Digital Optical / AC3 RF de-mod audio cards?

Hi all,

I've been reading around trying to find which audio card is appropriate to use for LD captures.  The LD player I'm using is a CLD-D925 and I've also got the an external pioneer AC3 demodulator.  The D925 has digital optical output for the 2ch audio/DTS and the AC3 demodulator also has digital optical outputs.

Could someone please recommend a particular brand+model of audio card (pref PCI) that is up to the job? (hopefully not too expensive as I've already spent my LD budget this year).  This should allow me to capture both digital audio stream from the D925's optical output as well as he 5.1 optical from the AC3 demodulator.

Many thanks in advance.

Don't know if you checked Google's archive of alt.video.laserdisc from around 2003, but Karyudo's success in doing this was discussed around this time.

Here's one discussion:

Donald Haas said:

What Audio card allows the saving AC-3 right from the TOSLink datastream without decoding?

Karyudo said:

Ah, now there's a question! This is the hardest part. Because many
cards are TOSLink/coax capable, but then mangle the bitstream. Even
some cards with the same chipset do different things: one will be
fine, and another will butcher. You're gonna have to try some, I
guess. I've got an ISA card that works fine, but it's ISA. Some cards
-- even quite inexpensive ones -- with the CMI 83xx (<-- forget the
exact number) chipset are OK; others don't work.

This is one arena where throwing more money at the problem isn't the
fastest way to a solution. A cheap-o CMI-based card could well
out-perform (in this aspect, anyway) the most expensive Creative
Audigy something-or-other.

Best of luck -- and please post your results (good or bad), because
that's what everyone's looking for!

D. Carroll said

Sound cards that have codecs (coder/decoder) that conform to the AC97
spec will resample all audio to 48kHz.

Most "consumer" (read cheap) soundcards conform to the AC97 spec.
AFAIK, the only cheap soundcard that doesn't conform to the AC97 spec is
those based on the Cmedia 8738 chipset (thanks to its built in
proprietary codec). 

However, "Prosumer" (read expensive) cards based on VIA's (formerly
ICEnsemble's) Envy24 don't resample audio.  M-Audio makes some nice
cards.  Too bad their relatively cheap new consumer card based on the
Envy24, the Revolution, doesn't have any digital inputs.

I heard that Microsoft is working on an update to AC97 but knowing
Microsoft there will undoubtedly be some catch.

Karyudo has not been on here in years, but I know that Darth Editous and adywan have also managed to do this.

Thanks for the info Moth3r, I think that's pretty much exactly what I needed to find out.

Unfortunately though it looks like audio cards based on those two chips are no longer available new and are not currently for sale second hand on EBay either.  Bummer :(

However cards based on the newer CMedia CMI8768 might possibly work, so I might give one of those a try and see what happens.

On futher searching into this topic, I've found quite a few posts from people who mention all sorts of things which make this even more confusing.  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?

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?

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?

Many thanks again guys :)

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.

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.

Moth3r said:

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

Thanks for clarifying.  A load of old poo it is then :)

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.

Isnt that exactly the opposite of what D.Carroll was saying?

D. Carroll said 

Sound cards that have codecs (coder/decoder) that conform to the AC97
spec will resample all audio to 48kHz.

 Maybe the AC97 chip/codec just doesn't do a very good job of the resample in real time.  Probably safer to stick to what everyone is recommending and capture 44.1K and resample in s/w.

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

Sweet. Might just give this a go then just to see how that works.

Many thanks for the advice Moth3r, you've filled in a lot of the blanks :)

I'm using the Creative labs fatality with the front bay. This has optical and coaxial front inputs and is perfect for capturing digital audio

Thanks for the info Adywan, that's exactly what I needed to know.  Much appreciated mate :)

Update: FanFiltration has also recently managed to accomplish this task, details in this thread:

http://originaltrilogy.com/forum/topic.cfm/Bond-Laserdisc-Preservations-1962-1971/post/443970/#TopicPost443970

Excellent.  Thanks for that Moth3r, I had completely missed that James Bond  thread.