So I appreciate everyone's point of view and applaud using an empirical approach, for those of you who share the author's point of view, but I disagree unfortunately. For those of us who have worked in DSP, either using it or implementing new things with it, there's s highly mathematical reason to record the source of you audio with a higher sample rate than what the author suggests is a generous maximum.
It has to do with waveforms and how continuous they are. So, for starters, true, if you have a perfectly continuous wave form, at 22K, then your sample rate must be at least 44K. In fact, with sample rate of 44k you can perfectly discretize a continuous wave form, like a sine wave.
Does you see the problem with this? Sounds are not always continuous! If you look at the waveform of a violin, distorted guitar, cymbal, etc... They're very jagged. To effectively approximate these analog waveforms as a finite set of sums you need a much higher sample rate. It makes s HUGE difference, trust me.
So basically, technically speaking, 44K works just five if you only listen to music made by orjan pipes and penny whistles, but most sounds are very complicated, and to be properly captured you actually need a higher sample rate. It's simple and mathematical. Also, this whole "44.1K is all you need and if you don't agree with me then you're dumb and don't understand math" ra ra ra has been going on all over the Internet for ages, and while u appreciate the motivations that people may have, it gets a little annoying. Basically, instead of immediately jumping to the conclusion that people's ears are wrong, maybe the more patient and mindful approach is to ask oneself, "why does my mathematical knowledge of a subject fall short of explaining what many people seem to experience?".
Note: everything I said was regarding the source of capturing a sound. There's an entire science behind compression and all that sauce.
Also Stanford's DSP lectures (available online) explain this much more indepth, albeit abstractly.
As a former recording engineer, I completely agree that there's value in capturing audio at better than 16/44.1. 24-bit means I don't have to care so much about "filling up the bit bucket" when I set input levels because I have a lower noise floor. And if I'm doing any DSP, obviously I like having more information rather than less.
But I'm not at all convinced that distributing recordings at better than 16/44.1 has any real benefit. I've done some blind tests before and was never able to reliably beat 50/50 on figuring out which tracks were "higher quality" - and while I certainly don't have the best ears on the planet, I feel pretty confident that my hearing is more developed than the average person. Not to mention the fact that probably 90%+ of consumers are listening on systems with poor speakers and worse DACs.
I often hear audio people explain why music should be distributed at higher bit and sampling rates, but I have yet to see anyone who can reliably tell the difference - especially on a consumer-grade system.
Edit: Downvotes, really? Was there something objectionable in there?
I'm sorry, but if you have a DSP background you should at least be able to give a better explanation than 'trust me'. I have a really tiny DSP background (a 3 week lecture course), but specific questions that would help convince me of your point of view would be:
- Sampling at a frequency 2f lets us reproduce the specific component of every frequency <= f (proved by Shannon, Nyquist etc). This can be easily proven mathematically, and were it not the case in practice would have been largely discredited (it's a fundamental of many fields).These 'jagged' waveforms must then have their frequency components <= f perfectly represented - the only components that are ignored are those that are > f (I've idealised here, but I'm assuming we've filtered correctly). If f is greater than the maximum frequency we can hear, why does it matter at all?
- When the entire field of digital signal processing (and anyone who uses it, such as a sound engineer) relies on such math from top to bottom to make the recording process work, why should we ignore it as soon as we start experimenting and simply consider what our ears tell us? Why should you ask the question 'why does my mathematical knowledge of a subject fall short of explaining what many people seem to experience?' in the first place? It seems to imply that your ears are more accurate than a century of study, which at least seems arrogant.
- A number of the points you've made (regarding people hearing different things) were explained in some way in the article. What are your views on that?
I'm largely convinced that you're wrong (you've given no specific justification for any of your points), but I would be interested to hear what you have to say.
"everything I said was regarding the source of capturing a sound"
Well, the rant is about reproduction of the sound. For capturing sound he basically agrees with you, 24/192 has its place in studio, recording and music production.
The articles entire point is that statements like this cannot be trusted. Read under "Listening tests", and it's clear that there is no benefit to the higher samling rate. I highly encourage you to do your own double blind ABX - if you can select 10 tracks where people can reliabilly tell the 44K from the 192K I will reconsider. But until then, everything we know so far tells us that 44KHz is more than enough.
It has to do with waveforms and how continuous they are. So, for starters, true, if you have a perfectly continuous wave form, at 22K, then your sample rate must be at least 44K. In fact, with sample rate of 44k you can perfectly discretize a continuous wave form, like a sine wave.
Does you see the problem with this? Sounds are not always continuous! If you look at the waveform of a violin, distorted guitar, cymbal, etc... They're very jagged. To effectively approximate these analog waveforms as a finite set of sums you need a much higher sample rate. It makes s HUGE difference, trust me.
So basically, technically speaking, 44K works just five if you only listen to music made by orjan pipes and penny whistles, but most sounds are very complicated, and to be properly captured you actually need a higher sample rate. It's simple and mathematical. Also, this whole "44.1K is all you need and if you don't agree with me then you're dumb and don't understand math" ra ra ra has been going on all over the Internet for ages, and while u appreciate the motivations that people may have, it gets a little annoying. Basically, instead of immediately jumping to the conclusion that people's ears are wrong, maybe the more patient and mindful approach is to ask oneself, "why does my mathematical knowledge of a subject fall short of explaining what many people seem to experience?".
Note: everything I said was regarding the source of capturing a sound. There's an entire science behind compression and all that sauce.
Also Stanford's DSP lectures (available online) explain this much more indepth, albeit abstractly.