I now use the optical out on my Onkyo WAVIO SE-90PCI soundcard which brings my computers sound to my Benchmark DAC1, and that has worked very well so far. The only problem is that the SE-90 isn't supported in Linux yet, and it is quite a problem not having sound when I use Ubuntu. Therefore I am considering using the onboard optical out on my motherboard instead. My only "worry" is that the onboard optical out is not of the same quality as that of the Onkyo card. Is that possible? Or is a digital signal the same however it travels to my DAC1?
My motherboard has a soundcard of its own, so I won't be able to use it for a few days since it's on its way here by mail. My motherboard is Asus P5N32-E SLI, if that matters.
No input on whether it's possible to hear a difference between different soundcards digital output?
I am considering an alternative. An external usb soundcard would be very practical since I then easily could switch between my macbook and my desktop, so I am considering buying Asus Xonar U1. Do you guys have any input on that card? I will only use the digital output. Are there any disadvantages with an external usb card like the Asus?
There is no difference, if the digital output is working properly. (drivers dont botch up anything and so on, speaking from personal experience it does happen) The data gets moved in digital domain exactly like computer files from internet to your HD, zeroes and ones, bits that is, unless errors in trasnfer occur (in music playback it shows itself as jitter, in computer files it shows as corrupted files and CRC errors) due to reason or another, but its irrevelant. Its the conversion from digital to analog where the differences happen depending how its done and how its amplified.
Originally Posted by MaZa /img/forum/go_quote.gif The data gets moved in digital domain exactly like computer files from internet to your HD, zeroes and ones, bits that is, unless errors in trasnfer occur (in music playback it shows itself as jitter, in computer files it shows as corrupted files and CRC errors) due to reason or another, but its irrevelant.
Not exactly.......S/PDIF is a different format than data transfer over a USB, FW, or other data-specific interface because the clock is embedded in the data stream.
And audio jitter has nothing to do with the bits being in error......jitter occurs when the correct data arrives at the wrong time.
Originally Posted by sejarzo /img/forum/go_quote.gif Not exactly.......S/PDIF is a different format than data transfer over a USB, FW, or other data-specific interface because the clock is embedded in the data stream.
And audio jitter has nothing to do with the bits being in error......jitter occurs when the correct data arrives at the wrong time.
Thats how I understand it also but the question is how significant is this?
Is the jitter enough of an issue on on-board audio that USB or a dedicated soundcard is perfered? Also how dose digital jitter effect the sound?
The specs page for the Xonar U1 appears to indicate that it resamples everything to 48 kHz, which is not good.
That motherboard looks rather interesting. I may not be up to speed on the latest motherboard configurations......but it appears that there are both coax and optical S/PDIF outputs directly off the motherboard, but the analog outputs are out of a daughterboard. Is that common? If it were me, I'd try the motherboard connections first before worrying about inserting another link in the chain.
When I click on the link on the overview page to get the detailed specs, IE gives me an error....so I'm unsure how those connections are set up. I presume that they would pass whatever bit depth/rate is being fed by your player app, if the daughterboard can handle 24/192 streams.
The Asus P5N32-E SLI uses the Analog Devices ADI1988B onboard solution, which adheres to the Intel HD Audio specification. So, you should have bit-perfect digital output from the motherboard.
Digital jitter is an issue, but in the case of reclocking DACs like the DAC1, it's probably not significant enough to make an audible difference, even in an uber-expensive rig.
Mathematicaly jitter higher than 346picoseconds makes you loose "bitperfection"
As i understand the article above , different types of jitter should have different impact aswell.
Quote:
Originally Posted by Julian Dunn in her beforementioned paper [..]This shows that for low jitter frequencies the jitter amplitude may be very large before the sidebands
become audible, because the sidebands will be close enough to the original signal to be strongly
masked. As the jitter frequency rises above 200 Hz our sensitivity to the modulation increases
rapidly. This happens because for low signal frequencies the sidebands are no longer masked,
though for higher signal frequencies they still fall below the threshold. For jitter frequencies above 1
kHz the signal frequency that can cause unmasked sidebands increases. As sideband level is
proportional to signal frequency this results in increasing sensitivity with increasing jitter frequency.
This effect produces a slope of - 6dB per octave on the graph. The intercept level of this final slope
is determined by the non-masked threshold of hearing relative to the signal level. The figure was
drawn for a playback level of no more than 120dB SPL, assuming the worst case of sidebands being
audible at 0dB SPL.
This plot can be used as a specification for allowable sampling jitter in Nyquist sampled systems. At
20 kHz the peak to peak sampling jitter must be less than 20 ps, increasing at 6dB per octave for
lower frequencies until approximately 500 Hz where the limit is 1 ns. Below 200 Hz the jitter may up
to 500 ns in amplitude before the sidebands could become audible.
For oversampled systems the sampling jitter sensitivity may be worse. As the sampled signal could
have a higher frequency than the Nyquist worst case figure of 24 kHz the sensitivity increases
further. For a delta-sigma DAC any jitter at 150 kHz, for example, may modulate with the shaped
modulator noise at around that frequency creating modulation products falling in the most critical
parts of the audio spectrum.[..]
As infinitesymphony said, in the the end it's about how the DAC can handle the jitter
I have now gotten my digital out from the integrated toslink on my motherboard to work (Vista actually did all the job), and I can't spot any differences when it comes to sound quality. I have a couple of questions on how I should set the settings for optimal sound quality. Here are a couple of images of what I can adjust, should I keep it the way it is now?
Originally Posted by Svirre /img/forum/go_quote.gif I have now gotten my digital out from the integrated toslink on my motherboard to work (Vista actually did all the job), and I can't spot any differences when it comes to sound quality. I have a couple of questions on how I should set the settings for optimal sound quality. Here are a couple of images of what I can adjust, should I keep it the way it is now?
In the supported format box, uncheck Dolby Digital, and check all of the possible sample rate options.
It might be a good idea to switch to 2-channel / 24-bit / 44100 Hz for shared mode operation. In the DAC1 thread, it was said that 16-bit is not always bit-perfect in Vista.
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