sidechain linking techniques

How an audio compressor responds to stereo content depends largely on how the channel linking is implemented in the sidechain. This has a major influence on how the spatial representation of a stereo signal is preserved or even enhanced. The task of the compressor designer is to decide which technical design is most suitable for a given overall concept and to what extent the user can control the linkage when using the device.

In analog compressor designs, in addition to unlinked “dual mono” operation, one usually finds simple techniques such as summing both stereo channels (corresponding to the center of the stereo signal) or the extraction of the maximum levels of both channels using a comparator circuit implementing the mathematical term max(L,R).

More sophisticated designs improve this by making the linking itself frequency dependent, e.g. by linking the channels only within a certain frequency range. It is also common to adjust the amount of coupling from 0 to 100%, and the API 2500 hardware compressor serves as a good example of such frequency dependent implementation. For the low and mid frequency range, simple summing often works slightly better in terms of good stereo imaging, while for the mid to high frequency range, decoupling to some degree often proves to be a better choice.

The channel coupling can also be considered as a summation of vectors, which can be easily realized by sqrt(L^2+R^2). As an added sugar, this also elegantly solves the rectification problem and results in very consistent gain reduction across the actual level distributions that occur between two channels.

If, on the other hand, one wants to focus attention on correlated and uncorrelated signal components individually (both of which together make up a true stereo signal), then a mid/side decomposition in the sidechain is the ticket: A straight forward max(mid(L,R), side(L,R)) on the already rectified channels L and R is able to respond to any kind of correlated signal not only in a very balanced way but also to enhance its spatial representation.

More advanced techniques usually combine the methods already described.

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.

I just had to have this

  • Hardcover book with almost 300 pages
  • Covering most nerdy vintage studio classics from AKG, AMS, Dynacord, EMT, Lexicon, MXR, Quantec, Roland, ….
  • Phaser, delays, reverbs, pitch shifter, vocoder, exciter, multi fx – 68 devices presented in total

  • Great pictures (color and b&w) as well as insightful stories and statements from artists and manufacturers
  • Very fun to read or just to browse – inspirational in all regards
  • Available at Thomann for 69 smackers

next level saturation experience & still missing VoS plugins

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.

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.

sustaining trends in audio land, 2022 edition

Forecasts are difficult, especially when they concern the future – Mark Twain

In last years edition about sustaining trends in audio land I’ve covered pretty much everything from mobile and modular, DAW and DAW-less up to retro outboard and ITB production trends. From my point of view, all points made so far are still valid. However, I’ve neglected one or another topic which I’ll now just add here to that list.

The emergence of AI in audio production

What we can currently see already in the market is the ermergence of some clever mixing tools aiming to solve very specific mixing tasks, e.g. resonance smoothing and spectral balancing. Tools like that might be based on deep learning or other smart and sophisticated algorithms. There is no such common/strict “AI” definition and we will see an increasing use of the “AI” badge even only for the marketing claim to be superior.

Some other markets are ahead in this area, so it might be a good idea to just look into them. For example, AI applications in the digital photography domain are already ranging from smart assistance during taking a photo itself up to complete automated post processing. There is AI eye/face detection in-camera, skin retouching, sky replacement and even complete picture development. Available for all kinds of devices, assisted or fully automated and in all shades of quality and pricing.

Such technology not only shapes the production itself but a market and business as a whole. For example, traditional gate keepers might disappear because they are no longer necessary to create, edit and distribute things but also the market might get flooded with mediocre content. To some extend we can see this already in the audio domain and the emergence of AI within our production will just be an accelerator for all that.

The future of audio mastering

Audio Mastering demands shifted slightly over the recent years already. We’ve seen new requirements coming from streaming services, the album concept has become less relevant and there was (and still is) a strong demand for an increased loudness target. Also, the CD has been loosing relevance but Vinyl is back and has become a sustaining trend again, surprisingly. Currently Dolby Atmos gains some momentum, but the actual consumer market acceptance remains to be proven. I would not place my bet on that since this has way more implications (from a consumer point of view) than just introducing UHD as a new display standard.

Concerning the technical production, a complete ITB shift – as we’ve seen it in the mixing domain – has not been completed yet but the new digital possibilities like dynamic equalizing or full spectrum balancing are slowly adopted. All in all, audio mastering slowly evolves along the ever changing demands but remains surprisingly stable, sustaining as a business and this will probably continue for the next (few) years.

Social Media, your constant source of misinformation

How To Make Vocals Sound Analog? Using Clippers For Clean Transparent Loudness. Am I on drugs now? No, I’ve just entered the twisted realm of social media. The place where noobs advice you pro mixing tips and the reviews are paid. Everyone is an engineer here but its sooo entertaining. Only purpose: Attention. Currency: Clicks&Subs. Tiktok surpassed YT regarding reach. Content half-life measured in hours. That DISLIKE button is gone. THERE IS NO HOPE.

The (over-) saturated audio plugin market and the future of DSP

Over the years, a vast variety of vendors and products has been flooded the audio plugin market, offering literally hundreds of options to choose from. While this appears to be a good thing at first glance (increaed competition leads to lower retail prices) this has indeed a number of implications to look at. The issues we should be concerned the most about are the lack of innovation and the drop in quality. We will continue to see a lot of “me too” products as well as retro brands gilding their HW brands with yesterday SW tech.

Also, we can expect a trend of market consolidation which might appear in different shapes. Traditionally, this is about mergers and aquisitions but today its way more prominently about successfully establishing a leading business platform. And this is why HW DSP will be dead on the long run becuse those vendors just failed in creating competitive business platforms. Other players stepped in here already.

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

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.

 

 

 

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