[BHTX]

I just purchased a new Presonus Central Station to use as a DAC + amp. At the moment, I need a sound card with S/PDIF (toslink/coaxial) outputs (and maybe inputs) to run to the inputs of the Central Station..as I seriously need to build another pc, which I plan to do as soon as budget permits.

So, is it worth spending the money to get a good card when all you're doing is outputting the digital signal to an external DAC? And is an external USB sound card still the better choice in this situation? Or does it really not matter?

Also.. is there any possible way to input a 2nd digital source into the sound card, such as my Dish HD receiver, process it with a DSP or VST plugin (like the Bauer stereophonic-to-binaural headphone plugin that I use with Foobar), and then output that processed digital signal to the Central Station? In other words, is there any way to use this plugin or a similar one to process the digital audio from the toslink output of my Dish receiver before sending it to the DAC inputs?

 

[midnite8791]

i was under the impression it doesnt really matter, but i could be wrong

 

[BHTX]

Yeah, that's what I'm thinking too. However, I still wonder whether or not it might be beneficial to use an external interface. If not, I guess I'll just go ahead and get the cheapest decent PCI card I can find with S/PDIF outputs, and use that until I can get another pc together. But if I should be picky about it, I might as well spend the bucks and do it right. Anyone?

 

[Maniac]

Check out the thread in the dedicated source forum about PS3 as a source, where difference between sources are found to be quite noticeable.

 

[Balisarda]

S/PDIF output is digital. Any soundcard that outputs in S/PDIF will send out the same 1s and 0s.

Audible differences in sound quality between digital sources probably result from differences in the accuracy of the clocks controlling the digital output. An inaccurate clock will introduce timing inaccuracies into the output of the S/PDIF signal. If the DAC doesn't buffer the incoming digital signal and reclock it, those timing inaccuracies will be translated into jitter in the final product.

Most soundcards probably use the motherboard's master clock to time the digital signal output, so your choice of sound cards is probably unimportant. High-end cards like the Lynx Studio products may use their own clock, however--I don't know.

Some DACs, like the LessLoss DAC 2004, can actually control the timing of the outgoing signal in the source device, which solves your jitter problems without having to resort to buffering and reclocking (since the reclocking is performed at the source).

 

[Riboge]

Lynx cards do have their own clocks, and no, all cards with spdif output do not sound the same. The Lynx AES card is the best sounding card I have been able to identify, with their spdif output card second. The offramp turbo 2 by Empirical Audio, an external transport to spdif via USB, sounds the best.

 

[thomaspf]

Quote:

Originally Posted by Balisarda /img/forum/go_quote.gif Most soundcards probably use the motherboard's master clock to time the digital signal output, so your choice of sound cards is probably unimportant. High-end cards like the Lynx Studio products may use their own clock, however--I don't know.

Most sound cards have their own crystals which run completely asynchronous from the various clocks on the motherboard. Nevertheless the quality of the on board clocks might still be dubious. However, many modern DACs provide various forms a dejittering so that changes in the quality will result in very little if any change to their outptu quality.

Cheers

Thomas

 

[Sycraft]

For the most part, it doesn't make any difference. It is all digital. There are three possible considerations:

1) As mentioned, clock source. A better card should have a better clock. Does it matter? Well I dunno, good DACs will reclock anyhow, but maybe you don't have one that does.

2) No resampling. This is a little more important. Some cards like to resample all the audio they get. Some also do a pretty lousy job of it. The AC97 codecs you find on older motherboards are particularly bad. So you have your nice 44.1kHz audio poorly resampled to 48kHz and your sound suffers.

3) Good drivers. If the drivers are crap, it might affect sound, and also might just give you a bunch of headaches.

So I wouldn't necessairily cheap out and get the lowest cost card you can find, but then I wouldn't go and spend a ton either. Maybe get something from M-Audio.

In general save the money for the DAC. Get something good like a Benchmark DAC-1 and it'll handle all the reclocking itself complete with a buffer so more or less source jitter is totally irrelevant. Better to spend $1000 on a DAC that does that and $50 on a soundcard then $900 on a great soundcard and $150 on a crappy DAC.

 

[legcramp]

My onboard realtek HD audio sounds a tad warmer than my Xtrememusic running through my Zhaolu.

 

[Maniac]

I'll just say that jitter problem is never really "solved", and it would really be best to try it out to see it for yourself. Theory is theory, and it is nothing without a few experiments to prove it right or wrong. It's actually kinda fun to try this out.

 

[YZYgold]

Well, ive upgraded from onboard spdif to a prodigy hd2 spdif to my benchmark dac1 and i must say theres a huge difference.

 

[Hybrys]

Then you set something up wrong, or it's placebo. = D

 

[sonance]

This has been discussed quite a bit in a hundred different threads. Bottom line, some DACs are more susceptible to jitter than others. A really bad digital PC source can have tremendously high amounts of jitter compared to a decent CD player, to the point where a halfway decent head-fi rig can detect the differences if you have a NOS / non-reclocking DAC. Many modern DAC ICs and overall DAC designs do jitter compensation, and if that's the case, the differences in digital out will be greatly reduced.

 

[Hybrys]

Quote:

Originally Posted by sonance /img/forum/go_quote.gif This has been discussed quite a bit in a hundred different threads. Bottom line, some DACs are more susceptible to jitter than others. A really bad digital PC source can have tremendously high amounts of jitter compared to a decent CD player, to the point where a halfway decent head-fi rig can detect the differences if you have a NOS / non-reclocking DAC. Many modern DAC ICs and overall DAC designs do jitter compensation, and if that's the case, the differences in digital out will be greatly reduced.

We're not comparing it to a CD player. Thanks for throwing your own relevance out the window.

 

[Maniac]

Just to make it even more of a headache for those who feel digital signal are easily reclocked and processed... Here's a few points for you to ponder on.

#1 Think of digital signal as a water input pipe to your own water tank (reclock) which feeds your home (DAC). If you use a precise clock to reclock the signal, it might be just a little bit faster or slower than the input. And after a while, the small difference will translate into an overflow buffer or an empty buffer with nothing left in it.


#2 Sure, some uses VCXO to adjust the clock's freq to match the input more closely and use buffer's level as an indicator to speed up or slow down the clock. However, VCXO are often extremely jittery, which means the data could be more jittery AFTER the reclocking attempt. (I saw some company that actually did that... who? well, I guess I'll keep quiet on that...

)


#3 Async reclocking, which actually uses an SRC to convert any sample rate to a fixed sample rate. This will have an extremely stable output signal, however, I feel this is rather like the following example:

Think of digital signal as cookie dough, what Async SRC did is rather like taking in deformed cookie, bake them and package them in extremely good looking packages. The end product looked extremely good, but once you open it up (listening), it is still odd shaped signal as before, but the package looked good (digital signal looked nice). The main problem with this is that after the conversion, the hidden jitter will be impossible to remove (ie: baked cookie...)


The one method I admire the most is Genesis' Digital Lens, which uses method 1 (reclock with buffer), but they did it with a very carefully tuned buffer that will not overflow or underflow until about 74minutes (the length of a CD) after the start of the music. Very clever, and they did it like 15 years ago...