how I listen to audio today

Developing audio effect plugins involves quite a lot of testing. While this appears to be an easy task as long as its all about measurable criteria, it gets way more tricky beyond that. Then there is no way around (extensive) listening tests which must be structured and follow some systematic approach to avoid ending up in fluffy “wine tasting” categories.

I’ve spend quite some time with such listening tests over the years and some of the insights and principles are distilled in this brief article. They are not only useful for checking mix qualities or judging device capabilities in general but also give someΒ  essential hints about developing our hearing.

No matter what specific audio assessment task one is up to, its always about judging the dynamic response of the audio (dynamics) vs its distribution across the frequency spectrum in particular (tonality). Both dimensions can be tested best by utilizing transient rich program material like mixes containing several acoustic instruments – e.g. guitars, percussion and so on – but which has sustaining elements and room information as well.

Drums are also a good starting point but they do not offer enough variety to cover both aspects we are talking about and to spot modulation artifacts (IMD) easily, just as an example. A rough but decent mix should do the job. On my very own, I do prefer raw mixes which are not yet processed that much to minimize the influence of flaws already burned into the audio content but more on that later.

Having such content in place allows to focus the hearing and to hear along a) the instrument transients – instrument by instrument – and b) the changes and impact within particular frequency ranges. Lets have a look into both aspects in more detail.

a) The transient information is crucial for our hearing because it is used not only to identify intruments but also to perform stereo localization. They basically impact how we can separate between different sources and how they are positioned in the stereo field. So lets say if something “lacks definition” it might be just caused by not having enough transient information available and not necessarily about flaws in equalizing. Transients tend to mask other audio events for a very short period of time and when a transient decays and the signal sustains, it unveils its pitch information to our hearing.

b) For the sustaining signal phases it is more relevant to focus on frequency ranges since our hearing is organized in bands of the entire spectrum and is not able to distinguish different affairs within the very same band. For most comparision tasks its already sufficient to consciously distinguish between the low, low-mid, high-mid and high frequency ranges and only drilling down further if necessary, e.g. to identify specific resonances. Assigning specific attributes to according ranges is the key to improve our conscious hearing abilities. As an example, one might spot something “boxy sounding” just reflecting in the mid frequency range at first sight. But focusing on the very low frequency range might also expose effects contributing to the overall impression of “boxyness”. This reveals further and previously unseen strategies to properly manage such kinds of effects.

Overall, I can not recommend highly enough to educate the hearing in both dimensions to enable a more detailed listening experience and to get more confident in assessing certain audio qualities. Most kinds of compression/distortion/saturation effects are presenting a good learning challenge since they can impact both audio dimensions very deeply. On the other hand, using already mixed material to assess the qualities of e.g. a new audio device turns out to be a very delicate matter.

Lets say an additional HF boost applied now sounds unpleasant and harsh: Is this the flaw of the added effect or was it already there but now just pulled out of that mix? During all the listening tests I’ve did so far, a lot of tainted mixes unveiled such flaws not visible at first sight. In case of the given example you might find root causes like too much mid frequency distortion (coming from compression IMD or saturation artifacts) mirroring in the HF or just inferior de-essing attempts. The most recent trend to grind each and every frequency resonance is also prone to unwanted side-effects but that’s another story.

Further psychoacoustic related hearing effects needs to be taken into account when we perform A/B testing. While comparing content at equal loudness is a well known subject (nonetheless ignored by lots of reviewers out there) it is also crucial to switch forth and back sources instantaneously and not with a break. This is due to the fact that our hearing system is not able to memorize a full audio profile much longer than a second. Then there is the “confirmation bias” effect which basically is all about that we always tend to be biased concerning the test result: Just having that button pressed or knowing the brand name has already to be seen as an influence in this regard. The only solution for this is utilizing blind testing.

Most of the time I listen through nearfield speakers and rarely by cans. I’m sticking to my speakers since more than 15 years now and it was very important for me to get used to them over time. Before that I’ve “upgraded” speakers several times unnecessarily. Having said that, using a coaxial speaker design is key for nearfield listening environments. After ditching digital room correction here in my studio the signal path is now fully analog right after the converter. The converter itself is high-end but today I think proper room acoustics right from the start would have been a better investment.

FerricTDS mkII released

FerricTDS mkII – the updated award winning Tape Dynamics Simulator

New in version 2:

  • Introducing operating level calibration for better gainstaging and output volume compensated processing
  • Metering ballistics revised and aligned accordingly
  • Updated tape compression algorithms increasing punch, adding 2nd order harmonic processing, less IMD
  • Updated limiter algorithm featuring ADC style converter clipping
  • All non-linearities are running at higher sampling frequencies internally
  • Adding a sophisticated analog signal path emulation

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

freeware tip: TDR Feedback Compressor II

If you did not tried this one out yet but are looking for an absolutely clean and transparent compressor then do check this one out!

The TDR Feedback Compressor II is a major design update of its critically acclaimed predecessor. The compressor is dedicated to the highest fidelity stereo program (2-buss) compression, but shines equally in classic mixing tasks. – Fabien from TDR

It features detailed control of compression behaviour, extremely low distortion and it’s compression is almost “invisible”. Download the freeware over there at tokyodawn.net.

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…]

major mkIII update for Density bus compressor released

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announcing Density mkIII – providing depth and dimension, mastering grade

As hinted earlier in a facebook post, the TesslaPRO update will be delayed and probably released after the summer break – it is just not there yet. Instead, a major update for the Density compressor plug-in already made it and it will going to see the light somewhere later this month.

So, lets talk about the Density mkIII update. During the last two years or so I’ve received quite a lot of feedback to the Density mkII release – mostly coming from mastering engineers – on how to improve its dynamic response towards todays mastering needs. Some very early efforts did not made it but with the learnings from the ThrillseekerLA compressor design and the emerging stateful saturation approach everything was possible, all of a sudden. [Read more…]

ThrillseekerLA – the short story behind

The Oscar credit for the most addictive GUI artwork goes to Patrick once again.

There are actually two stories behind the ThrillseekerLA venture: One being the creation of a cutting edge compressor design for the digital domain while the other one is about taking a huge leap forward on my journey towards stateful saturation. [Read more…]

the side effects of intermodulation in audio processors

typical IM distortion in a digital compressor

The general and most obvious effect of intermodulation components in audio signals is distortion of course – hence the concept of “intermodulation distortion” (aka “IM distortion” or simply “IMD”). IM distortion and harmonic distortion are two pairs of shoes and must be defined individually as already shown in the short essay about “myth and facts about aliasing” but more on this later on.

The existence of intermodulation components can affect the performance of an audio production in various ways. In the best case, IMD components are a desired artistic effect e. g. to obtain heavily crushed audio effect signals but in the worst and rather common case, they are one of the contributing factors which deteriorate the overall audio quality and might ruin a production. [Read more…]