Element’s Sampling Filter

[nimrodadar]

Does element features an option to choose the type of sampling filter it uses? (Minimum phase, linear Phase, iir based, fir based etc.)? If it is, where do I find it? if its not, which sampling filter does it use?
Thanks!

[mfisher31]

Element doesn't do any filtering of its own. Are you talking about an e.q. or something?

[nimrodadar]

No, in every a/d conversion there has to be an anti-aliasing filter
if we’re getting an analog signal we have no way of knowing if it’s bound by 20k, since we’re sampling at 41,48,96 etc, there might be frequencies above our hearing range (especially if saturation is involved) which are not sampled accurately since they’re above the nyquist limit dictated by the sample rate, and the computer reads them as lower frequencies in the hearing range which adds unwanted artifacts to the sound- aliasing. To solve this problem before the conversion devises use a low pass filter to bind the signal by 20k+-, as with all filters, linear phase filters are more “cpu expensive” and can introduce pre-ringing, and minimum phase are more efficient but introduce more change to the phase of the signal.

So my question essentially : are elements anti-aliasing pre/post-sampling filters minimum phase or linear phase?

(since I want to use element for oversampling in order to when necessary minimize accumulative aliasing from saturation processing on tens up to hundreds of different tracks in projects that run on 48 sample rate)

[mfisher31]

Element doesn't do any filtering on the input and output signal. Maybe check with the manufacturer of your audio interface?

[mfisher31]

It uses JUCE audio to run the audio device, and I don't think it does any filtering either.

[nimrodadar]

When resampling (which element does since it lets you choose a sample rate) there will be an anti-aliasing filter added. My understanding is that every audio software needs to sample for itself so there’ll always be an anti-aliasing filter imbedded within the system. Also I’m 3rt year into an electrical engineering degree and in the Signals and Systems standard course filtering is a part of every d/a a/d conversion.

In audio for music oversampling will be used for this case predominantly - a saturation plugin cracked hard adds harmonics that reach well beyond 20k and those who are beyond 20k we don’t need, the plug-in operates on sample rate which is too small so it can’t read those frequencies above 20k and they will be aliased and return as artifacts below 20k. If we oversample for the plug-in so it can deal with let’s say up to 80k, it keeps the frequencies up 80k as they really are and then before we downsample back to a 20k frequency limit we cut all frequencies above 20k, that way getting rid of the previous aliasing.
There are many plugins that offer oversampling, and some specific ones even offer changing the anti-aliasing filter type, but I’m searching for a host for a chain of general plugins that can do the oversampling for them, so I wondered if such a thing is possible with element

I’ve checked out JUCE and there is a sufficient amount of discussion about anti-aliasing filters for specific applications on their forum, but since it is a general platform for making audio software i don’t know what kind of a design within JUCE element specifically use.

Anti-aliasing filters are not exactly eq in the sense that generally developers don’t have a lot of reasons to talk about them with the costumers, they’re supposed to work transparently and smoothly under the hood, and frankly it isn’t such a big deal for me, it’s just that if I’m already going for the trouble of oversampling saturation and compression I might as well also go over this little detail

thanks again

[mfisher31]

When element changes the sample rate, it is telling the audio driver to change sample rate. After this, the audio engine re-configures itself to process the new rate without any resampling. Each node has an over sample option though. Right click one, it's in the menu.

The node oversampling uses a Half Band Polyphase IIR, but there are other options that could be parameterized.

element/src/engine/oversampler.cpp

Line 43 in b901160

processors.add (new ProcessorType (channels, f + 1, ProcessorType::FilterType::filterHalfBandPolyphaseIIR));

[nimrodadar]

That’s the answer, half band IIR, it’s a minimum phase filter, can the other options be easily excessible through the ui?

[mfisher31]

Yeah, sure. If you're looking at taking a stab at it, src/ui/contextmenus.hpp is where the over sample menu code is. With new menu item, it then could call another new function on element::Processor to change the filter. With all that said, both of those areas are something that needs badly cleaned up. e.g. oversampling code in general shouldn't directly live inside element::Processor, and the context menus aren't the easiest to work with (due to a bad initial design).

I'd be ok with adding this as an option, but not if it requires adding tons of new code to element::Processor. Would have to go looking to see what's really required, but as far as I know it's the one single parameter utilized in the juce::DSP stuff.

[mfisher31]

Issue created: #728