Quite a few users have complained about chipcrusher's peculiar 'dry' frequency response compared to what they get with other common decimator plugins. This post hopefully will explain a few things.
Lets say we bypass the bit reduction, distortion and post filtering and only concentrate on the task of downsampling the plugin's input signal. which would be say at 96kHz. and that chipcrusher's re-sampler would be at 44.1kHz, its internal maximum.
There are two important aspects to consider:
1)Typical results achievable using a vintage sampler is very different from 'your typical Bitcrusher VST'.
99% of bitcrushers/decimator plugin out there use the same tired algorithm that was posted more than 10 years ago on musicdsp.org. This method does NOT band limit the input signal prior to the downsampling, it just sample and holds using a counter... any sample!
This is not what classic samplers did. Any engineer with half a brain at least tried to filter analog audio signal so it wouldn't contain harmonics over the Nyquist frequency of the target sample rate!. If you skip this pass, you will get extra aliasing all over the spectrum.
2)Not all lowpass filters are created equal.
All versions of chipcrusher prior to v1.005(available soon) used a CPU friendly downsampling setting which - in retrospect - might not have suited everyone's taste since it was not steep enough for high frequency content.
You can see chipcrusher's default precision somewhere in this animation made using 96kHz -> 44.1kHz with a white noise as source. All the other settings will be available. We have added a new 'Precision' parameter to set the steepness/cpu use ratio you desire. BTW The first picture in the lot is from a "do not pre filter" setting, we offer 6 such settings, from 6 point spline to truncation. Aliases like crazy, but to each is own.
Thursday, October 10, 2013
Saturday, June 29, 2013
Thursday, June 6, 2013
Looks fun? This is what I did for each and every speaker impulse in chipcrusher's Post Processing section.
The goal is to capture not only the frequency response of the speaker itself, but also the effect of its casing and internal components: resonance, cancellations etc.
Its thus very important to make sure to properly close the unit (which can be complicated by the tight confined space) with your soldered speaker leads dandling out without changing the tonal balance of the unit.
A few carefully created test tones are then 'injected' through the leads and recorded with one or more microphones in a mostly anechoic space at a few inches from the device.
Next the recordings are processed with custom software.
Once thats done, and we are sure the recorded IR data is valid, we need to do the inverse: reopen, unsolder,
close and make sure it works. While the microphones are set up I usually also record native console sounds (games or test code), through that same setup, for later comparison.
Luckily no unit were destroyed, and everything worked just like it did before.
(But you can imagine the stack of devices that I have in the office and in my basement)... there is a psychological condition for that, and also a TV show about it.... Rest assured I ONLY keep tech stuff. :)