ralphpnj wrote: 
> The MQA process starts off by making an invalid assumption and then
> builds up from that falsehood. An analog audio signal (which is simply a
> type of electrical signal) is converted to a digital audio signal by an
> ADC (analog to digital converter) and then the resulting digital audio
> file is send to a DAC (digital to analog converter) where the digital
> signal is converted back to an analog audio signal. The present day ADC
> / DACconversion chain already produces an analog audio signal that is
> exactly the same as the original unconverted analog audio signal. So
> where is all this improvement happening? Please explain.
I didn't design the process so I am not in a position to explain what it
does or why it does it. I am referring to the concept only of a
calibrated record and playback chain.

I am however aware that there is no ideal piece of electronics on the
planet and therefore I don't think I agree that the analogue audio
signal is perfectly retrieved in all cases. Good though most systems
are, there are differences and maybe a branded certification process is
what's required to pull the music industry into shape, regardless of the
merits of the maths this particular certification process includes.

Personally I'm of the opinion that 16/44.1 is good enough, in general. I
don't have a problem with people mastering to 24/192 if they can,
however, as the data rates are still small enough to be pretty portable
and in theory the extra bits gives you more latitude in mastering.




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