what did I do wrong to lock up Audacity with SPDIF in?

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- was checking a M-Audio 2496 SPDIF-IN functionality with a dvd player coax out and a commercial cd - recording levels looked fine - I had not figured a way to monitor - - - upon playback, Audacity barely ran - even after rebooting, then Audacity would not play this file nor earlier files. Was it a type of copy correction? - I'm trying to get it setup for ripping lps with an A/D that relies upon SPDIF out.
 
What are you using for a sound server? (If you're using Windows or a Mac, this may seem like a strange question, so "I don't know" is a good answer if you can mention your OS and version.)

I'm running OpenSUSE primarily because they still maintain KDE 3.x, and until recently it worked fine. But it seems like the old aRts sound server isn't playing nice with the latest versions of ALSA. Even after I ripped out KDE3, I'm still having problems with certain programs like Audacity and VLC, and the symptoms seem to be a lot like what you describe.

The M-Audio 2496 supports LPCM and AC-3 over S/PDIF, so if you're using a DVD player set to AC-3 or Dolby output, you might try switching to plain stereo. Just a wild guess at this point; any further info would help.
 
found the problem back on that day - the sampling rate was unchecked - I've a question re: the 2496 card - what is its analog input sensitivity for 0db?- 36-40dB gain phono stages seem to come up perhaps 10db-14db short of hitting near 0db with a cheap AT moving magnet cartridge. Is this a configuration problem or does the card need a strong signal? at some point I'll be trying a PCM422 AD to digitize old lps and think will be short on gain there too.
 
The manual suggests that there are two different settings. one for "Consumer setting", and the other presumably for "-10dBV setting". I'm confused by that, since -10dBV is the nominal consumer level! The terminology seems pretty dodgy, so who knows? Ever since the IHF dissolved, audio specs have been pretty much up for grabs. I'd interpret to mean that 1V p-p is the max, allowing for 2-4dB of headroom/error.

On one hand you'll want enough gain to get at least the ideal S/N ratio for digital capture, but on the other you don't want to run the risk of hard clipping, or literally running out of bits. If you're used to making analog recordings and keeping your gain as close to the red line as you dare, recording to digital takes some getting used to.

Even a good record and phono preamp isn't likely to give you much more than 60-70dB of dynamic range. So being down 20dB or more can be a Good Thing, as you're not going to gain anything by using more more significant bits, but can ruin the entire recording by exceeding the redline once.

I've only used my 2496 for its digital capabilities, so I'm absolutely not the utmost authority on this card. Good luck though!
 
one of my turntables sounds like a slow freight train for rumble😀😱
That's what I was talking about. If you crank up the gain to get every most significant bit that you can muster, all it does is pull stuff like turntable rumble and phono preamp noise into the foreground. If you let the whole dynamic range of your turntable/preamp slide down into the middle of the overall dynamic range that you get digitally, the rumble is more "down in the weeds", but still above the level of decimation.

In analog recording, the custom is to record as "hot" as possible to get the best S/N ratio. With digital recording (you're making a digital recording of your records being played), it's almost impossible to have too much headroom, so some find it preferable to put their analog noise floor just above the digital noise floor, and optimize for headroom. The ideal analog to digital transfer would have the analog envelope squarely within the digital envelope.

I hope that makes sense. I'm far from a master at this, and I'm only repeating what I've been told to do. Again, good luck, and I hope they come out sounding great!
 
found the problem back on that day - the sampling rate was unchecked

It should be set to external spdif clock (or similar wording) if you want to record from spdif input. Otherwise the reception does not have to be correct as there are two clock domains clashing - the one in incoming SPDIF and the other one of your soundcard internal clock.
 
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