Orinoco_Womble said:

... Some people have mentioned that the 2-ch audio sample rate for PAL is 44.1KHz (same as CD), but NTSC is 44056Hz. I'm pretty sure at best 'cheapie' audio cards with optical SPDIF in will only manage 44.1KHz. Any idea if this is true? bob23 mentions it here and Buster D mentions it here. It's also mentioned in the LD FAQ here, but others have said it's a load of poo. Dnewhous thinks so according to this post here. Who's right?

The 44056Hz rate was also mentioned here; I'm leaning towards the "load of old poo" answer.

Also, you mentioned in this thread about having to up-sample the 2-ch 44.1KHz audio to 48KHz for DVD player compatibility. If this is the case, then why not just capture at 48KHz and have the audio h/w do the re-sampling for you? Or is this one of those cases where the cheap HW will give audio artefacts and it's more reliable (pure?) to use s/w for the up-sample?

I was under the impression that some audio cards could not resample a digital input in realtime. I could be wrong, I haven't captured many PCM streams to my PC. If the card can do it then fine - but I would still compare the results against a software upsample.

Lastly, Darth Editious mentions having to strip out padding from the captured AC3 raw audio stream here. Has anyone else come across this and can suggest an easy method to remove the padding please?

I've also read this in several places - if you zoom in on the waveform of a captured AC3 in an audio editor, you can apparently see "packets" of data with "silence" in between. Something like AC3Fix or BeSliced should be able to extract the real data.

PS: Not that I really need to capture LD - DTS digital audio streams, but has anyone figured that out? I can't seem to find anything much on that at all.

I've heard that DTS is actually easier - no demodulator required, for a start.