djtoshi182

I am trying to make a simple patch for teaching about audio sampling rate and bits difference to students, using live control and oscilloscope visual.

also, audio quality and dithering for wav and mp3 export.

could you help me to design a simple interface for teaching to students on these subjects?

I am trying to make a simple patch for teaching about audio sampling rate and bits difference to students, using live control and oscilloscope visual.
also, audio quality and dithering for wav and mp3 export.

teaching-material-for-sample-rate-and-bit-explanation

What do you mean by 'live control' ? The ability to change those parameters on the fly ?

how to analyze the sampling rate from the sound input?

yes. I am looking at degrade~
how to analyze the sampling rate from the sound input?
also, from the phasor~ noise?


I have tried this without degrade~ but by trying to virtually resample a signal with various objects (snapshot~, sah~) and have yet to get anything like the results I'd expect (there should be some old threads around). Pretty confusing, but hope to get back to it at some point soon.

I have tried this without degrade~ but by trying to virtually resample a signal with various objects (snapshot~, sah~) and have yet to get anything like the results I'd expect (there should be some old threads around).  Pretty confusing, but hope to get back to it at some point soon.


On a slightly related note, you might want to play this movie to them too. It is very interesting and related:

I built this patch to explain digital audio principles to my students.

I didn't have much time to optimize it, but it gets the job done.

Hopefully, some of you'll find it useful.

in trying to explain the digitalization process (A/D), we have to represent analog (continuous) audio through a digital (discrete) signal, therefore the premise is:

- "normal" digital audio (44.1/48 kHz, 16 or 24 bit) is the "analog" signal;

- the resulting digital audio has to be of a substantially lower sampling rate / bit depth in order to see and hear the artifacts of the digitalization process.

If any of you decide to improve this, post it so we can make this topic a valuable resource in the future regarding this subject.

I built this patch to explain digital audio principles to my students.
I didn't have much time to optimize it, but it gets the job done.
Hopefully, some of you'll find it useful.

One note however:
in trying to explain the digitalization process (A/D), we have to represent analog (continuous) audio through a digital (discrete) signal, therefore the premise is:
- "normal" digital audio (44.1/48 kHz, 16 or 24 bit) is the "analog" signal;
- the resulting digital audio has to be of a substantially lower sampling rate / bit depth in order to see and hear the artifacts of the digitalization process.

This looks potentially very cool, and certainly along the lines of what I was looking to do. However, I'm also not quite sure exactly what I'm looking at ("monitoring"), and have yet to be able to get the response I'd expect. For example, with the Sig Freq @ 100 and the SR @ 500, shouldn't the signal from the reconstruction filter then match the original? Maybe it's just a question of initially picking numbers that happen work well visually with the Jitter stuff? The guts look straightforward (aside from the Jitter stuff, which I don't know much about), and was close to the approach I had, thinking it would be quick and easy to assemble-- It was anything but that.

This looks potentially very cool, and certainly along the lines of what I was looking to do.  However, I'm also not quite sure exactly what I'm looking at ("monitoring"), and have yet to be able to get the response I'd expect.  For example, with the Sig Freq @ 100 and the SR @ 500, shouldn't the signal from the reconstruction filter then match the original?  Maybe it's just a question of initially picking numbers that happen work well visually with the Jitter stuff?  The guts look straightforward (aside from the Jitter stuff, which I don't know much about), and was close to the approach I had, thinking it would be quick and easy to assemble-- It was anything but that.

With your values, I get the expected results: a relatively similar waveform with the same frequency, and with a phase/delay offset typical of the digital audio process.

Try the following: with the reconstruction filter applied, progressively change the filter value upwards from the initial value of 250 (sampling frequency / 2). You can see the inevitability of the phase shift. The digital values can only be rounded/smoothed by a D/A converter after they are received, or, in other words, they can't be predicted, hence the offset between the original waveform and the resulting one. Hope it make sense.

Hi, Brian.
With your values, I get the expected results: a relatively similar waveform with the same frequency, and with a phase/delay offset typical of the digital audio process.
Try the following: with the reconstruction filter applied, progressively change the filter value upwards from the initial value of 250 (sampling frequency / 2). You can see the inevitability of the phase shift. The digital values can only be rounded/smoothed by a D/A converter after they are received, or, in other words, they can't be predicted, hence the offset between the original waveform and the resulting one. Hope it make sense.


teaching material for sample-rate and bit explanation