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Originally Posted by badmonkey 
Because otherwise noobs are going to think it is actually worth doing this, and your particular brand of idiocy is likely to reoccur, and surface along with others writing "guides" such as those instructing everybody with a PC + Foobar to resample their music to 48KHz without exception.
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I asked why studios do it, and if theres an advantage, nowhere did I tell anyone to resample to 48khz, thats a separate discussion about the hardware I've got, and I'm not saying it improves quality
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Originally Posted by badmonkey
It appears to be that time of the month for you hmmmn but let's just reiterate: a SPDIF in from some other device vs digital playback from a modern CD-ROM drive will either 1) make no difference, assuming you avoid any problems or 2) screw something up. It is NOT better and doing it without some good reason is stupid.
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so because you don't like someone else's system, or their method, that makes them stupid? thats the most solid logic i've ever heard in my life, in fact, I might as well stop attending university, stop reading entirely, and just subscribe to your logic...
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Originally Posted by badmonkey
You keep mentioning DACs, you appear to have no idea that this is a totally separate issue to your query. In either case you apparently are currently using the DAC on your soundcard, sourced either from the SDDIF-in or the OS audio layer. I don't understand the continual referencing of "a $300-$30,000 DAC".
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because you want to sit there and fling random comments that I should burn my soundcard and get a "real DAC" just to "actually be able to listen to CDs", as far as how its connected, I'm pretty sure we've covered that a few times in this thread, you might wanna invest in some better glasses

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Originally Posted by badmonkey
Instead of mucking about with this, you should be giving some attention and $ to upgrading your soundcard which is apparently the weak part of your system. Look at the Auzentech Prelude, which is an excellent card and will maintain the Creative X-Fi features of the Audigy.
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um, why is it the weak part? its got better SNR and THD figures than mostly anything on the market, excepting a few semi-pro boards, as far as "the x-fi features of the audigy", do you even know what the X-Fi and Audigy actually are? or do you just love reading marketing literature and believing that the X-Fi Prelude is some saviour of the audio world, in reality, its the second best X-Fi based card, and the 4th best Creative based card (arguably 4th, you could say 3rd if you like, and I wouldn't think less of you)
so again, why is it the weak part? because it isn't what you have? because you don't like it? because its not "perfect"?
now, as far as reading it as data blocks, yes, thats been discussed, and yes, it creates a more accurate read, because of error checking, and the ability to read the digital data as data, nowhere am I saying that my method is actually better, I'm saying it sounds better to me (and if you're seriously incapable of reading and accepting that, just stop posting, because like I've said three times now, I couldn't care less how it sounds to you, its my system and my ears)
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Originally Posted by infinitesymphony
Everything I've read has stated that the Live! and Audigy line resample all rates below 48 kHz to 48 kHz, with the exception of the semi-rare Audigy 2 Platinum eX. The Audigy 2 ZS resamples.
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i'm curious (not trying to fight with you) on your sources, since I've read a bit of the opposite, first of all, lets not bring Live! into this, since its a different generation and different hardware (lol), but I've read that Audigy 1 resamples everything, and Audigy 2 and 2 ZS (the whole product line, since they're all the same chip (actually the Live! and all of E-MU's current PCI cards are also that same chip, only deliniated by revision (rev 1 is Live!, rev 2 is everything else)) don't resample, unless you enable EAX, EQ, effects, etc (if you bypass all of that (which imho sounds like garbage half the time anyways) you get proper 44,1000 or 96,000 or 88,200, etc, the same is also true if you're doing ASIO in/out, it just requires bypassing the effects toys)
and yeah, I'm with you on it being strange in a studio, thats why I made the thread, just figured maybe some studio people knew something I didn't, apparently the playback sounds nicer (as my ears are telling me), but it also apparently pisses people off because it isn't identical to what they're doing