Thanks for the answers. I am using the latest versions of Asio4all and Pd. I would like to ask you a question that may be a bit far-fetched. Right now I use Pd to process white noise using [noise~] with plenty of [osc~] for "ring modulation" and going through a multitude of filters. I use [vline~] and [line] on most oscillators and filters to change the frequency. My question is: Would it be feasible to use the MME controller to process all this without problems or is it totally inconceivable? I know that when recording audio or playing a MIDI instrument it is almost unthinkable to use MME when ASIO exists due to latency, so I imagine that latency would affect me too when working with [vline~] and [line] or Could I be missing something in terms of how these tools work? Thanks greetings!

Hi, I am using pure data with the asio4all driver. Sometimes when I open Pd there is no output driver selected and I have to select asio4all again. Does anyone know why this happens??

Some time ago I bought an apogee interface that its controllers don't get along very badly with Pd and I'm forced to use asio4all, but I don't feel it gets along with pd either. All these problems with asio drivers come since I upgraded to windows 10. I would like to know if any of you have had similar problems with windows 10 and Pd or if there is a certainty that windows 7 has behaved better with pd than w10. I'm at a point right now that I don't even want to imagine the problems if I upgraded to w11. I wish it was just my problem, but I'm afraid it's not.

So, if I understand correctly, and taking into account that I have been using Pd with the same equipment and the same interface, and the only thing that has changed is that I am now using Windows 10, I must think that where the conflict is in "Pd and windows 10"??

If so, I have a big problem because for other reasons I will not return to Windows 7 (although I really want to)

Another thing I want to comment on is that the BIOS is possibly misconfigured because the battery is running out (when turning on the PC the message "the Bios has been reset" appears and the clock time is also slipping)

Due to my ignorance, I don't know if this could be related to the problem, although as I already said, all the software I use doesn't have any problem...

I will continue to investigate even if I have less and less hope...

Thanks to all for the help!!

The problem keeps happening even with the dsp turned off. I have created an empty patch with about 50 "bang" buttons to test, I have dragged one of them with the mouse and it is a horror to see Pd in this way... I would not dare to say that it is unusable, but it is close. ..

I will continue to investigate the ASIO driver issue further as you have suggested.

Thanks for your help.

Hello, first of all I want to say that I have been using Pd for 6 years on a PC i5 (fourth generation) 16gb ram and Windows 7 ultimate. Pd and other DAWs I have have always worked without any stability problems. A year ago I upgraded to Windows 10 home (I have always optimized Windows following the indications offered by Merging Technologies and some professional DAWS) Today all the software related to audio works perfectly and without any problem except PS that drives me crazy.

Just create a new patch with 2-3 tables and 2-3 signal objects to start the instability in Pd.
The first symptom that appears is the "feeling" of "overloaded". For example, when dragging an object with the mouse, the object is delayed with respect to the mouse pointer (the more the object patch is loaded, the more noticeable the delay is)

If I save the content of a table of only 8820 samples for example (200ms) with [soundfiler], it can take between 2 and 3 seconds to wait until the message appears in the console

It has also happened to me (although on fewer occasions) that when I press a bang button, the button remains pressed in black and Pd blocked (although I am not aware that this problem is directly related to the other comments).

I want to make it clear that I have not had any problems related to the audio, it is with the GUI that makes me think that something is not right. The first of the problems that I have named is the one that I consider serious for me, because it prevents me from working at ease and in the end I end up not going ahead with any patch. This happens to me in version 0.53.0 that I have now and in the previous 4 versions. I have used the same patches on a laptop that I have and everything works normally. I can't get to understand what is really happening and what the blissful problem may be related to, I have asked several Pd users on telegram and they have not been able to help me either...

Any ideas that occur to you will be welcome. Thank you!

I'm going to try, thanks David

other than linux

And do you have an idea how to patch it in pd? especially the exit to the phone

Hello,

Can you talk on the phone and have the listener hear the voice previously processed with PD?

Now I see it even clearer !!
Thank you very much for the help and for your time !!

A greeting!

Hi

It is not easy for me to explain why I need maximum precision, either because of the complexity that would be explained to me and because my English is not entirely good. Although if I can try to explain a simple example (although it is far from the real purpose)

Suppose I have 2 audio samples. The first sample lasts exactly 1000ms (44100 samples) and I loop it every 1000ms.

If I had another exactly 125ms sample and looped it back at its exact time (125ms) alongside the first 1000ms sample, the beat would be perfectly in sync since it's the exact eighth part of 1000.

On the other hand, if I can only get a sample of 5512 samples (124.988..ms) and I loop it along with the 1000ms sample, as time progresses and the 124.988 sample is repeated more times ... ms, the desynchronization of time will increase ... (rhythm)

I know you can tell me that I can solve the "desynchronization" by repeating the sample every exact 125ms, but I have already said that this is just a simple example to try to explain and does not address the real problem.

As I have understood from your explanations, that sometimes it will be impossible for me to record samples to the exact duration that you want, but I can reproduce them within Pd exactly with the desired duration using the examples that you have given me. What invites me to think about the idea of ​​not using any other means than Pd to get closer to my purpose, even if it takes more time and effort at first; But it is what I will do ..

Thank you

A greeting!!

t was impossible for me to record with [writesf ~] respecting the actual duration. (I can't even get to record 44100 fair samples)

I don't know if I use [block ~] incorrectly, but at the moment I have only got a real precision, using soundfiler (although this limits me since the table only accepts whole samples as we have already spoken ..) So, until I get that [writesf ~] record correctly, it is useless for me to use [tabread4 ~] or [tabplay ~] ...

By changing the frequency to 88200, (both Pd's and my interface's) and trying to draw waves on a table with [tabwrite ~], for some unknown reason, Pd crashes me ..

So I will leave it for the moment and I will return to the problem again later.

Thank you!! Greetings!!

although it would not solve it either, since the "stop" message would only work from sample to sample, and not between samples, right?

Hello,

I still have the same problem recording. I cannot achieve maximum precision with PD.

For example:
If I record the contents of a table containing a 1Hz sine wave, with a table size of 44100 samples, [soundfiler] records the 44100 samples perfectly (1000ms).

But if I want to record an 8Hz wave (5512.5 samples) (125ms), the table will only accept 5512 samples, (124.988..ms)

This time difference may seem insignificant, but for my work it is becoming a horror.

You have advised me to use [writesf ~] in an subpatch with [block ~ 1] but I don't know exactly how to do it. (Sometimes I learn better with an image than in writing) and I have not seen clear examples of [block ~] in the help file.

Any ideas? Thank you!!

Agree!!

Thank you very much to the 2 for the answers!

A greeting!!

Hello,

When I use [writesf ~] and I send "stop" with a low delay time (as in the photo) it doesn't record me respecting the indicated time. (about 3.6 ms ahead) If the sample should last 35.19274376, the end result is 34.82993197 ...

This becomes a serious problem for me because I cannot work the way I would like.

If I record the contents of a table with [soundfiler], the problem disappears, but I have not been able to record at 24bit (as with [writesf ~])

I would like them to tell me what I am doing wrong and how I could solve it.

Thank you!!