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.

The renaissance of the Baxandall EQs

Already in 1950, Peter Baxandall designed an analog tone correction circuit which found its way into some million consumer audio devices later on. Today, it is simply referred to as a Baxandall EQ.

What the f*ck is a Baxandall EQ?

Beside its appearance in numerous guitar amplifiers and effects, it made a very prominent reincarnation in the pro audio gear world in 2010 with the Dangerous Music Bax EQ. The concept shines with its very broad curves and gentle slopes which are all about transparancy and so it came to no surprise that this made it into lots of mastering rigs right away.

And it also had a reason that already in 2011 I did an authentic 1:1 emulation of the very same curves within the Baxter EQ plugin but just adding a dual channel M/S layout to better fit the mastering duties. For maximum accuracy and transparancy it already featured oversampling and double-precision filter calculations to that time and it is still one of my personal all time favourite EQs.

BaxterEQ

During the last 10 years quite a number of devices emerged each showing its very own interpretation of the Baxandall EQ whether thats in hard or software and this was highly anticipated especially in the mastering domain.

A highly deserved revival aka renaissance.

When comparing units be aware that the frequency labeling is not standardized and different frequencies might be declared while giving you same/similar curves. More plots and infos can be found here (german language).

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/

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”/>

mastering a track using only FREE Plugins

Featuring BaxterEQ and FerricTDS.

(via)

compressor aficionados (4) – Bob Olhsson

Bob, you are a professional recording and mastering engineer for over thirty years and already legend. What is more important: The ability to hear or the ability to use the right device according to the context and to apply the right adjustment?

You need to BOTH be able to hear AND find the right device and settings! Probably the most important thing to understand is that just raising the volume a tenth of a dB. will always sound better. All comparisons must be checked with the average level compensated. Otherwise, it’s easy to wind up with something that’s louder but far worse sounding.

Another challenge is mastering for what the recording needs and not what your monitors want. Our goal is for it to sound great everywhere. The best way I’ve found to tackle this problem is sticking to broad strokes unless I’m removing distractions. If a half dB. sounds different but not better, I’ll generally leave it alone. [Read more…]

working ITB at higher sampling rates

Recently, I’ve moved from 44.1kHz up to 96kHz sampling rate for my current production. I would have loved to do this step earlier but it wasn’t possible with the older DAW generation in my case. With the newer stuff I was easily able to run a 44.1kHz based production with tons of headroom (resource wise – talking about CPU plus memory and disk space) and so I switched to 96kHz SR and still there is some room left.

I know there is a lot of confusion and misinformation floating around about this topic and so this small article is about to give some theoretical insights from a developer perspective as well as some hands-on tips for all those who are considering at what SR actually to work at. The title already suggests working ITB (In The Box) and I’ll exclude SR topics related to recording, AD/DA converters or other external digital devices. [Read more…]

a Density mkIII review

Christopher A. Dorval Dion kindly gave me permission to re-blog his review he made for Quantum-Music.ca:

Variety of Sound – Density MKIII (Review)

[Read more…]