You are misconstruing Monty's argument here. He is very much against mp3...in fact he says he could tell the difference between high bitrate mp3 and 16 bit 44k wav. The real point of the video is that 16 bit 44k wav is beyond sufficient...don't need to go beyond that to 24-bit 192kHz.
In the early days, when all mp3 encoders were pretty bad, I could tell which encoder produced a given file. mp3 encoders today are vastly better. I've not been able to do that party trick in a long time.
But 256+ and I certainly cannot tell the difference reliably.
http://abx.digitalfeed.net/list.html
Pretty much found the same thing for myself at 256+.
CD quality is the very least you could want for a serious big club or theater system (much less auditorium). Between peaks and the requirements for deep bass, the peak in a digital audio file is (a)much farther above the body of music than you'd think, and (b) should never be reached, because that's clipping.
People routinely behave as if the theoretical maximum dynamic range of a Red Book CD is relevant to anything. It's incredibly easy to play back music loud enough that the noise floor will get obnoxious and relevant to listening, it's only 96 db down. Any small system in an enclosed live space can push louder than that. Cranking music over headphones will blow way past that and you won't even hear the peaks, but you'll be able to hear how bad the subtleties (or lack of same) on 16 bit red book CD are.
Electronic music, especially live, is totally relevant to high resolution audio. I more than suspect some of the big name acts (for instance, Deadmau5) are using converters to the mains running at not less than 24/96. Certain synthesizer mixes are very revealing of faults in the playback. If the live performance over a modern PA sounds absolutely huge and awesome, but not strained or grainy, then they're not using CD quality. The SPLs are more than enough to make these distinctions obvious.
Anyone can get these SPLs over headphones, trivially, and headphones that'll handle it cleanly are only a few hundred dollars.
(Personally I downloaded some of those killer samples and couldn't tell the difference, but other people reliably tell them apart in an ABX test.)
I ABXed 320kbit mp3 from an uncompressed original, I think 9/10 IIRC. It was a recording of castanets, and listening for frequency response differences was useless so I keyed off of 'personality' differences in the sounds and did it that way.
I was also just as horrible at detecting wow and flutter, as I was good at detecting lossy compression 'changes of sonic personality'. Comes from my experience being with analog audio, which is less true of people these days.
The idea that 'the best' listeners cannot tell 256k from even something as limited as 16/44.1 is ridiculous. Any mp3 format… any lossy format… is seriously compromised. Talk to techno/house DJs and producers about how useful it is to perform off mp3s, this is not a hypothetical argument.
Usually what is happening here is you're getting a master that hasn't been compressed to death (see loudness wars). Vinyl is a shit source for 'quality', but most records aren't compressed to death so they can still sound better on a good set of speakers due to the dynamic range.
* With analogue equipment the latency between the input and output of the effect is often below 1 ms (unless the effect is supposed to have a delay).
* Standalone digital effect equipment that processes the samples one at a time can also have a latency below 1 ms. (48kHz ≈ 0.02 ms)
* If you use a computer for real-time effects, the samples are not transferred sample by sample from the audio interface, but in a block of many samples. The number of buffers and the size of them can usually be changed by the user. With a buffer of 1024 samples, the oldest sample is already about 21 ms old before the computer can process it. After the buffer is processed it has to be transferred to the audio interface, and that will add another 21 ms. So the minimum latency for any real-time effect in the computer is about 42 ms at 48 kHz if the size of the buffers is 1024 samples. Often it is much worse because the operating system adds more latency. If the equipment can handle a sample rate of 192 kHz, the same latency is about 10 ms. If the computer can handle smaller buffers, the latency can be lowered. With 256 samples per buffer the minimum latency will be about 11 ms at 48 kHz and 3 ms at 192 kHz.