital imaging spectrum

[Read more…]