lost & found

Stuff to share from the interwebs. All related to music making, sound design, audio production.

You know you’re getting old if you once were used to those:

  • The Museum of Endangered Sounds: Imagine a world where we never again hear the symphonic startup of a Windows 95 machine. Imagine generations of children unacquainted with the chattering of angels lodged deep within the recesses of an old cathode ray tube TV …
  • »Conserve the sound« is an online museum for vanishing and endangered sounds. The sound of a dial telephone, a walkman, a analog typewriter, a pay phone, a 56k modem, a nuclear power plant or even a cell phone keypad are partially already gone or are about to disappear from our daily life …
  • And also this:

Modern scoring: ‘Tenet’: Ludwig Göransson Put Chris Nolan’s Breath in His Score — and Rethought Composing Altogether

About the future of audio mastering: How AI is solving one of music’s most expensive problems

FerricTDS is about glueing things together and not about distortion (I already told you):

Weird sound devices:

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 (10) – Vladislav Goncharov

Vlad, what was your very first DSP plugin development, how did it once started and what was your motivation behind?

My first plugin was simple a audio clipper. But I decided to not release it. So my first public released plugin was Molot compressor. I was a professional software engineer but with zero DSP knowledge (my education was about databases, computer networks and stuff like that). I played a guitar as a hobby, recorded demos at home and one day I found that such thing as audio plugins exist. I was amazed by their amount and also by the fact that there are free plugins too. And I realised that one day I can build something like this myself. I just had to open a DSP book, read a chapter or two and it was enough to start. So my main motivation was curiosity, actually.

Was that Molot compressor concept inspired by some existing devices or a rather plain DSP text book approach?

That days there was a rumour that it’s impossible to make good sounding digital compressor because of aliasing and stuff. I tried to make digital implementation as fluid as possible, without hard yes/no logic believing this is how perfect digital compressor should sound. And the way I implemented the algorithm made the compressor to sound unlike anything I heard before. I didn’t had any existing devices in my head to match and I didn’t watch textbook implementations too. The sound was just how I made it. I did 8 versions of the algorithm trying to make it as usable as possible from user point perspective (for example “harder” knee should sound “harder”, I removed dual-band implementation because it was hard to operate) and the last version of the project was named “comp8”.

Did you maintained that specific sound within Molot when you relaunched it under the TDR joint venture later on? And while we are at it: When and how did that cooperation with Fabien started?

TDR Molot development was started with the same core sound implementation as original Molot had. But next I tried to rework every aspect of the DSP to make it sound better but keep the original feel at the same time. It was very hard but I think I succeeded. I’m very proud of how I integrated feedback mode into TDR Molot for example. About Fabien: He wrote me to discuss faults in my implementation he thought I had (I’m not sure it was Molot or Limiter 6), we also discussed TDR Feedback Compressor he released that days, we argued against each other but what’s strange the next day we both changed our minds and agreed with our opposite opinions. It was like “You were right yesterday. No, I think you were right”. Next there was “KVR Developer Challenge” and Fabien suggested to collaborate and create a product for this competition. That was 2012.

And the Feedback Compressor was the basis for Kotelnikov later on, right?

No, Kotelnikov is 100% different from Feedback Compressor. Fabien tried to make the sound of feedback compressor more controllable and found that the best way to achieve this is just to change the topology to feedforward one. It’s better to say, Feedback Compressor led to Kotelnikov. Also the interesting fact, early version of Kotelnikov had also additional feedback mode but I asked Fabien to remove it because it was the most boring compressor sound I ever heard. I mean if you add more control into feedback circuit, it just ruins the sound.

Must have been a challenge to obtain such a smooth sound from a feed-forward topology. In general, what do you think makes a dynamic processor stand out these days especially but not limited to mastering?

I think, it’s an intelligent control over reactions. For example Kotelnikov has some hidden mechanisms working under the hood, users don’t have access to them but they help to achieve good sound. I don’t think it’s good idea to expose all internal parameters to the user. There must be hidden helpers just doing their job.

I so much agree on that! Do you see any new and specific demand concerning limiting and maximizing purposes? I’m just wondering how the loudness race will continue and if we ever going to see a retro trend towards a more relaxed sound again …

I think even in perfect loudness normalized world most of the music is still consumed in noisy environments. The processing allowing the quietest details to be heard and cut through background noise, to retain the feel of punch and density even at low volumes is in demand these days. Loudness maximizers can do all this stuff but in this context they act like old broadcast processors. In my opinion, the loudness war will continue but it’s not for overall mix loudness anymore but how loud and clear each tiny detail of the mix should be.

Can we have a brief glimpse on what you are currently focused on, DSP development wise?

You may take a look at Tokyo Dawn Labs Facebook posts. We shared a couple of screenshots some time ago. That’s our main project to be released someday. But also we work on a couple of dynamic processors in parallel. We set high mark on the quality of our products so we have to keep it that high and that’s why the development is so slow. We develop for months and months until the product is good enough to be released. That’s why we usually don’t have estimation dates of release.

Related Links

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)