interview series (12) – Daniel Weiss

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:

  • A/D and D/A converters had to be designed with audio in mind.
  • Digital audio storage had to rely on video tape recorders with their problems.
  • Signal processing was hardware coded, i.e. very inflexible.
  • DAWs as we know them today have not been feasible due to the lack of speedy processors and the lack of large harddisks. (The size of the first harddisks started at about 10 MByte…).
  • Lack of any standards. Sampling frequencies, wordlengths and interfaces have not been standardized back then.

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…

Related Links

a brief 2021 blogging recap and 2022 outlook

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

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

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

64bit plugin rollout started, announcing mkII plugin versions

The very first 64bit plugin versions are out now, starting with the plugins from the public beta test earlier this summer: epicVerb, BaxterEQ, preFIX, NastyDLAmkII, NastyVCS and DensityMkIII. All versions have been carefully revised, are backwards compatible and some includes bugfixes and improvements as well. VST3 versions are not (yet) included due to stability issues. For further release notes and downloads please refer to the download page.

The remaining VoS plugins are planned to be (re-) released one-by-one until end of this year. Most likely, they will reappear as mkII versions 🙂 First one will be FerricTDS mkII which is already in the finishing process and to be released early September.

stay tuned!

A brief update on the future of VoS plugins

First of all: I hear you!

While receiving so many questions and requests on this topic on all the different channels these days a brief update here seems to be overdue.

Plugin downloads might be online again as soon as I manage to find a decent free cloud space to offer hassle-free downloads without any traffic/bandwidth limits (or advertising). Hints appreciated.

There are no plans to make them payware.

Some of my plugin designs needs to be revised to assure that they will stand the test of time for yet another decade 🙂 and this means also that they are going to support 64bit of course.

With some selected VoS plugins, there will be a 64bit public beta which is planned to be held this July. I hope you’ll gonna join.

The story continues, stay tuned!

Herbert

42 Audio Illusions & Phenomena

In a comprehensive series of five YouTube videos, Casey Connor provided an awesome overview and demonstration of 42 (!) different psychoacoustic effects. Watching and hearing (headphones required) not only is so much entertaining and educational but also provides some deep insights why we all do not hear in the exact same way. Relevant for all of us in the audio domain whether it is sound design, mixing, mastering or development. Highly recommended!

blogging VOS – 5th anniversary

Today – five years ago – I’ve started blogging about the Variety Of Sound venture. Your feedback was truly overwhelming: During 2013 this blog was viewed more than a million times and visitors were coming from over 200 countries in all. Lots came searching specific plug-ins but most traffic came from the vast number of reviews and top10/best-of lists around the net. Most attention this year went to the ThrillseekerVBL release around July 1st and most concern is about having 64bit versions available.

quote of the day

I’ve always been very interested in the technical side. And I’ve always wanted to get to the source of sound. So coding plugins gets me quite close. It wasn’t really a decision, more of a smooth transition. PlugIn coding almost replaced writing songs for me. It’s an equally great way to express myself and to create something, that might be enjoyable for others too. – Tony from Klanghelm

I could not have said this better myself! Read the full interview with Tony here in a couple of days.

what next?