Absolute Beginner: BPM Clock + Sequencer + Waveform Meter Questions

P.S. The waveform peaks are the same when it’s a square wave… and vary for all the other ones (saw, sine, etc). Maybe this helps solve the mystery. :slight_smile:

Oscillator sync restarts the wave when the gate triggers. If you slow down the frequency you can see it.

Thank you. I understand now.

If the peaks in the waveform vary does this mean that the pitch varies (or just the level of the sound)? I don’t really understand how to read the oscilloscope.

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This video might possibly be helpful.

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I think all the values vary a little bit in Audulus (again, not 100% sure of this). But it’s so small that it shouldn’t matter in most cases. You won’t hear a difference. It can matter though in a gate signal though. So watch out for that. There are ways of making it exact. I use the expression (g > 0) often.

Interesting. I can see it when I use a metering > meter and play with the ADSR + speed a bit… The level varies noticeably visually (in this case i cannot tell the difference in the audio :)) with identical step settings. I wonder what the people who wrote Audulus have to say about this. :slight_smile:

Btw, I could not see the start gate being not 0 when sync is off and I slow down the clock. I’m not sure how to configure the ADSR to see it in the oscilloscope…

Yeah, strange. Not sure why it’s not working.

Is this what you mean?

I think it’s just the waveform node (aka oscilloscope) sampling at different points in the cycle. It won’t affect the pitch just the amplitude. Also If you record a bit of it and look at it in the DAW I’m pretty sure it will be anticipated behavior.

First off, welcome!

Clock Master.audulus (91.9 KB)

You might enjoy this clock a little more - it’s a part of the upcoming library redesign, which you can read more about here:

In that post you can download the whole zip of the new library.

The first part of this document explains the signal standards and should help with some of the questions you had above:

If this is in reference to the waveform node pic you shared above, it’s because there’s only a finite amount of resolution that the waveform node is dealing with - it takes samples of the audio and displays the amplitude of it. When it’s fast moving (audio instead of an LFO or envelope), it shows an average amplitude by drawing a top and bottom line which is an approximation. Since your attack and decay cycles are set so fast, it appears to vary in amplitude when it really doesn’t. If you set them to longer times with a slower clock cycle, you’ll see that its more consistent.

Here’s an example patch that is close to what you have set up in the original patch you uploaded.

Sequencer Example.audulus (415.2 KB)

notes 3.audulus (92.6 KB)

Got the sync working. Had it hooked up wrong :l

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I got silenced by the forum temporarily as I exceeded the number of replies I’m allowed on my first day. :slight_smile:

Thanks for having me over here!

I downloaded the new library and started reading the docs PDF. Is there a way to install the library (replacing the current one that comes with Audulus) instead of opening them via the finder?

The signal types intro was indeed very helpful. :slight_smile: I will post some questions related to this in the Learn category.

As far as the waveform question: yes, I was asking why the peaks in the image I uploaded are not even. You are basically saying that the readout is inconsistent due to its rate while the signal peaks themselves are consistent? So if I record the output and open it in a DAW the peak would be uniform over time. I will try this.

I will come back to the sequencer business after I cover the fundamentals. Thanks for sharing the example!

Thank you! In the example above the gate out from the sequencer is reseting the sine wave, so that it always starts at the 0 crossing and is going up from there?

I went in an upped your user trust so that won’t happen again - it’s clear you’re going to be a valuable forum member! :slight_smile:

You can’t replace the library, but if you go to Audulus 3 in the menu bar and go to Open Modules Folder, you can place whatever you want in there and it will appear at the bottom of your right click menu.

Yes. What you’re seeing is the limitation of the measuring instrument. Imagine this:

x = where a sample of the incoming audio is taken and displayed on the waveform node
r = rising signal
f = falling signal

x--x--x-- Waveform Node Sampler
rrffrrffr Incoming Signal

result:

rrrffffff Waveform Node Display

So you can see how the output can become unreflective of the input - but over time it basically works fine for what it needs to be. In Audulus 4 we’ll have a more malleable scope that will allow you to change the window size. Right now it’s set to 5 seconds and you can’t change it - in the future you’ll be able to zero in on it closer so get more accurate picture of individual wavelets of audio.

LOL… Thank you! I do like to understand things in detail. Well enough to explain them to other people.

Thank you! That works well for me.

Basically the input is higher resolution than the measuring instrument and the measurements are not taken at the exact same spots of the input signal (which itself is consistent)?

It’s funny that you bring this up… My next question was going to be if there is a way for me to modify the waveform meter module to zoom in on smaller portions of the input as snapshots and replace the current snapshot based on a timer. Ex. it produces a snapshot of 1 second of input every 5 seconds. [which gives me 5s to look at 1s, before I see the next one… obviously I would only be seeing 1/5 of the signal in this way… but presumably there could be a way to offset the location of the 1/5 interval, which is being rendered]

It sounds like you’ve already though through this problem and I can just wait for Audulus 4 to enjoy zooming in on smaller parts of the signal.

This is a hack that kinda does something similar. Scope Demo.audulus (42.0 KB) It uses a S&H and phasor node to sample the input waveform periodically to drive the display. There are two variants, one takes a Hz or 1 per octave signal as a trigger and the other uses a zero crossing detector to set the sample rate. It works best on a fairly stable input waveform.

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Yep!

Yes, @stschoen showed the patch. It will accurately reproduce the waveform so long as you’re not modulating the waveform much - so FM won’t look correct on the scope, but a stable waveform will.

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Did you actually come up with a way to take a hardware sequencer clock and pass it through a non-dc coupled audio interface, then reconstruct it in order to sync up?

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You should be able to do an input expression like pulse>threshold with a knob attached to the threshold input. The AC interface will allow a little blip to pass through and that should be enough. You want the threshold control there because if you just do x>0, when the pulse comes in, it might ring a little and go above 0 multiple times.

Korg has an app for syncing their little boxes that just uses the headphone output to send a clock out - so it’s possible to do it both ways.

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Right, “SyncKontrol.” I was also fooling with MIDI Link Sync, and some others for a while. I had resolved the issue by acquiring the ES-8, but I was surprised that this could be done in Audulus. Very cool.

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Yeah! I covered the basic concept of how to do that in Audus about 6 minutes into the video and you can do the same in modular using a module like Mathes.

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