Thanks for the reply but I don't think biquad~ can accept signal rate inputs, only message rate. Or am I missing something?

Why don't you just take the first and last values and draw a line between them?
Or sample a few more values if you want to approximate the original data a bit more?

I am really enjoying some of the filter abstractions I have found here, but they are quite CPU intensive so I am looking for good externals. So far the ones that sound good only accept message rate signals to their control inputs. Please recommend filter externals with signal rate inputs, at least for cutoff. Preferably LP/HP/BP/Notch and EQs would be nice too. Thank you all!

...but something like a Kurzweil K2000 has plenty of aliasing and seems like a pretty naive implementation of oscillators and filters but mostly sounds great, at least in the lower octaves.

Hi. Your filter looks interesting. I have an Xpander here, the filter is probably not as exciting as you would imagine But still a nice synth...... anyway, on topic: I just wondered, I mean I have digital synths that sound good, from the Yamaha DX era up to modern softsynths. I just wonder what is going on under the hood besides the main stuff. I think my problems are mostly caused by aliasing but I can't really oversample much because I'm running out of CPU. Perhaps I'll reduce polyphony to 6 notes and try some different techniques for bandlimited oscillators.

Yeah, funny that they keep putting off fixing it, it seems quite an easy thing to repair. Who wants backwards compatibility to something that's broken?

This is interesting. But surely a [hip~ 5] ought to enough to remove a DC offset. I don't understand how DC offset would cause oscillators to drift apart...???

I want to use this filter because it sounds good and has much less CPU overhead that making filters with biquad~ / czero~ etc. But the cutoff & resonance inputs are via a [pack]ed set of floats. What's the best way to use this if I want to modulate the filter with LFOs or envelopes that are at signal rate? Just use [snapshot~] and a fast [metro] and perhaps a [line] to smooth it out? Or should I find a better filter external?

@alexandros
I'm using your "multifilter" abstraction among other things. sounds great, thank you! lop~ isn't really very exciting for a synth and not much use as a brickwall for antialiasing either.... biquad~ maybe I need to learn about
@seb-harmonik-ar
Thanks for the nice clear answer. I will try out some externals to try and reduce CPU load a bit.

I have made a polyphonic synth in vanilla PD which is pretty great but it has a lot of aliasing. When I start to implement oversampling I quickly run out of processor time and start getting dropouts.

I have never used externals.

If I use filters from a library like iemlib, will this make my synth more efficient?
If so I will look for libraries with anti aliased oscillators and nice sounding filters....

Or is the efficiency just the same, and some people just find externals more convenient?

Thanks. I was just wondering if it's "standard practise" for people making DSP synths to have an EQ on theoutput that's concealed from the user.
I've been mixing records for 25 years so I'm comfortable with using EQ but surprised by the strange peaks in my Pure Data sounds.

Thanks for your thoughts. I will work more on anti aliasing.

I have been making some quite complex synths which can make amazing sounds but when I try to use them in recordings, they often need a lot of EQ,with strange bumps in the upper mid or lower mid regions compared to my hardware synths.
Do you tend to have any overall EQ or filtering before the output of your PD synths?

Some of it's possibly to do with aliasing.... but not the low mid stuff.

Or you can send a "clear" message to clear a whole subpatch.

I am fairly relaxed about aliasing but I am using a lot of non-linear stuff like FM, wavefolding etc and it still seems like the most efficient thing to do is upsample/filter/downsample.

Thanks for the reply. I was hoping for thoughts about what's cleanest to save me doing the analysis myself
I am doing 8x oversampling to try and alleviate aliasing and wondering the pros and cons of each one in terms of both sound and time/CPU efficiency.

Today I stumbled across the help file for "inlet~ / outlet~" which contains some interesting info about how PD performs upsampling.

Basically you have a choice of sample & hold, zero-padding, and linear interpolation.

Can anyone comment on the pros and cons of each option?

Thanks for the fascinating link.

Thanks, will check it.

Yes butterworth sounds good but uses a lot of CPU.
I am usually filtering around 18kHz, I can't hear up there anyway.