xiphmont’s ‘There is no point to distributing music in 24 bit/192 kHz’, Article: “24/192 Music Downloads are Very Silly Indeed” Options

[NullC]

Of interest, but certainly not news to anyone here is Monty's educational piece regarding the lack of value of 24/192 as a distribution format:

http://people.xiph.org/~xiphmont/demo/neil-young.html

Some of the commentary on the internet has been not especially well informed, for example there are some crazy claims at https://news.ycombinator.com/item?id=3668310. If you're in a "someone is wrong on the internet!" correcting mood, you might want to go leave some comments in furtherance of the collective intelligence of mankind.

Cheers

[Patrunjica]

True, but each time they repeat they are different then before, dividing up the sampling rate to that particular frequency results in a number with an infinite number of decimal places, approximating the location of one of the sampling points will influence the position of the next sample point and so on and so forth so that each time the wave repeats it shifts out of phase relative the sample points. And since the respective frequency is closer to the Nyquist any such deviation is more noticeable compared to a deviation that might occur using a higher sampling rate.

I know the waveform can be rebuild since obviously I can hear it, but I can't help but doubt that it's a more of an imperfect reconstruction then compared to pretty much anything lower then a 11025Hz sine.

[splice]

... I know the waveform can be rebuild since obviously I can hear it, but I can't help but doubt that it's a more of an imperfect reconstruction then compared to pretty much anything lower then a 11025Hz sine.

Play the waveform back through a digital to analog converter. Look at the resulting waveform with an oscilloscope. It's a smooth sine wave. Magic.

The magic trick is that the D to A converter doesn't just "join the dots". It passes its output through a reconstruction filter that only passes frequencies below half the sampling rate. Say you digitise a 20 KHz signal at a 44.1 KHz sampling rate. That 20 KHz signal can only ever be a sine wave. For it to be any other shape it would have to contain harmonics, and those harmonics would start at 40 KHz. The harmonics would be filtered out at the input to the A to D converter.

When you think about it, any digitised signal above 11.025 KHz must be a sine wave. It may itself be a harmonic of a lower frequency signal, but it won't in turn have any higher harmonics because they would be above 22.05 KHz.

So if you take your "jagged" join-the-dots line and join the dots with a sine wave curve, you'll find that there's exactly one curve that will join all the dots: one with the same frequency as the original frequency. Drawing that curve is exactly what the D to A reconstruction filter does.

This post has been edited by splice: Oct 14 2012, 01:45