David Nahmani

A sound that is more complex than a sine wave can be deconstructed in the sum of multiple harmonics that are sine wave. If the complex sound has harmonics above 20 kHz and you sample the sound with a sample rate of 40 kHz then you'll record all harmonics up to 20 kHz and won't record any of the harmonics that are above 20 kHz. However, since you can't hear them anyway, you don't need to record them! Think of every sound as a sum of pure sine waves. You can't hear any of the sine waves that are above 20 kHz, so if you filter them out, it won't make any difference to you. Take any record, and run it through an EQ that filters out everything above 20 kHz, and you won't hear any difference at all. So the idea behind the Nyquist-Shannon sampling theorem is that if you need to be able to record all kinds of sine waves (in order to reproduce any sound, no matter how complex) up to a certain frequency, you need to sample at twice that frequency.

If you're trying to record a 20 kHz sine wave with a 40 kHz sample rate then you'll record only two alternative values, one high and one low. After reconstructing an analog signal with the DAC, you'll get a square wave, however when you filter off the high frequencies you're back to the original sine wave.

You're welcome! 🙂

I've added it. 🙂

In the Mixer, choose Options > I/O Labels. Press Command + A to select all the labels, and click one of the radioboxes in the 'Channel' column.

Yes. Well...it's not me saying it, it's Nyquist and Shannon. And in your sentence, you can replace "that much better" with "any better at all". Check out this oscilloscope view of a square waveform (produced by a synth). It's only two values, but when I run it through a low-pass filter, it ends up being a sine wave: Before I cut off the high frequencies, you can see all the higher frequency harmonics in the EQ Analyzer, that are responsible for those angled edges in the waveform, that give you a square wave. Now imagine that those are above 20 kHz, meaning you can't hear them. Once you run that reconstructed signal into a low pass filter at the right frequency, you get a sine wave at 20 kHz, the limit of human hearing. Meaning you got rid of anything unnecessary and end up with the same smooth waveform you had originally recorded.

You're talking about the Nyquist–Shannon sampling theorem Yes. However with sound, keep in mind that you're going to route the digital audio signal through a low-pass filter, which will reconstruct the signal in its smooth form. You need limited information about the value of the audio signal certain positions in order to recreate the smooth analog signal. For example in the image below, in the DAC, the reconstructed squared signal fed to a low-pass filter will result in the smooth sine wave like the original signal.

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The Tuner plug-in should work the same way as any other plug-in? Are the guitar players on different channel strips? Are you using software monitoring in Logic Pro?

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Ok that behavior has changed, because now you can have drummer regions on a MIDI track. That's a new feature. You can still Control + click the drummer region and choose Convert > Convert to MIDI region.

You can still Option + drag to copy a region of any kind in Logic Pro 10.8.1.

Click the tiny down arrow symbol at the very right of the LCD display, and choose Beats & Projects.

Thank you for that. That definitely shows IMO a bug in 10.8.1.

Yes, definitely agree, try to never have 2 projects open at the same time in Logic Pro.