Does the soundcard matter when using a digital output? I have a cheapy SB Live that has 1/8 digital output that i just ran a convesion plug to the coax connection on my reciever, and it sounds pretty good.

Is there any advantage buying more expensive cards for SPDIF/COAX outs? Considering its only a digital signal, im not sure what the advantage would be.

Are there hardware/software issues with sending surround encoded material (Dolby/DTS) through certain cards' digital connections?

Any info on the subject would be much appreciated.

I'd also like to know this. Also: is there any quality difference between motherboard spdif and sound card spdif?

I have the same issue. I have integrated nForce Soundstorm which is connected to a lovely Yamaha RX-V650 via coax SPDIF.

I think the only problem we may have is the 44.1kHz -> 48 kHz resampling, which most cards do.
Now resampling is pretty standard DSP technique and I do not think it is so difficult to get it right especially that the whole thing is just arithmetic wizardry.

Triza

Some sound-cards/drivers resample everything to 48KHz in a crappy way (leading to clipping), even when digital-out is used. You can test your setup with udial sample (be aware of danger, read the thread before playing with it).

rutra (and Jan S., if listening): udial.zip seems to be corrupted.

Can you confirm or it is just me?

Sergio

Something seems to be wrong with the archive indeed. But I succeded to take the sample out of it with 7zip - first you extract udial.zip and you get a file named [Content], then you extract that [Content] file and you get udial.ape.

i couldnt figure out how to have my SB Live card play surround through the digital output. Does this card only output to stereo through the digital connection, or do i need specific drivers? Audition couldnt even use it in its multichannel encoder.

If you mean outputting AC3/DTS you have to find a new item in the control panel, installed by creative software.

its often called Creative Control, Audio device control or something of the sort.

there will be an option to either have AC3/DTS decoded by the soundacard, or passed onto the S/PDIF.

If you mean multi-channel output like that of games or a 6-channel wav file, then its simply not possible. A digital-out can only pass 2-channel uncompressed audio. you can output AC3/DTS because they are compressed audio formats, similar in MP3 et al in the broad concept.

since there is no commercially-priced card that can do live ac3 encoding (that i know of), youll have to connect your amp's 6-channel in to you audigy's 6-channel out.

..i hope that's clear enough its really late and im tired

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regarding the non-resampling digital-out issue... its generally accepted that creative cards have sub-par resampling to 48KHz... this might have changed in the very latest generation of cards, but it hadnt in the last audigy2 i had.

in most of the new ones it is possible to select 44.1 as a mixing sampling rate, but since the digital-out is not capable of 44.1 it gets resampled anyway.

in much the same way, the windows xp/2000 mixer and your soundcard driver will resample to 48KHz internally (please confirm/deny?) and again this resampling is sub-par.

there are 3 ways to get around this:

1) use foobar2000 as a music player, add the PPHS Resampler (set to 48KHz) to you DSP stack, and use kernel-out to output your music.

2) if you have one of the top-end latest-gen audigys they support ASIO2.0. in which case any music player with good resampling to 48KHz and ASIO output capability (either builtin or though plugin) should be able to produce bit-perfect output to you digital receiver.

3) buy a non-creative soundcard. you can get some under-40$ cards that do bit-perfect 44.1KHz output on the digital connections.


as to why a sup-par resampling will adversely affect your music playback quality, i suggest checking the FAQ to this site, and then searching for such topics as "dithering" and "quantization noise" (spelling?) and "aliasing" in the forums here.. though i suspect youll find more than enough info in the FAQ.

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please disregard any idiocies/breaches of terms/etc as products of my insomnia-induced dementia. I blame my linear algebra lecturer for only managing to confuse the course material further.

JT

There is the resampling issue, but you also have to take jitter into account.

Do you know where i can get this software? I looked on the creative site, but for the Live value cards it only had Liveware drivers without the creative control panel.



I was able to create a 5.1 AC3 mix and play it back in Dolby Digital through my Santa Cruz card. Are there any other compression formats that can output lossless 5.1 through a digital connection, or are there none, and would that answer why formats like DVD-A and SACDs play surround through analog outputs?

2 questions

1) regarding the F2k/PPHS/kernal suggestion, why resample at all?

2) I've got a laptop with an integrated Conexant XMC Audio 'soundcard' . I run the audio from one of the laptop's three USB ports to my receiver (which has a USB input),using F2K for playback. Any idea if this is resampling or not? The Windows 'Advanced Audio Properties ('Performance' tab)' control panel does have a 'sample rate conversion quality' slider, suggesting it does (it's set to 'Best'), though I have the Speaker setup set to 'no speakers'. I've been using Direct Out in F2K so far, haven't tried kernal streaming on this computer yet.

When you output 48KHz audio stream, some soundcards/drivers won't resample it as it's already 48KHz, so there won't be clipping. Sometimes it won't help as some soundcards/drivers still resample to 48KHz even when the stream already is 48KHz.

Resampling to 48KHz is a technical need (AC97 specification), so it does resample for sure, the question is if it's done in a good way or in a crappy way (leading to clipping). To check it out use the udial sample I mentioned a couple of posts above.

The problem with resampling is not clipping. It is aliasing.

then it will be somewhere in the creative mixer or eax mixer. creative cards come with a needless amounts of control panels and extra bulk software. most of the options in there are either uselss or detrimental to sound quality.

just give a good look at all the setting and youll find some option like:

use inbuilt ac3/dts decoder [checkbox]

OR

enable AC3/DTS pass-through [checkbox]



AC3 and DTS arent lossless. in fact a stereo AC3 track is 192KBps. DTS is larger in size, and supposedly better in quality, but i must admit to having near to zero experience regarding DTS.

The latest beta of the AC3filter DirectShow filter is capable of AC3 encoding, but that still wont help for games.

unless youre soundcard does hardware ac3 encoding of discreete multi-channel sources (and im quite sure the creative cards dont do it) then discreete 6-ch out is all youll get for non-AC3/DTS sound.

JT

agreed. but some sound cards are so bad at it that they introduce clipping as well. see my laptops godawful onboard soundcard.

i cant be sure, but i think both of the last creative cards i had resampled anyway, even with 48KHz streams through kernel-out.

my philips aurilium usb still only outputs @ 48KHz in digital, but the sound is appreciably better than the Audigy2NX i had before. even my flatmates remarked on it:

"what did you do to your pc music sounds better now"
"new sound card!"
"aaahh" sort of thing

hence i assume the 2NX resampled the 48KHz streams anyway, and this philips card doesent.

FB2K 44.1 -> 48 & Creative 48-> 48

is still better than

Creative 44.1 -> 48

but not by much to be honest. might have been placebo.

JT

But that aliasing comes from clipping, doesn't it?

Are you talking about the Auto Matrix option? Cause I want to do something like Dolby Pro-Logic II processing for getting sound from all my 5.1 speakers when listening to stereo sources, so I enabled Auto Matrix in AC3Filter. But I'm not very satisfied with it. The surround speakers sounds very low, and they produces the same sound (a mix of the L and R channels) fucking up the L/R distribution. I mean, it doesn't sound great as in a Dolby Pro-Logic II system.
But you're saying AC3Filter can do real-time AC3 encoding. Does it means that AC3Filter can do 5.1 distribution of stereo sources like Dolby Pro-Logic II? How?
I'm using a SoundBlaster Live 5.1 with Logitech Z-640 5.1 speakers, which has analog connections. But, so that nobody can say that I am offtopic: if AC3Filter does this I'm asking, as I could get 5.1 from the analog outputs, you could get 5.1 (AC3 encoded) from the digital output! Right?

DTS is capable of lossless 2496 6.1
Its the future. Should be less trouble than AC3 too.