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#1 |
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diyAudio Member
Join Date: Apr 2004
Location: MN
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Ok I have read that paper but I need help clear up a few things for me.
From what I understand, all arguments in that paper held against spdif hinge on the assumption that the connection is severly band-limited. Which boils down to low bandwidth/quality cable and mismatched terminations (i.e not 75ohm on either end). And once that happens, the amount of jitter depends on the data value, or more precisely the amplitude of the original signal. Lower amplitudes yield higher jitter. What I am having a tough time understanding is that if the connection is NOT band-limited and is perfectly matched then is it still a problem ? Shouldn't it be just fine to use spdif then ? Is it really difficult to implement a not-so-band-limited spdif connection ? Or what am I missing here ? |
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#2 |
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diyAudio Member
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You're right that a higher bandwidth cable will improve S/PDIF jitter performance. It's stated right there in the Hawksford paper. But you still have the data dependence, because you get a different number of transitions on 1 bits compared to 0 bits. There's no way to solve this problem.
These days we have a far better means of transmitting digital audio: I2S protocol using LVDS on shielded twister pair cable. Simple, cheap, and effective. Still, nearly every amateur will be implementing at least one S/PDIF receiver for compatibility reasons. |
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#3 |
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diyAudio Member
Join Date: Mar 2002
Location: Hawaii
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Bandwidth works two ways. Keep in mind you also need very good low frequency performance with S/PDIF. The better you do the bottom end, the less jitter you get from the HF edges.
jh |
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#4 | ||
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diyAudio Member
Join Date: Apr 2004
Location: MN
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Quote:
if the above statement is correct then if there was some way of eilminating the pll (fixed clock(s) with "manual" sample rate selection) then even this wouldn't be an issue, right ? Quote:
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#5 | |
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diyAudio Member
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Quote:
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#6 |
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diyAudio Member
Join Date: Apr 2004
Location: MN
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ok got it!
although with that approach and, also with I2S, it would mean sending a very vulnerable XX Mhz square wave signal over a wire. Wouldn't that be an open invitation to RFI/EMI ? Guess what I am saying is that it might fix one problem but create another, and still not be the silver bullet we are seeking ? Seems like discarding or not even recovering the clock at all and implementing a "buffered" DAC with its own independent clock would be a perfect solution, atleast theoritically. |
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#7 |
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diyAudio Member
Join Date: Feb 2004
Location: Illinois
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As mentioned earlier, if you used LVDS (and good wire) to transport the signal I'd reckon that would almost be a non-factor. LVDS is used in hundred of megabit connections (100s of MHz+) on backplanes etc. A couple of MHz of I2C should be no big deal.
Proper termination would still be important (LVDS looks for 100ohm differential) and you couldn't expect to have 100s of feet of cable either. I'd imagine the inductance would go through the roof and square waves don't like inductance. |
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#8 | |
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diyAudio Member
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Quote:
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#9 |
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diyAudio Member
Join Date: Apr 2004
Location: MN
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I2S with LVDS does look promising.
jwb, I found a really old thread started by you, in which you discussed about using LVDS. There weren't any updates later though. What was your experience with that ? How did it go ? I'd interested in checking out work done in this area. |
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#10 | |
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diyAudio Member
Join Date: Feb 2004
Location: Illinois
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Quote:
Theoretically, nothing is a square wave outside of an idealistic world. I think it is a safe enough assumption that sub nanosecond rise/fall times with few MHz cycle time signal can be lumped as "square". |
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