Converting Flac to Wav & Upconversion

Thanks DOB
No offense.
I'll check out Musica Pristina
It is a great hobby and trying to get the most from digital is quite an undertaking. We are pioneers. And there seems to be more than one way to get there.
Steven

Dob wrote:
"He will never guess, that "1" is actually electrical pulse, say 1 v amplitude for illustation and could be a) mispaced in time due to the jitter and b) greatly distorted (say reduced to 0.499v) due to noisy power supply, non linear DC:DC convertors etc and be mischaracterized as a "zero""

Well, this is certainly possible, although not very probable. Most of the effects in digital audio are not due to bit-errors, but rather from jitter.

This is why it is critically important to address the jitter of the source deviceand master clock as a higher priority than the format, computer, software or even the DAC quality. All of these things are second-order effects compared to jitter.

Steve N.
Empirical Audio

When FLAC and wav sound different, its probably more due to the different software in play for each and how that is written, designed, and performs more so than the format. Decompressing FLAC files will probably require more CPU processing, but should not be a problem if done right. Of course, things are not always done right, and many factors can come into play when playing digital music files, so differences in performance between the two in any particular case would not surprise me and reasons why may not be apparent.

FWIW, I think it's odd that different software players (that can be tested to show bitperfect output) can sound different playing back the same WAV or FLAC file. Heck, even different versions of the software can sound different playing back the same file.

It is thus not inconceivable that WAV and FLAC despite having bitperfect data can sound different. I remember that the Pure Music designer mentioned the need to minimize sudden/minute spikes in CPU load to improve performance and that was one of the goals of their software update. So it is not just a % of the CPU load that is averaged over time that we need to see but those sudden spikes. There was a talk at RMAF about 1-2 years back by an ESS engineer which talked about the need to look beyond steady states but also how the system reaches steady states (ie does it oscillate through large swings in values before reaching steady states). He found that large swings seemed to have a negative impact on the sound quality. I think there's a lot to the computer playback chain that we are only just beginning to understand.

For example, 96k files require twice as many mixing resources as 48k, which means that only half the number of channels on the console would be available.

Mr.Jazz1959,I do not know a lot about recording ,could you please explain to me how recording at 96k will use twice as many of available channels on the recording console as recording at 48k.Thank you