What loudspeakers and audio transformers do have in common

Or: WTF is “group delay”?

Imagine a group of people visiting an exhibition having a guided tour. One might expect that the group reaches the exhibitions exit as a whole but in reality there might be a part of that group just lagging behind a little bit actually (e.g. just taking their time).

Speaking in terms of frequency response within audio systems now, this sort of delay is refered to as “group delay”, measured in seconds. And if parts of the frequency range do not reach a listeners ear within the very same time this group delay is being refered to as not being constant anymore.

A flat frequency response does not tell anything about this phenomena and group delay must always be measured separately. Just for reference, delays above 1-4ms (depending on the actual frequency) can actually be perceived by human hearing.

This always turned out to be a real issue in loudspeaker design in general because certain audio events can not perceived as a single event in time anymore but are spread across a certain window of time. The root cause for this anomaly typically lies in electrical components like frequency splitters, amplifiers or filter circuits in general but also physical loudspeaker construction patterns like bass reflex ports or transmission line designs.

Especially the latter ones actually do change the group delay for the lower frequency department very prominently which can be seen as a design flaw but on the other hand lots of hifi enthusiast actually do like this low end behaviour which is able to deliver a very round and full bass experience even within a quite small speaker design. In such cases, one can measure more than 20ms group delay within the frequency content below 100Hz and I’ve seen plots from real designs featuring 70ms at 40Hz which is huge.

Such speaker designs should be avoided in mixing or mastering situation where precision and accuracy is required. It’s also one of the reasons why we can still find single driver speaker designs as primary or additional monitoring options in the studios around the world. They have a constant group delay by design and do not mess around with some frequency parts while just leaving some others intact.

As mentioned before, also several analog circuit designs are able to distort the constant group delay and we can see very typical low end group delay shifts within audio transformer coupled circuit designs. Interestingly, even mastering engineers are utilizing such devices – whether to be found in a compressor, EQ or tape machine – in their analog mastering chain.

Lets talk about mixing levels (again)

Some years ago we had lots of discussions about proper mixing levels in the digital domain – with mixed (sic!) results, IIRC. Meanwhile, more and more influencers are claiming that targeting -18dBFS with a VU meter readout is the “digital audio sweet spot” and the way forward in terms of plugin gain staging. In practise that would imply mixing digital peak levels at around 0dBFS again but maybe I’ve missed something during my absence in recent years. So, to what mixing levels are you up to in your DAW today?

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!

A more realistic look at the Pultec style equalizer designs

One of the few historic audio devices with almost mystical status is the Pultec EQP-1A EQ and a lot of replicas has been made available across the decades. Whether being replicated in soft- or hardware, what can we expect from a more realistic point of view? Lets have a closer look.

Some fancy curves from the original EQP-1A manual
  • In the top most frequency range a shelving filter with 3 pre selected frequencies is offered but just for attenuation. Much more common and usable for todays mixing and mastering duties would be an air band shelving boost option here.
  • Also in the HF department there is just one single peak filter but this time just for boosting. It offers 7 pre selected frequencies between 3 and 16kHz and only here the bandwidth can be adjusted. However, the actual curves could have been steeper for todays mixing duties.
  • There is no option in the mid or low-mid range at all and also no high pass option. Instead, there is a shelving filter for the low-end which allows for boost and/or attenuation around four pre selected frequencies between 20 and 100 Hz.

All in all, this appears to be a rather quirky EQ concept with quite some limitations. On top of that, the low frequency behaviour of the boost and cut filters is rather unpredictable if both filters are engaged simultaneously which is exactly the reason why the original manual basically states “Do not attempt to do this!”.

Nowadays being refered to as the “Pultec Bass Trick” the idea is that you not only boost in some low end area but also create some sort of frequency dip sligthly above to avoid too much of a boost and muddiness in total. In practise, this appears to be rather unpredictable. Dial in a boost at 3 and an attenuation at 5, just as an example: Does this already feature a frequency dip? And if so at which frequency exactly? One has no idea and it even gets worse.

Due to aged electronics or component variety one has to expect that the actual curve behaviour might differ and also to see each vendors replica implementation to be different from another. In practise this indeed holds true and we can see the actual bass frequency dip at a much higher frequency within one model compared to another, just as an example.

… the more I boost the EQ the more it makes me smile …

A reviewers statement misguided by simple loudness increase?

Fun fact: Like the original device, all current (hardware) replica models do not have an output gain control. Also they increase the overall signal level just by getting inserted into the signal path.

So, where is the beef? Its definately not in the curves or the overall concept for sure. Maybe I’ll take some time for a follow-up article and a closer look into the buffer amplifier design to see if all the hype is justified.

Further Links

Not really demystifying but fun to read:

In the VoS plugin line you can find some Pultec style low end performance within NastyVCS: https://varietyofsound.wordpress.com/2010/05/07/nastyvcs-released-today/

Also interesting to read and hear: https://www.sweetwater.com/insync/pultec-shootout-with-sound-samples/

sustaining trends in audio land, 2021 edition

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

The mobile audio evolution

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

The retro cult continues

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

Modular madness

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

Look mom no computer

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

Loudness war, quo vadis?

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

ITB production finally took over

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

Game of DAWs

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

The pandemic impact

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

Stay healthy!

 

Rebuilding my Studio

Everything is finished, it just has to be done
– Andreas Pflüger

So, since some 5 years or so I did not had the time and room to make any music at all but at the end I’ve missed it so much that I finally decided to restart all over again and finally rebuild my studio. During my very last attempts in creating music I got stuck somehow inbetween all those endless digital options and just looping stuff in Ableton on my laptop but never managed to get things finished anymore – some of you might know this kind of desease? Anyhow, I decided to setup a small studio production environment like I once had before those laptop times and where I was a little bit more productive, if memory serves me right.

Not a huge setup at all but a small desktop centric approach at home with some few but well selected outboard gear which not only inspires but also invites to perform, record and collaborate whole tracks in a fun way. Luckily, some outboard gear was carefully archived here in my basement but others I do not have anymore, e.g. a real mixing desk. But hey, there is so much cool stuff out there today to consider and to choose from!

Such new setup raises a lot of questions of course concerning workflow and where to better rely on ITB or OTB plus the usual digital versus analog considerations. This time it was clear for me to have best of both worlds right from the start. Gaining advantage from DAW based sequencing, mixing and mastering – speaking in terms of precision, affordability and total recall – but also the fun and hands-on experience just real outboard gear can give you and I’m not talking about cheap plastic controllers here (don’t get me started on that one).

For the DAW itself I just reactivated my rather aged PC based workstation which complained to need some hundred or so updates but then afterwards actually performed quite smoothly again. The whole installation is much more lean now and I did not included all that sh*tload of huge sample libraries just as an example. For the time being, I can live with its constraints and can focus on other and more important stuff now. Also, there won’t be a mixing desk anymore and I just added some more converters to my old but trusty RME card via ADAT. Nice to see the old standards still working flawless. Now the fun part begins: connecting all the stuff.

To be continued.

Kotelnikov GE – mastering

Here is my go-to mastering preset for Kotelnikov GE. Just change the threshold and you are there.

<TDRKotelnikovGE thresholdParam=”-24.0″ peakCrestParam=”-3.0″ softKneeParam=”6.0″ ratioParam=”3.0″ attackParam=”6.0″ releasePeakParam=”20″ releaseRMSParam=”300″ makeUpParam=”0.0″ dryMixParam=”off” outGainParam=”0.0″ keyHPFrequencyParam=”60″ keyHPSlopeParam=”6.0″ keyStereoDiffParam=”80″ keyStereoBalanceParam=”Center” fdrVisibleParam=”On” fdrActiveParam=”On” fdrTypeParam=”Shelf A” fdrFrequencyParam=”50″ fdrAmountParam=”80″ yingParam=”On” yangParam=”Off” deltaParam=”Off” bypassParam=”Off” equalLoudParam=”Off” qualityParam=”Insane” modeParam=”Stereo” grDispScaleParam=”4″ grDispModeParam=”Gain Reduction”/>

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. [Read more…]

processing with High Dynamic Range (2)

This comprehensive and in-depth article about HDR imaging was written by Sven Bontinck, a professional photographer and a hobby-musician.

A matter of perception.

To be able to use HDR in imaging, we must first understand what dynamic range actually means. Sometimes I notice people mistake contrast in pictures with the dynamic range. Those two concepts have some sort of relationship, but are not the same. Let me start by explaining in short how humans receive information with our eyes and ears. This is important because it influences the way we perceive what we see and hear and how we interpret that information.

We all know about the retina in our eyes where we find the light-sensitive sensors, the rods and cones. The cones provide us daytime vision and the perception of colours. The rods allow us to see low-light levels and provide us black-and-white vision. However there is a third kind of photoreceptors, the so-called photosensitive ganglion cells. These cells give our brain information about length-of-day versus length-of-night duration, but also play an important role in the pupillary control. Every sensor need a minimum amount of incitement to be able to react. At the same time all kind of sensors have a maximum amount that they may be exposed to. Above that limit, certain protection mechanisms start interacting to prevent damage occurring to the sensors. [Read more…]

processing with High Dynamic Range (1)

Back in time when I was at university, my very first DSP lectures were actually not about audio but image processing. Due to my interest in photography I followed this amazing and ever evolving domain over time. Later on, High Dynamic Range (HDR) image processing emerged and beside its high impact on digital photography, I immediately started to ask myself how such techniques could be translated into the audio domain. And to be honest, for quite some time I haven’t got a clue.

MM

This image shows a typical problem digital photography still suffers from: The highlights are completely washed out and so the lowlights are turning into black abruptly w/o containing further nuances  – the dynamic range performance is pretty much poor and this is actually not what the human eye would perceive since it features both: a higher dynamic range per se but also a better adoption to different (and maybe difficult) lighting conditions.

On top, we have to expect severe dynamic range limitations in the output entities whether that’s a cheap digital print, a crappy TFT display or the limited JPG file format, just as an example. Analog film and prints does have such problems in principle also but not to that much extend since they typically offer more dynamic resolution and the saturation behavior is rather soft unlike the digital hard clipping. And this is where HDR image processing chimes in.

It typically distinguishes between single- and multi-image processing. Within multi-image processing, a series of Low Dynamic Range (LDR) images are taken in different exposures and combined into one single new image which contains an extended dynamic range (thanks to some clever processing). Afterwards, this version is rendered back into an LDR image by utilizing special  “tone mapping” operators which are performing a sort of dynamic range compression to obtain a better dynamic range impression but now in a LDR file.

Within single-image processing, there must be one single HDR image already available and then just tone mapping is applied. As an example, the picture below takes advantage of single-image processing from a RAW file which typically does have much higher bit-depth (12 or even 14 bit as of todays sensor tech) opposed to JPG (8 bit). As a result a lot of dynamic information can be preserved even if the output file still is just a JPG. As an added sugar, such a processed image also translates way better over a wide variety of different output devices, displays and viewing light conditions.

MM-HDR