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.

NastyVCS – behind the scenes

NastyVCS tube Planning

When preparing the final bits of an upcoming plug-in release, other research and prototyping typically has already been done for other potentially upcoming projects. This usually is the time when the decision is made about the next project and the new plug-in gets outlined and planned in detail. In this case it was around 9 month ago in August 2009 when FerricTDS (our entry for the KVR DC ’09) finally went into the late beta testing stage. Since quite some time I had in my mind to improve the “Nasty” plug-in series package but to that time it had become finally clear that the next step would be to put it into one single and consistent “console strip” style plug-in. [Read more…]

FerricTDS 1.5 – public beta (closed)

The public beta for the upcoming FerricTDS 1.5 has been closed today – thanks for all the contributions! A final version will be released here somewhere in March.

[Read more…]

FerricTDS 1.5 beta, when?

I’m currently preparing the bits for the upcoming FerricTDS beta test. Since this beta is going to be public nobody needs to apply for joining the party, just come back here during the next days or so, pick up a copy and have a ride with it (and give feedback of course). The version number jumps from 1.0.2 straight to 1.5 which indicates some vast improvements under the hood. I’m already really excited about this one.

SSE1 compatible systems – some testers needed

I do need some more testers with SSE1 compatible systems (e.g. old AMD Athlon series) – if you can perform some tests for me with a prepared Density mkII version please drop me a short comment here with your email.

Density mkII – bugfixes, which and when?

Just a short update on whats going on with some already reported issues concerning Density mkII (released around two weeks ago):

  • There will be a SSE1 compatible version
  • The CPU spiking “denormal” issues and heavy CPU loads are already fixed
  • A fix for the loading issues in Wavelab is currently under investigation and test
  • Some crash reports will be further investigated

The current plan is to release a 2.0.1 bugfix release to end of September containing whats fixed and stable then.

Density mkII – reworking the audio engine

While working on the Density overall re-design I was working extensively on the audio engine as well but just for one single reason: The one thing I was missing in the original design was to have some more “responsive” gain riding possibilities but without the usual tradeoff of introducing more distortion or compromising the transparency. And (unsurprisingly) that turned out to be not that easy.

Developers friend: the oscilloscope

Developers friend: the oscilloscope

[Read more…]

in the lab – audio listening tests

AKG K 702 and the Benchmark DAC 1

AKG K 702 and the Benchmark DAC 1 (behind)

Most people have no idea where probably most “development time” is actually spent: it’s in the audio listening tests and not the coding.

[Read more…]

compressor gain control principles

A short compendium on digital audio compression techniques.

Basic compressor configurations

Compression vs. limiting

Technically speaking the same principles are used in audio signal limiting and compression processors but just the transfer curves and envelope follower settings are different. Ultra fast attack rates and high ratio amounts are used for limiting purposes which causes just very few peaks to pass on a certain threshold.

In digital implementations limiting processors can be more strict due to look-ahead and clever gain prediction functions which guarantees that no peak information passes the threshold. That is called brickwall limiting then.

[Read more…]

Ableton Live – the ultimate creative production tool?

Some time ago I’ve become part of a music project which is heading more towards the dancefloor compatible side of electronica. We are currently testing Ableton Live and thinking about switching over to use it as the main production tool. I’m undecided yet. What do you think about Live – is this really the ultimate solution for electronic based music production? What are the known drawbacks?