384kHz/192kHz Audio Foolness

[scottjoplin]

The programming behind it is beyond me, I suspect. I think I'm happier messing with the bits of cardboard and magnets at the other end of the signal chain.

Chris

I'm with you there Chris 🙂 I tend to think the source and amplifier technology we have now is near enough perfect and the interesting bit is how you get your speaker, room and ears up to the job in hand, they have by far the largest role to play in the creation of what can be a wonderful image... 😀

 

[DF96]

Is it possible to take into account the phase response of the anti-aliasing filter, when designing an anti-imaging filter?

I am not sure. Causality can be a constraint on what is possible with filters, although some digital filters have a way of sidestepping constraints.

2 - It does look suspiciously like a beat frequency, which would be 100Hz. So the image filter looks at what's sampled, compares it to the sample rate, figures out what's an image (ie, what's above the Nyquist limit?) and what's actual signal, and filters accordingly. Neat.

Not quite. The image filter does not know anything about sampling, it just removes everything above 22.05kHz.

So Nyquist was right, so long as you've got the ability to do some checks on the sampled signal. With no filtering on the input or output, you would need a very high sample rate but the filters we can apply mean the extra bandwidth isn't necessary, so you can save space on your harddrive.

You need a bandwidth-limited signal, which means some sort of filter somewhere or an unusual source which always produces a bandlimited signal. High sampling rates can give the illusion of not needing filters.

 

[diyralf]

A filter is always needed, independent from sampling frequency.

 

[DF96]

A filter is always needed, independent from sampling frequency.

?

The filter specs are strongly related to the sampling frequency, so in what sense can 'strongly related' become "independent from"?

 

[TNT]

What you need is a sinX/X interpolator - which happens to be a LP filter 😉

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[lauda]

44kHz is the absolute minimum birate needed to describe a 22kHz frequency, so long as they're in-phase. If you get a 22kHz sine wave, phase shift it by 90 degrees and then take 44kHz samples, every sample will read zero since they coincide exactly with the zero points of the 22kHz sine wave.
Chris

What missing from the above is this: The Nyquist frequency defines the upper frequency bound for sampled systems but isn't inclusive. So, mathematically, a 22kHz signal sampled at 44kHz will give you a DC offset that's a function of the phase relationship between the signal and sampling, but a 21.99999kHz signal will come back just fine.