I have made a sampler in PD that allows variable sample rates.
The problem I have is that when I record with PD running at low sample rates, it automatically lowpass filters the audio to remove aliasing. I don't want it to do that! I want to leave all the high frequencies in the recording.
I am simply using ADC into THRESHOLD and TABWRITE~.

It's simple to always record at a high sample rate and then convert down to acheive the result I want, but is there a way to record at low sample rates without automatically anti-aliasing?

Sorry, not sure to understand what you're saying. At a given SR, frequencies above SR/2 simply can't be digitally represented, as the Nyquist–Shannon sampling theorem states. So if you don't lowpass-filter, you won't never ever get higher frequencies, just additional crap in the [0-SR/2] hz interval (aliasing).

That's right, I would like to keep the "crap" in my recording.

I don't think that's a Pd thing. It's likely the adc from your audio interface is doing the low-pass filtering. If you want to keep the crap, record at a higher sampling rate and then downsample.

Edit: to clarify, analog-to-digital converters have built-in low-pass filters to ensure that the digital recording doesn't alias. When you set Pd to a lower sampling rate, it's telling the adc in your audio interface to set its low-pass filter so that it won't alias when it records. So if you want the aliasing, you'll have to record at a higher sampling rate first and then downsample without filtering.

I'm on the same idea of Maelstorm, I had to contact Focusrite for a doubt regarding my Saffire Pro 24 and they told me that there are many "safe device" on both the converters on the inputs and outputs of the sound card included high and lowpass filters (actually the lp filter is an active part of any dac converter).
Maybe contacting the manufacturer of your sound card can help you with that.

Thanks for the replies.
Just to be clear (politely) I am aware of the function of the lowpass filter in a DAC…..

So can you think of a way to downsample in realtime as the audio is coming in?
Or will it always have to be 2 separate processes: first record at high rate, then resample?
Thanks for any further suggestions!

I wish I could set PD to a lower sampling rate but somehow fool the ADC that it's still in 44.1!
So that tabwrite~ would happen at a slower rate but without the LPF moved lower.

I suppose I can just automatically start the resampling a few ms after starting the recording and that would work…. just a lot more to do.