Hi, I am at the moment stuck with a 44Khz sample rate, which is fine with the sounds I'm working with (in the 800-7000Hz area) but which makes 1000 sample ffts too slow. So I wondered if anyone's done audio interpolation with Pd or how you'd go about doing this. Right now the only option I'm aware of is writing an external with the interpolation and try to spit out the audio at a higher rate than it comes in. Will Pd at all allow for this, or is it locked to the sample rate of my hardware? I've been googeling for externals that do this already but found none. For the interpolation I was thinking of using line~, but it doesn't seem fine-grained enough, asking for milliseconds.
Speaking of fft, can anyone point me in a direction of how to narrow the analysed bandwidth down to the 800-7000Hz area that I'm interested in? Still reading up on it, I'm hoping that should let me get a higher precision with less data.
Sorry if this is all very newbie.
Cheers
Nik
Niklas Saers wrote:
Hi,
interpolation with Pd or how you'd go about doing this. Right now the only option I'm aware of is writing an external with the interpolation and try to spit out the audio at a higher rate than it comes in. Will Pd
this will not work at all.
at all allow for this, or is it locked to the sample rate of my hardware? I've been googeling for externals that do this already but
no, pd can use quite any samplerate (but your hardware might not support it...)
found none. For the interpolation I was thinking of using line~, but it doesn't seem fine-grained enough, asking for milliseconds.
no this is something completely different. [line~] is to convert messages to signals (with slopes), it has nothing to do with interpolating signals.
what you are looking for is called "upsampling".
pd's dsp tree needs to have one sub-patch to have a fixed size block-size, overlap-factor and sample-rate. (because that is the way how pd works.) this is the main reason to not being able to make an external that outputs more samples than are coming in (i think max is different here)
anyhow, you can do upsampling (and downsampling) with the third argument of the [block~] object.
there is a help-patch in 3.audio.examples/ that explains how to use this feature.
mfg.ads.r IOhannes
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IOhannes m zmoelnig -
Niklas Saers