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dadasane
@RetroMaximus said:
I was hoping that was not the case. I went over the help a few times and was crossing my fingers that I was missing something. Well its a shame I'm going to have to put this feature on hold until I can figure out a way to get the second version working consistently. Thx for the input Ill keep hacking away at it.
The question is... Why do you want to use [tabplay~] over [tabread4~] so much ?
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dadasane
@RetroMaximus said:
I figured as much . I have made a lot of progress since this version. And yes it is a big patch apologies.
Are you getting anything in the console? One thing you can check is the location of the wave samples they cannot have any spaces or even in the file names. that might help.
Thanks for taking a interest in the patch.
I dont know what was wrong when I first tried but it works now for some reasons....Anyway YES, I do keep a wide open eye on your patch Sir... as I am myself just begining with puredata and trying to put together a vintage sampler/sequencer patch pretty similar to yours. (with much more than 16 steps in the sequencer and definatey less pads....lol)
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dadasane
@weightless Do you have an example of the implementation of loading the sample in an array,and recording the output into another array ? I would like to give it a try
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dadasane
I have managed to do what I wanted... And yes I am using tabread
Here is a demo : https://mega.nz/#!8oVn0QQK!ywcBonPZhRAeA7lWpPKRUZGTgziW1woOxelL44AqLUg
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dadasane
So I have searched and searched again and the best way for me to acheive what I want is to be able to modify the playback speed AFTER the sample and hold .... And this is where I am stuck....
Basically my patch load the wav file, then I have a phasor with a hslider to control the playback speed and another one for the sample rate and then I have the sample and hold and then I end up on the tabread and the dac ....
Is there any way to control the speed at which the rabread actually go through the array ? (I am not even sure that it would have the desire effect) ... Once again, my goal is to pitch up or down the aliased signal in order to generate even more aliasing .
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dadasane
Yeah the frequency is by no mean an multiple of the original sampling rate (44.1Khz).
I'll figure something out.Thanks Seb
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dadasane
Hi guys.... I am currently working on putting together a custom vintage sounding sampler in PD....
I realized that most tutorials, exemples and ressources online are very concerned about ANTI-ALIASING (ie how to not get aliasing in the sound).... My goal is pretty much the opposite ... I DO want aliasing and I want the texture of the aliasing to change when the playback speed of the wav file changes... So I am guessing the pitching part has to come after the part that introduce aliasing to the sound...
So far, I am playing with the phasor, samphold and a vslider to pitch up and down .....I would like to know if there are any other ways to do this and/how I can improve the process ?
Thanks in advance (while I keep browsing the web looking for ideas and inspiration )
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dadasane
Thanks a lot ! The admin has manually validated my account... .It seems like a problem with the website and hotmail....
I am guessing I am not the only one....Thanks again !