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.

that unique plate reverb sound

Unlike digital reverberation, the plate reverb is one of the true analog attempts in recreating convincing reverberation build right into a studio device. It is basically an electro-mechanical device containing a plate of steel, transducers and a contact microphone to pickup the induced vibrations from that plate.

The sound is basically determined by the physical properties of the plate and its mechanical damping. Its not about reflecting waves from the plates surface but about the propagation of waves within the plate. While the plate itself has a fixed, regular shaped size and can be seen as a flat (two dimensional) room itself it actually does not produce early reflection patterns as we are used to from real rooms with solid walls. In fact there are no such reflections distinguishable by human hearing. On the other hand there appears to be a rather instant onset and the reverb build-up has a very high modal density already.

Also reverb diffusion appears to be quite unique within the plate. The wave propagation through metal performs different compared to air (e.g. speed/frequency wise) and also the plate itself – being a rather regular shape with a uniform surface and material – defines the sound. This typically results in a very uniform reverb tail although the higher frequencies tend to resonate a little bit more. Also due to the physics and the damping of the plate, we usually do not see hear very long decay times.

All in all, the fast and consistent reverb build up combined with its distinct tonality defines that specific plate reverb sound and explains why it is still so much beloved even after decades. The lack of early reflections can be easily compensated for just by adding some upfront delay lines to improve stereo localization if a mix demands it. The other way around, the plate reverb makes a perfect companion for all kinds of delay effects.

everything just fades into noise at the end

When I faced artificial reverberation algorithms to the very first time I just thought why not just dissolve the audio into noise over time to generate the reverb tail but it turned out to be not that easy, at least when just having the DSP knowledge and tools of that time. Today, digital reverb generation has come a long way and the research and toolsets available are quite impressive and diverse.

While the classic feedback delay network approaches got way more refined by improved diffusion generation, todays computational power increase can smooth things out further just by brute force as well. Still some HW vendors are going this route. Sampling impulse responses from real spaces also evolved over time and some DSP convolution drawbacks like latency management has been successfully addressed and can be handled more easily given todays CPUs.

Also, convolution is still the ticket whenever modeling a specific analog device (e.g. a plate or spring reverb) appears to be difficult, as long as the modeled part of the system is linear time invariant. To achieve even more accurate results there is still no way around physical modeling but this usually requires a very sophisticated modeling effort. As in practise everything appears to be a tradeoff its not that much unusual to just combine different approaches, e.g. a reverb onset gets sampled/convoluted but the reverb tail gets computed conventionally or – the other way around – early reflections are modeled but the tail just resolves into convoluted noise.

So, as we’ve learned now that everything just fades into noise at the end it comes to no surprise that the almost 15 years old epicVerb plugin becomes legacy now. However, it remains available to download for some (additional reverb) time. Go grab your copy as long as its not competely decayed, you’ll find it in the downloads legacy section here. There won’t be a MkII version but something new is already in the making and probably see the light of day in the not so far future. Stay tuned.

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.

the twisted world of guitar pedals II

Meanwhile I had the opportunity to put my hands on some Fairfield Circuitry effect pedal stuff mentioned earlier here and the “Meet Maude” analog BBD delay was right here on my desk for a deeper inspection. My actual experience was a rather mixed one.

Focusing on a rather dark and LoFi sound quality on the one hand plus a rather simplistic feature set concept wise on the other, they do not appear to be very flexible in practise and this at a rather steep price point. They appear to be very noisy featuring all kinds of artifacts even when integrated to the mixing desk via reamping. One may call this the feature itself but at the end it makes it a one-trick pony. If you need exactly that, here you have it but you get nothing beyond that. To me this trade off was too big and so I send it back.

However, I found their nifty low pass gate implementation (very prominently featured within their “Shallow Water”) that much unique and interesting that I replicated it as a low pass filter alternative in software and to have it available e.g. for filtering delay lines in my productions. The “Shallow Water” box made me almost pull the trigger but all in all I think this stuff seems to be a little bit over-hyped thanks to the interwebs. This pretty much sums it up for now, end of this affair.

Timeline & BigSky – The new dust collectors?

Going into the exact opposite direction might be a funny idea and so I grabbed some Strymon stuff which aims to be the jack of all trades at least regarding digital delay and reverb in a tiny stomp box aka desktop package. To be continued …

Further readings about BBD delays:

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

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

the twisted world of guitar pedals I

Quite recently I had a closer look into the vast amount of (guitar) effect pedals out there. Most are already DSP based which surprised me a little bit since I still ecpected more discrete analog designs after all. While looking for some neat real analog BBD delay I finally stumbled across Fairfield Circuitry’s “Meet Maude” which got me intrigued, having a rather rough look&feel at first sight but some very delicate implementation details under the hood.

Their delay modulation circuit has some randomness build in and also there is a compression circuit in the feedback loop – both designs I’ve also choosen for NastyDLA and which makes a big impact on the overall sound for granted. But the real highlight is the VCF in the delay feedback path which actually appears to be a low-pass gate – a quite unique design and soundwise also different but appealing in its very own regard.

They employed very similar concepts to their vibrato/chorus box “Shallow Water” featuring also random delay modulation and a low pass gate but this time a little bit more prominent on the face plate. On top, their JFET op-amp adds some serious grit to any kind of input signal. All in all, I did not expect such a bold but niche product to exist. If I ever will own such a thingy, there will be a much more detailed review here for sure.

TesslaSE mkII released

TesslaSE mkII – All the analog goodness in subtle doses

TesslaSE never meant to be a distortion box but rather focused on bringing all those subtle saturation and widening (side-) effects from the analog right into the digital domain. It sligthly colors the sound, polishes transients and creates depth and dimension in the stereo field. All the analog goodness in subtle doses. It’s a mixing effect intended to be used here and there where the mix demands it. It offers a low CPU profile and (almost) zero latency.

With it’s 2021 remake, TesslaSE mkII sticks to exactly that by just polishing whats already there. The internal gainstaging has been reworked so that everything appears gain compensated to the outside and is dead-easy to operate within a slick, modernized user interface. Also the transformer/tube cicuit modeling got some updates to appear more detailed and vibrant, while all non-linear algorithms got oversampled for additional aliasing supression.

Available for Windows VST in 32 and 64bit as freeware. Download your copy here.