Updates, basically. I’m still struggling with a preFIX update due to some technical issues I just can’t get, errrm, fixed (some VST hosts still do not respond to reported delay changes). Well, lets see how many month or years this will gonna take to work around or maybe then those hosts are just obsolete 😉 At least, I’m pretty much sure that we will see two other updates here during Q2: First, there will be a smaller update for the BaxterEQ where small is indeed the proper wording because it’s mainly about an additional but just smaller GUI version. [Read more…]
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…]
Last year he missed the show but this year, Bootsy (Chief Technical Dude of Variety Of Sound) is sneaking around again (unrecognized) in one of the largest music equipment trade shows. Lets see which secrets he obtained by fraud this time 😉
Still today, most developers are sticking to static waveshaping algorithms when it comes down to digital saturation implementations. This wasn’t very convincing to me from the very beginning and in fact it was one of the motivations why I’ve started my own audio effect developments – to come a little bit closer to what I thought what saturation and non-linearity in general is all about.
And so the Rescue audio plug-in was born in summer 2007 and was already an approach to relate audio transient events to the signal saturation itself. Not that much later TesslaSE appeared which was a different exercise leaning towards a frequency dependent non-linearity implementation coupled in a feedback structure. I still really love this plug-in and how it sounds and prefer it over much more sophisticated designs even today in quite some cases. Following, the pre-amp stage in BootEQmkII then focused on “transformer style” low-end weirdness and did feature oversampling on the non-linear sections of the device. A really great combination with the EQ – smooth and very musical sounding. The TesslaPRO thingy sums up all this and puts it into one neat little device with an easy to use “few knob” interface. Don’t let you fool by this simplistic (but so beautiful) design: It already features everything which makes a saturator to stand out from the crowd today: transient awareness, frequency dependency, dedicated low-end treatments. Sound-wise this results in a way smoother saturation experience and a better stereo imaging en passant.
With FerricTDS not only the notion of subtle frequency dependent compression got extended to a core saturator algorithm. Since revision 1.5 I’ve ditched the oversampling based core and included a version which premiered the notion of memory into the non-linearity which transforms it from a stateless into a stateful algorithm. One could basically see this as a system which reacts different on the very same actual input signal depending on the recent history of events (on a very microscopical level). The input stage algorithms which I’ve included in NastyVCS and NastyDLA (both are actually the same) are a cpu and feature wise stripped down version of that to have the basic sound of it already as an option when mixing the tracks and its according fx.
Quite recently, I’ve started to look into implicit stateful models where memory is not applied from the outside of the algorithm but the algorithm itself contains a sort of memory. As an example, I’ve implemented a stateful version of the well-known tanh() function so that it is aware of recently occurred events but provides the very same harmonic structure compared to the original. Given some analyzer plots it even shows the very same transfer curve but in fact it does not limit strictly anymore but allows some minor overshots of some peak signals. Interestingly, the sound appears a little bit brighter (without letting you see that through the analyzer plot) and the low-end appears not to be that hard “brickwalled” but a little bit smoother. Lets be assured that I’m going to follow this path and then lets see where this will lead to in 2011.