@whale-av RIght!

Yes, I should do that to see if there is a difference, but I doubt it would be noticeable.

Is there a difference between having two audio sources going directly into an object and using an additive object.

Essentially something like this:

|output1~| |output~2|
|                  |
| _______ /
|/
|*~ 1|

As opposed to:

|output1~| |output~2|
|                  |
|               /
|          /
|       /
|+~ 1|

Another way to explain this:

If you take two outputs put them in the left inlet in an object: |*~ 1|, that would be the same as one audio output into the left output and a second audio output into the right output of this object: |+~|

Hopefully this can should what I am trying to ask about. I do not think there is any difference between these two ways but want to be sure I am not missing something.

All of this has helped. Thank you so much.

Thanks. Where do I get ELSE? I do not mind using externals.

Thank you. This does seem close to what I want.

It is not working because the |filter~| does not exist in my library. Which library is that from?

I tried the link to www.katjaas.nl/ website but I get this:

403 Forbidden
Request forbidden by administrative rules.

Hello all,
So, lop~ attenuates the frequencies above the cutoff frequency, but it does not really take away that much. I always thought that lo pass filters were more cutoff filters and not attenuating filters.

Anyway, I linked a number of lop~ filters together on audio with the same cutoff frequency which did attenuate the frequencies above the cutoff more so. Is there a more elegant way to do this? Is there a patch?

Lastly, is there a reason the lop~ does not cutoff frequencies above the set frequency?

This all applies to hip~ as well.

Hello,
I am listening for any problems and do not hear any. However, is it ok to use |line| to change the delay of |delread~|? I am surprised this does not create any issues.

Yeah, I was thinking that this might not be possible or practical in Pd.

No help? Was my question clear enough?

Hello all,
New project I am working on which I would like to consider using additive synthesis. I do not want to overburden the CPU and will record the piece in stages.

Ok with that in mind, you can manually or automatically record an osc~ sine wave into an audio file in real time or even multiple oscillators at the same time.

But if you are to do that, then it would take the same amount of real time to record each portion of the piece and then put all of the pieces together. If the patches got really complex this could take days of computing time.

To be simple in showing this:

|osc~440|
|
|
|writesf~|

If I created a file name test.aif, and banged start as message to this, with a delay of 30 seconds to a stop.

A file named test.aif with a sine wave at 440 hertz for 30 seconds would be created, but it would take 30 seconds to create. Is there a faster method? If not, there should be. No?