artificial reverberation: from mono to true stereo

“True stereo” is a term used in audio processing to describe a stereo recording, processing or playback technique that accurately represents the spatial location of sound sources in the stereo field. In true stereo, the left and right channels of a stereo recording contain distinct and separate audio information that accurately reflects the spatial location of sound sources in the recording environment.

This is in contrast to fake/pseudo stereo, where the stereo image is created through artificial means, such as by applying phase shifting techniques to create the impression of stereo. True stereo is generally considered to be superior to fake stereo, as it provides a more natural and immersive listening experience, allowing the listener to better locate and identify sound sources within the stereo field. In the domain of acoustic reverberation, this is essential for the perception of envelopment.

Artificial reverberation has come a long way since its early beginnings. The first mechanical devices for generating artificial reverberation, such as spring or plate reverberation, were initially only available as mono devices. Even when two-channel variants emerged, they usually did summing to mono internally or did processing in separate signal paths, known as dual mono processing. Typically, in a plate reverb, a two-channel output signal was achieved simply by mounting two transducers on the very same reverb plate.

The first digital implementations of artificial reverberation did not differ much from the mechanical ones regarding this principle. Quite common was summing the inputs to mono and the independent tap of two signals from a single reverb tank to obtain a two-channel output. Then, explicit early reflection models were added, which were typically processed for left and right separately and merged into the outputs later to preserve a basic representation of spatial information. Sometimes, also the first reflections were just taken from a (summed) mono signal. The Ursa Major 8×32 from 1981 is a good example for this design pattern. Later, the designs became more sophisticated, and even today it is common to distinguish between early and late reverberation in order to create a convincing impression of immersion.

However, ensuring proper sound localisation through early reflection models is a delicate matter. First and foremost, a real room does not have a single reflection pattern, but a vast variety of ones that depend on the actual location of the sound source and the listening position in that room. A true-to-life representation of this would, therefore, have to be represented by a whole set of individual reflection patterns per sound source and listening position in the virtual room. As far as I know, the VSL MIR solution is the only one that currently takes advantage of this, and with an enormous technical effort.

Another problem is that first reflections can also be detrimental to the sound experience. Depending on their frequency and delay in relation to the direct signal, the direct signal can be masked and affected in terms of phase coherence so that the overall sound becomes muddy and lacks clarity. This is one of the reasons why a real plate reverb is loved so much for its clarity and immediacy: it simply has no initial reflections in this range. As a side note, in the epicPLATE implementation, this behaviour is accurately modeled by utilizing a reverberation technique that completely avoids reflections (delays).

Last but not least, in a real room there is no clear separation between the first reflections and the late reverberation. It is all part of the same reverberation that gradually develops over time, starting with just an auditory event. This also means that there is no clear distinction between events that can be located in space and those that can no longer be identified – this also continuously evolves over time.

A good example of how to realise digital reverb without this kind of separation between early and late reverberation and at the same time in “true stereo” was impressively demonstrated by the Quantec QRS back in the early 80s already. Its ability to accurately reproduce stereo was one of the reasons why it became an all-time favourite not only in the music production scene, but also in post-production and broadcasting.

Artificial reverberation is full of subtleties and details and one might wonder why we can perceive them at all. In the end, it comes down to the fact that in the course of evolution there was a need for such fine-tuning of our sensory system. It was a matter of survival and important for all animal species to immediately recognise at all times: What is it and where is it? The entire sensory system is designed for this and even combines the different sensory channels to always answer these two questions. Fun Fact: This is exactly why some visual cues can have a significant impact on what is heard and why blind tests (in both meanings) are so important for assessing certain audio qualities. See also the “McGurk Effect” if you are interested.

Have fun listening!

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 RMS (or vector) summing, 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.

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.

Image Courtesy of Music Tribe IP Ltd.

 

 

 

The Korg SDD-3000 – perfect for LoFi?

By accident, I recently stumbled upon the UAD Korg SDD-3000 digital delay version. When I noticed that they modelled also its amplifiers as well as the 13bit converters they immediately got my attention. Having also high- and low-pass filters on board, this could easily double as a great lofi device – so lets have a closer look.

As in the original hardware, the device offers several gain stage adjustments for both input and ouptut, intended to match different instrument or line level signals. These amplifiers are always in, no matter if the BYPASS switch is activated or not. Interestingly, UA also integrated this in its “Unison” interface feature as an preamp option.

Depending on how hard the input gain is driven, quite heavy distortion and saturation effects can occur. As soon as the Bypass is deactivated, the effect signal path containing the 13bit converted and HP/LP filtered signal can be dialed in with the LEVEL BALANCE. If this balance is now set to EFFECT only or just the WET SOLO option has been turned on (plus avoiding any amounts of feedback in this case) the device now offers a pretty much nicely degraded signal path for any sort of creative effects. Depending on the actual settings one can dial in now some really creamy or even gritty effects. Be aware, that this signal path contains an additional delay according to the DELAY TIME setting, of course.

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The analysis charts are showing – from left to right – the basic frequency response (in bypass mode), some example harmonic distortions when hitting the input gain quite hard and the filtered effect signal path frequency response according to the UI settings above. The slight frequency bump on the right side of the charts might be caused by the plugin oversampling filters – the original hardware does not show this and its spectrum ends somewhere around 17kHz.

As in the original hardware, all settings are just within limited ranges and so it is not that flexible in general. However, soundwise its pretty much awesome. Oh and by the way, it also doubles as a simple but impressive delay 😉

SlickEQ – some more release info

Just a couple of days ago we introduced the upcoming release of SlickEQ and lots of questions raised already. So, here is what Fabien already committed about it in a public forum:

• Win/Mac, AU/VST2/VST3 (+AAX planned and in process), x32/x64
• No linux builds planned, sorry.
• The name is “TDR VOS Slick EQ” and it will be available for free.
• Release is a matter of days. Maybe a week or two.

As of today I just want to add: With the introduction of TDR VOS SlickEQ, quite a number of amazing and previously unheard DSP algorithms will see the light of day – including (but not limited to) several Stateful Saturation algorithms running within an audio signal path entirely upsampled to a constant high sample rate for maximum precision.

Expect smoothness, best-in-class.

Related links:

• thread at kvraudio.com
• what I’m currently working on – Vol. 12
• vladg/sound
• sharing the same vision …

processing with High Dynamic Range (3)

This article explores how some different HDR imaging alike techniques can be adopted right into the audio domain.

The early adopters – game developers

In the lately cross-linked article “Finding Your Way With High Dynamic Range Audio In Wwise” some good overview was given on how the HDR concept was already adopted by some game developers over the recent years. Mixing in-game audio has its very own challenge which is about mixing different arbitrary occurring audio events in real-time when the game is actually played. Opposed to that and when we do mix off-line (as in a typical song production) we do have a static output format and don’t have such issues of course.

So it comes as no surprise, that the game developer approach turned out to be a rather automatic/adaptive in-game mixing system which is capable of gating quieter sources depending on the overall volume of the entire audio plus performing some overall compression and limiting. The “off-line mixing audio engineer” can always do better and if a mix is really too difficult, even the arrangement can be fixed by hand during the mixing stage.

There is some further shortcoming and from my point of view that is the too simplistic and reduced translation from “image brightness” into “audio loudness” which might work to some extend but since the audio loudness race has been emerged we already have a clear proof how utterly bad that can sound at the end. At least, there are way more details and effects to be taken into account to perform better concerning dynamic range perception.

interview series (8) – Sascha Eversmeier

Sascha, are you a musician yourself or do you have some other sort of musical background? And how did you once got started developing your very own audio DSP effects?

I started learning to play bass guitar in early 1988, when I was 16. Bass is still my main instrument, although I also play a tiny bit of 6-string, but I’d say I suck at that.

The people I played with in a band in my youth where mostly close friends I grew up with, and most of us kept on making music together when we finished school a couple of years later. I still consider that period (mid-nineties) as sort of my personal heyday, musical-wise. It’s when you think you’re doing brilliant things but the world doesn’t take notice. Anyway. Although we all started out doing Metal, we eventually did Alternative and a bit of Brit-influenced Wave Rock back then.

That was also the time when more and more affordable electronic gear came up, so apart from doing the usual rock-band lineup, we also experimented with samplers, DATs, click tracks and PCs as recording devices. While that in fact made the ‘band’ context more complex – imagine loading in a dozen disks into the E-MU on every start of the rehearsal until we equipped it with an MO drive – we soon found ourselves moving away from writing songs through jamming and more to actually “assembling” them by using a mouse pointer. In hindsight, that was really challenging. Today, the DAW world and the whole process of creating music is so much simpler and intuitive, I think.

My first “DAW” was a PC running at 233Mhz, and we used PowerTracks Pro and Micro Logic – a stripped-down version of Logic -, although the latter never clicked with me. In 1996 or 97 – can’t remember – I purchased Cubase and must have ordered right within a grace period, as I soon got a letter from Steinberg saying they now finished the long-awaited VST version and I could have it for free, if I want. WTF? I had no idea what they were talking about. But Virtual Studio Technology, that sounded like I was given the opportunity to upgrade myself to being “professional”. How flattering, you clever marketing guys. Yes, gimme the damn thing, hehe.

When VST arrived, I was blown away. I had a TSR-8 reel machine, a DA-88 and a large Allen&Heath desk within reach and was used to run the computer as a midi sequencer mainly. And now, I could do it all inside that thing. Unbelievable. Well, the biggest challenge then was finding an affordable audio card, and I bought myself one that only had S/PDif in & outputs and was developed by a German electronics magazine and sold in small amounts through a big retail store in Cologne, exclusively. 500 Deutschmarks for 16 bits on an ISA card. Wow.

The first plugin I bought was Waves Audio Track, sort of a channel strip, which was a cross-promotion offer from Steinberg back then, 1997, I guess. I can still recall its serial number by heart.

Soon, the plugin scene lifted off, and I collected everything I could, like the early mda stuff, NorthPole and other classics. As our regular band came to nothing, we gathered our stuff and ran sort of a small project studio where we recorded other bands and musicians and started using the PC as the main recording device. I upgraded the audio hardware to an Echo Darla card, but one of my mates soon brought in a Layla rack unit so that we had plenty of physical ins and outs.

You really couldn’t foresee where the audio industry would go, at least I couldn’t. I went fine with this “hybrid” setup for quite a long time, and did lots of recording and editing back then, but wasn’t even thinking of programming audio software myself at all. I had done a few semesters of EE studies, but without really committing myself much.

Then the internet came along. In 1998, I made a cut and started taking classes in Informatics. Finished in 2000, I moved far away, from West Germany, to Berlin and had my first “real” job in one of those “new economy” companies, doing web-based programming and SQL. That filled the fridge and was fun to do somehow, but wasn’t really challenging. As my classes included C, C++ and also Assembler, and I still got a copy of Microsoft’s Visual Studio, I signed up to the VST SDK one day. At first, I might have done pretty much the same thing as everybody: compile the “gain” and “delay” plugin examples and learn how it all fits together. VST was still at version 1 at that time, so there were no instruments yet, but I wasn’t interested much in those anyway, or at least I could imagine writing myself a synthesizer. What I was more interested in was how to manipulate the audio so that it could sound like a compressor or a tube device. I was really keen on dynamics processing at that time, perhaps because I always had too few of those units. I had plenty available when I was working part-time as a live-sound engineer, but back in my home studio, a cheap Alesis, dbx or Behringer was all I could afford. So why not try to program one? I basically knew how to read schematics, I knew how to solder, and I thought I knew how things should sound like, so I just started out hacking things together. Probably in the most ignorant and naive way, from today’s perspective. I had no real clue, and no serious tool set, apart from an old student’s copy of Maple and my beloved Corel 7. But there were helpful people on the internet and a growing community of people devoted to audio software, and that was perhaps the most important factor. You just weren’t alone.

announcing the “Slick” audio plug-in series

About history

In recent history, I’ve constantly extended and improved my Stateful Saturation approach and within ThrillseekerVBL I’ve managed to introduce authentic analog style sounding distortion right into VST land, which is what I’ve always had in my mind and dreamed of. And there’s so much and overwhelming feedback on that – thank you sooo much!

Best of both worlds

Since quite a while, I’ve dreamed about a brand new series of plug-ins which will combine the strength of both worlds: analog modeling on the one side but pure digital techniques on the other – incorporating techniques such as look-ahead, FIR filtering or even stuff that comes from the digital image processing domain, such as HDR (High Dynamic Range) processing.

First encounter: SlickHDR

High Dynamic Range (HDR) processing is something pretty much new in the audio domain. While there are lots of theories and implementations available about HDR imaging, this is quite new and sparingly adopted in the audio domain. SlickHDR is going to be a very first approach in applying high dynamic range processing to audio within a VST compatible plug-in.

interview series (7) – Dave Gamble

Dave, can you tell us a little about how you got into music, and your professional career as an audio effects developer so far?

Started writing trackers as a child, then wrote some code to allow me to DJ with trackers. By 14 I was writing commercial software. Had some great teachers and lecturers who helped me a lot. Did my final-year project with Focusrite. Won the project prize. Spent 4.5 years at Focusrite (I was employee 12 or 13) to add DSP to the company, during which time we acquired Novation, and grew quite a lot. We made a lot of money from audio interfaces, so that kinda took over, and I wanted to get back to the DSP (at Focusrite I did Forte suite, helped with Liquid Channel/Mix, Saffire suite, plus other non DSP projects). Left for Sonalksis, built all their shipping products (except CQ1 and DQ1), although I’d built tbk1 years before and they’d been selling it. Was fun but chaotic. Left to go freelance so I could start my own outfit, during which time I worked with Neyrinck, TAC System, Focusrite, Novation, Studio Devil, FXpansion, Brainworx/Plugin Alliance, etc. Then started dmgaudio. And here we are now.

interview series (6) – Christopher Dion

Chris, you are the man behind the Canada-based Quantum-Music studio. What was your journey towards this venture?

My father (Alain Dion) was an internationally renown live sound engineer and technical producer (Nat King Cole, Sting, Celine Dion, Cirque du Soleil, and many locally-famous artists). Therefore, I grew up in an environment where high fidelity audio was the standard. My father hated everything that sounded less than perfect. Unconsciously, he trained my ears. I owe him a lot for that. Nowadays, every time we see each other, we spend much of our time talking about which compressors, consoles and techniques.