Audigy 2 NX recording sample rate question

Hi, I have two questions about the sample rate used when recording in the Audigy 2 NX -
When recording an analog signal (Line In), is the actual?sampling rate at which the analog signal is being sampled different from the requested sample rate ? For example, when recording at XX Khz, does the actual sampling of the analog signal really?happen at XX Khz or does it get sampled at a different fixed rate(48khz?) and then gets re-sampled to XX Khz ?
When recording a digital signal (spdif in), if the recording sampling rate specified is the same as the sampling rate of the input signal then does the card record this digital input signal in a bit-perfect fashion ? For example, if the digital input signal is 6bit and 44.Khz, and the recording is also being done at 6bit and 44.Khz then does the card actually record the signal at exactly that rate without modifying the data in any way ? or does it record it at a different rate and then re-sample internally to arri've at the requested recording rate ?
Now I know this card works a little different in terms of re-sampling when the 'Digital Input only' checkbox is checked. So I am looking for the answers to these questions keeping in my mind this option as well.
Thanks!
Message Edited by whatcreative on 02-8-2009 08: PMMessage Edited by whatcreative on 02-8-2009 08: PM

thanks for your replies.
katman,
I checked that thread. Although I wonder if the same thing applies to recording at rates higher than 48khz as well ?as I am more interested in finding out if when recording 96khz does the signal really get sampled at true 96khz or something else ? But, Upsampling a recording (not playback)?to a higher frequency doesn't make sense?because you can't create what you lost by recording at a?lower sampling rate. A test for checking this would be to try and record a signal that has frequency content well above 24khz but less than 48khz. If that content gets captured that means it is indeed recording at 96khz, if it gets lost then its recording at 48khz.
flipflop,
I have opened up the soundcard and messed around with a lot. Here's a list of the chips alongwith their descriptions based on whatever I could find?about them -
Sigmatel STAC9460S - Codec
JRC 4556A - Headphone output opamp
ST MC33078 - Line level output/input dual opamp. (Total 4).
Philips UDA334 - Low Power DAC
Philips ISP58BD - USB device controller
Creative CA086 - DSP
Cirrus Logic EP7309 - High-Performance, Low-Power System-on-Chip Enhanced Digital Audio Interface
ST M29W400BB - 4Mbit Flash Memory
ST LD33 - 3.3v regulator (?)
ST LD8 - .8v regulator (?)
Although the datasheet of the codec suggests that it supports all sampling rates, I'd have to dig deeper to really find out if it is handicapped to only one standard rate for recording - http://www.alsa-project.org/~james/d...tac9460-ds.pdf
Message Edited by whatcreative on 02-2-2009 :35 AM

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Doogs wrote:
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dgoh88 wrote:
okay just got home, turned the PC back on and now all the sounds are distorted again. Tried completely disabling wireless ports, still the same. About to throw in the towl already
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