Hello all,
I am trying to make a granular synthesis operation and have an idea to make it more precise but do not know how to do a few things. Hoping someone can help.

I want to make a granular synthesis object that can play grains in a wide variety of ways. So what was my new concept?

Record the audio that I am going to manipulate.four times sampling rate faster if possible. then play it four times slower which would bring it down four 2 octaves of course. My thought was that I would record at a 44100 * 4 = 176400 sampling rate. Then play the file at 44100 sampling rate. Which would play it down 2 octaves, but the files would not be "grainy.."

Then I could use the granular synthesis operation to play each grain 4x faster which would bring the file back up to speed but would play the sound 4x longer in duration. Does this make sense?

This is coming up because there is too much distortion when I play the file 4x slower and then use granular synthesis to bring the speed back up.

OK, so I tried to use:

writesf~

With the rate changed to 176400. This does not seem to work.

I wrote it like this:

writesf~ -rate 176400

But that did not seem to work. So, I just am thinking, can you use sampling rate higher than set by preferences?

@raynovich

This was helpful.

in the help file for writesf~ it is mentioned that a resampled sound file will not have it's header altered, though it does sample at the desired rate. This accounts for the pitch shifting you're getting. I see three options;

rewrite the sfwrite. A herculean task, and you will win the respect and admiration of the forum.

get out your handy dandy hex editor and manually edit the headers ('wav header' and 'aif header' google searches bring back way too much useful info on what the headers look like.

use your favorite sound editing application to pitch shift the sf up to its original pitch. if you're going from 44.1 to 96, you have to pitch shift the sound by 96/44.1 = 2.17687. Depending on the algorithms the editor uses, you may or may not introduce artifacts.

no four.

@raynovich Reading the help file I think it says that the header is written according to the -rate flag..... useful..? I don't think so.
Pd only uses the header to identify the file format..... it cannot adjust its sample-rate like a media player app to play files at the correct speed.

You will need to actually sample the audio at 176400 before writing the file.
So for the audio to be sampled at 176400 you will have to set that rate for Pd in the Media Audio settings tab..... so that the Pd audio rate is 176400 when you record...... if your soundcard can do it without resampling (you could well be stuck with 192K and 48K as your sample rates for this project).
Then reset Pd to a sample-rate of 44100 and the file will play back 4x slower.

If you want to do it all on the fly you will need to [block~ something 1 0.25] your synthesis in a sub-patch..... while still running Pd at 176400. But you might run into problems re-blocking for [dac~].... Not sure without trying.
Or run 2 separate instances of Pd.
One for recording @176400 and one for playback @44100..
But you will need separate sound cards for each instance so they can be set to different clocks.
I think [pd~] cannot run at a different sample-rate to its parent....... so not useful for this.
David.

@whale-av. Thank so much for the response. I am sampling at a higher rate to get more specificity in the recording. It is working out somewhat well.