Okay, so it was inaccurately referred to as the Shannon-Nyquist theorem rather than the correct Nyquist Interval (or Nyquist rate). Big deal. Aside from the incorrect reference, the info is sound and serves the presentation's purpose of explaining how critical sampling intervals are integral to the audio data process.
Brad
No, actually I was wrong about the name. The video was actually just totally wrong about everything except the basic description of how PCM audio is encoded. All the electrical engineering parts and perceptual audio parts were beyond wrong. I wish he had posted a description of delta-sigma encoding to go along with it, that would have been cool.
Parts that were wrong: everything basically, but where to start...
The idea that our ability to hear up to a certain frequency is directly related to determining the ideal or acceptable audio sampling rate, in the first place, but also that the Shannon interval has anything to do with it if it were. That very idea is not-even-wrong and you can safely ignore anyone who starts out with that argument. In reality, a playback system needs to produce frequencies up to no more than 20kHz, this much is true, but that has very little to do with determining the sampling rate for recording or storage or processing of the digital audio signal for many reasons, not least of which leads directly to...
Considering sampling rate and bit depth independently, when actually they are completely dependent on each other when it comes to signal sampling and reproduction. So you can't talk about how much bit depth or sampling rate is "enough" by itself. In practice most consumer DACs (and probably most DACs period) are 1-bit DACs running at several MHz. That's because 1 bit is enough, if you have enough sampling rate. But 400kHz is NOT enough frequency at 1 bit, even though 400kHz is 20 times the "nyquist" rate, hopefully illustrating how wrong the idea is of basing the sampling rate on the frequency to be played back using some nyquist interval argument. In reality, this is not a valid application of the nyquist rate, and the supposed nyquist rate may be drastic overkill, or drastic underkill, or anywhere in between, based on other components of the system. Phillips engineers chose 44.1kHz just because they repurposed cheap video hardware.
The idea that the bandwidth of a signal has some relationship to the maximum frequency of the signal--wrong. Radio literally wouldn't work if this were true. Also, this is approximately why mp3-type compression works...the information bandwidth of a signal has only an indirect relationship to the frequencies contained in the signal. That's how come mp3's can be 1/10 the file size while sounding the same.
That the maximum frequency of audio is 20kHz and frequencies above that don't matter because they are ultrasound--totally wrong on an engineering level. There are sound synthesizers that work entirely on ultrasound sources. All of the audible output consists of mixtures of the ultrasound sources. Filtering out ultrasound in this case would result in total silence! Oops.
Also he is FOS about being able to tell that FLAC sounds better. Mp3 and other lossy codecs are tested for audio transparency, and have healthy margins. Most people really can't tell 64kbps...really they can't, they do blind testing all the time. But the standard is 256kbps just for gratuitous overkill, and you can always go higher if you want even more overkill, while still saving a ton of storage space.
The author thinks you are an "audio fool" for using 24bit PCM because "nobody can hear the difference" (whatever that means) but in the same breath he thinks you are smart for encoding your music collection in FLAC and dragging around 590% more data that you literally cannot hear, and we are supposed to believe both things at the same time.