Good Morning PureDaters!!!

I'm here other time asking about some pure data objects which i don't control at all.

As I told in another post, i'm currently doing my final project for my degree. It`s about sound effects developed at Pure Data. At the moment i have done a compressor which sound really better than the other ones which are a reverb implemented as moorer way, and a parametric eq.

I have having problems with some objects which i think doesn't run as they should.

They are send~ and receive~ and in the other hand throw~ and catch~. The problem is that i need to send the signal sample by sample because i need to delay the feedback signal just for one sample. The matter is that when i try to send it blocking the patch for one sample pack with block~. The problem is that when i use those objects appears a problem which say:

sigcatch 1243-signal1: unexpected vector size
throw~ 1243-signal1: vector size mismatch

or

sigsend signal1: unexpected vector size
receive~ signal1: vector size mismatch

Then i would like to know if exist another object or way to make it runs.Or if is it a bug in Pure Data. In case it would be a bug i must write it in the memories of the project. Then, i would like to be sure about what i will write.

Add to it i have another issue which is about the play objects. I'm working at the moment in a new patch to make a delay effect which runs which the tirhm of the song. I started to do it with something like a click which will let to configure the delay by each beat. Then, if i have a "¿PATH?"(i'm not sure it would be the word) which have any time or beat for instance 3/8 in 85 B.P.M. i would like to have 3 delays at the speed of the quaver note rithm.How ever i can't find an object which permits the sound flows when i play for instance the second of the three consecutive delays.However all the objects i have tried stop the sound when i try to play another delay above them, and if i try to do it whith consecutive delays i can't completly stop the 4th delay. Then i would like to know if anybody knows what i could do.

Then, those are my two issues i have today. I hope anybody was able to help me with them.

Thank you everyway. And excuse my english skills but i'm learning. If anybody want correct my enlish i'll be so glad.

Hi! Good Morning!!!

I'm writting for second time at this forum because i need your help one more time.

As I said in the last message i'm doing my final proyect with Pure Data. In this case i'm trying to develope a Wah-Wah effect which is based in a parametric filter which is moved in its main frequency along the time.
I found a path in this forum a few weeks ago which is called autowah.It's based on a parametric filter which change its main frequency according to a LFO which we can change its frequency and shape.

Then, here is my problem. I was developping this parametric filter wich i told before basing it in a paper which i found surfing the internet. The problem is that after i developed it in Matlab (a mathematics software) and run it right, i tried to translate it to Pure Data enviroment and it didn't run. I´m not sure if it's a patch's fault or a Software fault, because i couldn't fix the problem whatever i tried.

Explanation of the problem:

i have it implemented as i implemented it in matlab, and the filter coefficients are right if you compare it whith the Matlab ones. Then, it should work however it doesn't run. I controlled the problem and it appears when the output signal is put in the delay inlet which makes the main loop. The filter is based in a direct line from the main signal imput wich is summed to a delayed one sample signal, and two sample delayed one from the same signal. Add to it, the output is delayed in the same way one and two samples, and summed to the rest of the signals before the output. The problem appears when this output signal is delayed and summed at the end. Then the mean of the output signal (calculated with env~) is over four hundred dB and at the end most of the signals are inf# or over four hundred dB, except the main signal ways which bing values near to ninety dB. Then, I have tested the delays whith impulses in their entrance and they run well.

Thus i'm looking for your advices to make it runs well. The filter is similar to vcf~ but i need to do it by myself.

I left a link to my personal DropBox account where are all the files you need in Pure Data or even in Matlab(i have a problem uploading the things to this page). Add to it there is a video recording which explains perfectly which is the problem. (it is in spanish (i'm sorry)).

There you could find a carpet with the autowah.pd files found in this forum.
Another folder which contains the matlab files where the main one is which is called pruebaeqbanda2.m. In the last folder you could find the pure data files where the main one is called pruebafiltrobueno.pd. Add to it there is a .pdf file which explains the mathematic teory behind the filter and a video which i show how it works.

Thank you before your help, and excuse me about my english (i promisse i will do it better for the next time)

link: https://www.dropbox.com/sh/8mlb5xnnf5l8qot/OJ4IBX6OgG

Hi people! How are you?

At first sorry for my english, i'm just a learner.

Next to it, i would like to explain my problem:

I'm working in my " end degree project ". It's an aplication in Pure Data which at the end will be connected with a kinnect sensor. The kinect sensor will serve to change some parameters in the pure data's Fx.

Then, i need to make some sound effects in the beginning. I have started with a "Compander" in which i have wasted much time.

Then I have a lot of questions that i can't solve, and they are making my work so heavyly and hardly.

My first question is about [env~ $f1 $f2] object, which has two arguments into. Could those are changed? i can't understand so well how it work. Is the first argument the window's size? is it then the second one? I have tryed it in everyway i knew.(massage boxes, number boxes,...)

The second one, is about the program and its objects. I couldn't found anywere a manual enough useful or enough good.
I would like to know how to read all the out' arguments in all the objects.
For instance i don't know, how to read anyone argument in [openpanel]. I know it gift at least the size of the ".wav" in samples, or anyother property. Then, i want to know if is possible to use an object which tell me the properties of another object.

Add to it, i want to use some times the same effect in the same string. An example: At first we are going to use a compressor, next to it a reverb, and at the end the compressor another time. I think the variables and links names (senders and recievers) are going to bring me some problems. Then I wana know how to do it possible.

To say good-bye i'm lefting the patches of my "Compander Project" there. It's a broken sistem now. However i will try to finish it before the sunset.

Thanks everybody, and excuse me if you couldn't understand me so well.

i forgot to say that when i'm using any of the linking boxes ( send, receive, throw,catch,... ) I have an error which says:

sigcatch 1243-signal1: unexpected vector size
throw~ 1243-signal1: vector size mismatch

or

sigsend signal1: unexpected vector size
receive~ signal1: vector size mismatch

if i link with a direct wire from the patch output to the inlet of de delays into the patch which we need to delay the output signal, appears an error which say that is being a loop, and it make that the send of signal stops working.

The patch is now aviable. if you couldn't download, please, say to me. The main patch is called "yafunciona4.pd".

Thanks very much and read you soon.

Hi! Good morning another time! (I hope you are better than me)

I'm becoming crazy about my patch. It runs, however it doesn`t run as i want.
I've tried the throw~ and catch~ boxes, i've tried to use send~ and receive~ boxes and even i tested it with delwrite~ and delread~ how ever none of them runs as i want. I know the problem must be in the buffer length because i can't have a narrow peak in the frequency as the vcf~ and bf~ do. I know maybe i did not develope a filter whith order higher as they have, how ever, when i test it in matlab there is no problem. In addition, in matlab, my filter amplify the signal and attenuate it, how ever in pure data, signal is just attenuated in the main frequency.

I would like to know how it runs like that, how ever i can't find and fix the problem.

Have you any other idea? If you have anyone i would like to hear that as crazy as it looks.

Thank you for reading my bad english.

I hope the new files will be uploaded to the link above before four o clock.

Thank you very much Maelstorm!!

You were right, i have achieved to make it work with your advices, however i had to use another delay line for the main stream, and i delayed just for one sample the other ones , because the other way runs worse to me(maybe i didn't know how to run that).

I promise to upload that patch improved in a few days. I hope to use all your advices in the new implementation.

See you (every body) soon!!!

i'm trying to upload my patch, but i ca't