Coen, not to debate it, but to give some counterweight to people who might like to pick up something : still not true.
I chewed a little on my post over dishwashing and you are right (as allways
)! What I posted is not true and contains a flaw and you will end up with aliasing into the target bandwidth.
My post only makes sense if you filtered out all frequencies above the target nyquist freqency in advance. Only then you will (at least theoretically) end up with samplesets that all represent the same waveform (with synchrounous downsampling).
My synchronous 'experiment':
So take a 176.4 kHz original (88.4 kHz audio bandwidth). Filter it to 22.05kHz audio bandwidth digitally with a brick wall filter and then you will end up with four sets of 44.1 kHz samples all representing the same waveform. So no new samples on the time axis, but recalculated sample amplitudes.
So no way to downsample without filtering it first. This is the basis of Shannon!
The adc chips contain many of these filters. You will have to down sample from 24 MHz or something to 192k...
I'd say : try Linear Interpolation and hook up an analyser (if you don't hear it in the first place). That's just injecting samples like you describe ...
(I should remove that stupid option
)
Well this is not what I had in mind of course. With asynchronous conversion you also change the timebase of the samples (only once every x samples you end up with a sample timed on the original timebase). The filter will determine the amplitude of all samples, so no one in his right mind will use Lin Interp for that.
I have allways wondered what the above synchronous experiment will result in. Do the four 44.1kHz sub files all sound indentical after proper filtering of the 176.4kHz 'motherfile', like they theoretically should?
This would be quite illuminating. Actually my idea is that they will NOT...
Back to the sq of hirez material.
Theoretically the hires in native samplerate contains much more original information than the downsampled 44.1 red book variant. You will have to filter and reduce bits by some inherently lossy process to get from the master to red book format whatever you record with these days. I don't buy 'native' 44.1k.
In practice you seem to have ameliorated this loss to something insignificant, like we experience with xx and the NOS1.
Thanks for that!
Regards, Coen