Hi Guys! I'm currently working on an extended patch that has adapted current granular patches and added extra functionalities. The idea is to create a practical and creative sound design tool; focussing as mic input as the source...

In the main patch both locators for "pd sampler" work due to the stated sample rate of 44,100hz. However in "pd live granular" the right locator does not work! I know this is because "pd live granular" supports a varying sample rate; depending on the length of the recording.

Could anyone point me towards how to modify the right locator so that it functions for any sample rate?

11THAPRILWILLBSPOKEFORFORUM.pd

@willgerwat Hi, the right outlet of [soundfiler] outputs a list with info about the sound file you have loaded, the first element of that list is the sample rate so you can use that in your patch instead of hardcoding 44100.

@willgerwat Hello.
It's hard to tell because a few objects are missing, like [g_tgl on] and others.
But (sorry, I moved some stuff around so it is not exactly in the same place) you seem to have a gap in the calc here....... and the [*~] is not being triggered.

Just a first look.......
David.

@weightless @whale-av Thanks for your responses.

I can see that theres a gap in the calculation. If you look in "pd sampler" you can see the full calculation with full function *(ie the right locator works). *

As i mentioned before where the issue is occuring* ("pd live granular")* the sample rate is constantly changing with every recording. From what I understand of the calculation from "pd sampler", (where the right locator works), the sample rate needs to be expressed and converted to a fractional value in order for the locators to function?

@willgerwat Sorry to be so slow..... it was dinner time....
No, it's not the samplerate. It's the sample size that changes with each recording. [soundfiler] outputs the total number of samples.

So as the faders are all 0-1..... and the array has been resized to equal the samplesize......
Start point = left limit fader value x samplesize
End point = right limit fader value x samplesize
Samplerate will always be 44100 if Pd is set to that.

The phasor ramps from 0 to 1...... and is set to do so in the time that is required to play the range of samples.
So it's frequency is samplerate / (end point - start point) (e.g. ramp every 2sec for 88200 samples between the points....... 0.5Hz) if you want it to play at the correct speed.
Vary that frequency if you want to change the playback speed.

When it is at zero, you want it to output the starting sample number......
..... and when it is at 1 you want the endpoint sample number.
So the output of [phasor~] is multiplied by the "playback length" and then the Start point is added.
So (phasor output x (end point - start point) ) + start point........ is sent to tabread~
I hope that is easy to understand (and correct....... )

This might work........ bspoke.pd ..... or at least get you part way.......
I have not dived into your glitch patch yet to understand what it does..... how it works........... but that math should apply regardless.

David.