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48k/24 up to 96k/24 down to 48k/24 - do we end up where we started?

fatoldgit

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Simplistically, in theory (as far as I can tell)

- we aren't changing bit depth so no dither
- the 48k to 96k just replicates each sample (n*2)
- the 96k to 48k just removes every second sample (n/2)

So do we end up with the original 48K and the resultant 48k audio being bit perfect/identical?

Or is there some hidden processing that will ensure this doesn't happen?

Note the "stuff" doing the upsample/downsample is hardware not software so I have no control over what happens.

Thanks,

Peter
 
No. There is 'imperfect" interpolation and filtering. You can't just throw-away every-other sample because if you don't low-pass filter when down-sampling you can get aliasing (false frequencies).

All proper down-sampling algorithms include anti-alias filters and in general resampling algorithms are more complex than you might think.

But this is not considered to be a lossy process and as long as you stay above "CD quality" it's very unlikely to make an audible difference.

It's good practice to avoid unnecessary re-sampling but sometimes you can't avoid it, or for some reason it may make things easier or more convenient.

You wouldn't get aliasing with the simple process you propose but you aren't going to run-across that.

Note the "stuff" doing the upsample/downsample is hardware not software so I have no control over what happens.
I wouldn't lose sleep over it. ;)
 
Thanks for this.

As an insomniac with very little sleep to lose... I think I will go to plan B which is to skip the upsample so the end point gets the 48k input it natively desires.

I look at it this way... control what you can control and don't lose sleep over stuff you can't!!!

Peter
 
Since my AVR (and most others) resample to 24/48 for dsp room correction and most hdmi video streamers output 24/48, we should just make that the standard and call it a day. If 16/44.1 is perfect for audio, 24/48 ought to be even more perfect.
 
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