Any one doing music at 48 then SCR or recomverting to 44.1 after mixdown?

[Marc Edwards]

s.d. finley : Eh? -- there are a million reasons why 44.1 sounds worse than 48/88/96k ... check out some of the other posts on DUC!!

Think about the way a full volume 22.1k sine wave is represented in 1s and 0s.
sample 1 -- waveform is all the way to the top
sample 2 -- waveform is all the way to the bottom
sample 3 -- waveform is all the way to the top
sample 4 -- waveform is all the way to the bottom
_ _
| | | |
|_| |_|
1 2 3 4

Looks more like a square wave to me (ie. full distortion, not really a smooth sine). The only thing saving this is oversampling on DAs. So, 44k can't really record a 22k signal without screwing it.

FYI: I think the guy who worked this out went by the name of Nyquest or something (hence the name of the Nyquest frequency - half of the sample freq).

[Jules]

I'le tell you, after starting this thread and another couple that Roger Nichols and George Massenber chipped in on elsewhere, I AM NOT DECIDED ON HOW TO PROCEDE!.
I think the best thing ist to run @ 48k and buy an Ampex ATR 102 1/2 inch !!! (oh yes and a Neve 12 ch 'sidecar' desk to a) track through, then b) to sum mix stems from PT on before the ATR!)

Jules,

[Tertsipede]

<BLOCKQUOTE><font size="1" face="Arial,Helvetica,sans-serif">quote:<HR>Originally posted by Marc Edwards:

Think about the way a full volume 22.1k sine wave is represented in 1s and 0s.
sample 1 -- waveform is all the way to the top
sample 2 -- waveform is all the way to the bottom
sample 3 -- waveform is all the way to the top
sample 4 -- waveform is all the way to the bottom
_ _
| | | |
|_| |_|
1 2 3 4

Looks more like a square wave to me (ie. full distortion, not really a smooth sine). The only thing saving this is oversampling on DAs. So, 44k can't really record a 22k signal without screwing it.

FYI: I think the guy who worked this out went by the name of Nyquest or something (hence the name of the Nyquest frequency - half of the sample freq).[/B]<HR></BLOCKQUOTE>

Mark,
You may be forgetting that the A/D & D/A -conversion process consists of more than the components you described: there is also the lowpass filtering before the A/D, and especially in this case, the lowpass filtering AFTER the D/A. Hence the square wave should turn into a perfect sine wave at 22.1 kHz, because the edges round off in 22.1 kHz LPF.

BTW, another scenario of sampling a 22.05 kHz sine wave at 44.1 kHz rate could lead to sampling only the 0-crossing moments of the sine, so you would end up with silence. That´s why the samling frequency always has to be MORE THAN twice that of the upper end of the frequency band. (Yes, the Nyquist-theorem...)

This topic has been in several discussion forums and it will always show up again, since we all seem to "know" WHAT DIGITAL AUDIO REALLY IS.

I´m not saying that we all should have a master´s degree in digital signal processing, before sharing our thoughts here (although it really helps to understand the media!)

Reading books like Principles of Digital Audio can help us all in understanding what we´re dealing with.

yours, Tertsipede

[This message has been edited by Tertsipede (edited May 05, 2000).]