I'm having some trouble getting the line box to function properly in its intended way, where it rises (or falls) to a certain value within a set time frame. Instead, it has a brief period of distortion before reaching the intended value.

Is this simply something that can be fixed in the media tab, such as a different sample rate or block size? For reference, I have a MacBook Air running on macOS Monterey, and I'm using pure data "Pd-0.56-2".

so if you print it out , it doesnt make a smooth list of numbers? if youre using line tilda~
it needs to be multiplied with a tilda multiplier ~ Ive seen that I can hook a regular multiply into my volume fader line for scaling the output and then I get a slight distortion. so then I switched to multiply tilda and its better. otherwise you should do the grain number. and test different combinations .

@willblackhurst tilda is like a wave on top thats from somewhere in your computer/voltage. so it needs a matching system with the tilda voltage syncronized I guess.

@Hello_Hello Welcome to the forum..

Please upload your patch. It is not normal.... so we need to see why it might be happening.
When you start a post you will see an UP arrow above where you are typing...... click that and browse to your patch to upload it.
David.

@willblackhurst Tilde objects don't have anything to do with voltage inside the computer.

hjh

how do they work? I dont really know.

@willblackhurst It's all numbers in a computer... digital.
As simply as possible.........
Tilde objects set processing of the digital audio thread in Pd. The digital audio is a stream of numbers representing the height of the equivalent analogue signal wave..
These numbers only become a voltage..... an analogue waveform when they are passed to the DAC... the digital to analogue converter.... your soundcard output.
The ADC is where the analogue signal is converted to numbers for input to the digital audio thread.
Audio files are stored on the computer disk as numbers.....
The numbers are read, streamed, processed and output at the sample rate.
The not simple answer can be found searching for FFT.
David.

I heard you wouldnt need tilda then . because all you would do is read the numbers as usual. and that its because your computer was never fast enough to do it so theyve put a ramming mechanism that just shoves the array through at a constant rate.

@willblackhurst " snaphot " never makes any sense to anybody trying to get an accurate number. because there isnt really a number to look at somehow. so whats going on between an array list - and snapshot, why doesnt it just use the array numbers? or give you exact numbers at exact times etc etc.

@willblackhurst For the DAC the numbers have to be pushed out at the sample rate to create the analogue audio wave.... or it is just a meaningless data dump. And there is no "jamming".... it is just that the data then moves through the audio stream in Pd at that rate.

When a wav file is copied on the computer the data is of course moved as fast as the storage medium and the processor and the RAM I/O can manage..... which is also the case when you load audio to an [array].

[snapshot~] does what it should... outputs the audio stream value at the time that it is banged...... but because control rate operations are sent between blocks it reports the last value from the previous block.
So it does use the array list data if the array is being streamed (played) but only reports that one value at the point to which it is attached in your patch and at the time that it is banged.
And if you sample the stream from [tabplay~] after a [* 0.5] then the value will be halved.
If you want to see the whole previous block then use [print~] but that only reports the values to the console so it is not useful except for analysis when understanding how you want to use [vsomething~] objects.

How Pd works... is in your Pd folder....... \doc\1.manual\resources\chapter2 ..... part 2.5.2 for tilde objects (audio processing)...
David.