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Originally Posted by Gnus /img/forum/go_quote.gif
It's been awhile since I've fiddled with the X-Fi configurations, where can I find the option to change this? In foobar, mine is locked to 32-bit.
Also, someone mentioned selecting Audio Creation mode -> bit-matched playback will essentially do the same thing as manually changing it to 16-bit, is that correct?
Keep up the good work!
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In bit matched mode, the card will read out exactly what is in the file, at the bitrate the file indicates. Yes, getting bit-perfect output from the X-fi is that easy.
Enoyin: I don't mean to offend either, or imply that anything written is incorrect. The only thing I would want to steer people away from is the incorrect assumption that if they are using kernel streaming or ASIO to output bit-perfect information to a sound card, then the sound card will automatically be outputing analogue or digital based on that bit-perfect information, unaltered.
Some soundcards and most audiophile-grade DAC's will do so; Others, such as just about everything Creative produced prior to X-fi, will internally resample to 48 khz, no matter what, and there's nothing the user can do to bypass it. As far as USB consumer-grade soundcards, I know the M-audio Transit can pass bit-perfect 44.1 khz information from a CD to, say, an outboard reciever or dac; I also know the Creative Audigy 2 NX
cannot, at least not using ASIO, ASIO wrappers, or Kernel Streaming (It resamples everything to 48khz internally, regardless of whether it's talking to k-mixer or not). Regardless of what the NX ouputs over the optical/digital coax, it has at some point been resampled, in some cases twice (44.1-->48-->44.1), internal to the card. In the NX's case, whether or not its resampling has any impact on sound quality is something I don't know.
When I talk about the sampling rates not being integer multiples of each other, I mean that to go between two such rates requires precision math and can be computationally intensive. It has nothing to do with failing the nyquist rate, it has to do with the fact that if the two sampling rates don't "match up" nicely, then you have to sit and figure out what's the best way to render the new samples in the final rate based on where they fall in time in relation to the samples in the old rate. Sample rate conversion is entirely the reason to bypass k-mixer, I don't see why one would do all that work to bypass it, only to hand the info off to a soundcard that's going to do the same thing anyway...and I think it would be remiss to suggest that once you bypass k-mixer, all is well with regards to getting your original digital stream out un-molested.
