XMOS-based Asynchronous USB to I2S interface - Page 33 - diyAudio
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Old 23rd January 2012, 01:58 PM   #321
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@ adelias: boards are identical as previously made WaveIO cards. It's just a new batch, not a change in design.
Cheers,
L
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Old 23rd January 2012, 09:29 PM   #322
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Originally Posted by rsdio View Post
Excellent summary.

It is important to keep in mind the design constraints of any existing solution. I2S was never designed to connect two separate systems which do not share an identical ground reference, so I would say that not only does I2S work best over short distances, but it almost absolutely needs both ends running on the same power supply. As distances get longer, ground loops appear or ground potentials increase, and when the distance spans two enclosures you almost surely have a separate power supply, making the problem even worse.

If you use a standard outside of its design constraints, then you're very likely to fail to get the original benefits. Whenever you connect two separate enclosures with different power supplies, you need some kind of interface that is designed to deal with the different references. I2S was not designed for that. But it makes sense that "LVDS balanced I2S" could enhance plain I2S with the balanced additions handling the potential differences in reference levels (ground).

There's no need to get into the drawbacks of SPDIF and AES3, but the one thing those ancient standards have going for them is that they were absolutely designed to connect separate pieces of gear. While it's obvious that we need to look for modern replacements so that we can finally say goodbye to the failures of SPDIF & AES3, that doesn't make it possible to shoehorn something that was not designed to overcome the same obstacles into a solution without considering the additional requirements.

I think it's great that PS Audio, Wyred 4 Sound, and Twisted Pear Audio have pioneered a better solution. Are there any convenient links to the technical details of these interconnects? Sorry if the question has been answered in this thread already, but I recall suggesting LVDS interconnects on diyAudio and someone suggested that it would be a bad idea due to increased jitter. If the above three companies are having success with LVDS I2S, then I'd like to learn more.
The I2S/LVDS solution was originally developed in 2008 by Rockna Audio and PS Audio. A interface schematic is freely available here : I2S lvds interface | AD LABS
For any other info regarding the interface one should feel free to ask.
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Old 23rd January 2012, 11:40 PM   #323
rsdio is offline rsdio  United States
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Originally Posted by Lupulroz View Post
The I2S/LVDS solution was originally developed in 2008 by Rockna Audio and PS Audio. A interface schematic is freely available here : I2S lvds interface | AD LABS
For any other info regarding the interface one should feel free to ask.
Thank you!
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Old 27th January 2012, 08:38 AM   #324
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Quote:
Just see Art's detailed measurements on audio circle if you question that
could you please send a link to that? In addition, I would like for this thread to stick on topic as long as for now, WaveIO card does not support any kind of I2S transmission using differential PHYs.
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Old 27th January 2012, 04:57 PM   #325
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Originally Posted by Lorien View Post
could you please send a link to that? In addition, I would like for this thread to stick on topic as long as for now, WaveIO card does not support any kind of I2S transmission using differential PHYs.
Kind regards,
L
Some insight on how jitter is measured.

I have the legato so its asynch usb - dir9001 - pmd100 - pcm1704. There is no reclocking in the DAC, so if the cable/impedance match is good then the best I am doing is 3ps rms (legato) + 50ps (dir9001 instrinsic) + PMD (?) + PCM1704 (?).

The only galvanic isolation is the spdif pulse transformer but the usb +5V PS isn't used and the power supply is very unique, as is the reclocking prior to transformer to spdif. The sound of hirez (24/96 material) downsampled and dithered with Ozone 4 to 16/44.1 is better than a LiFePO4 Hiface at the musics native 24/96.

I guess what I am saying is SPDIF done right doesn't really add more jitter than i2S GMR's.


I will remove the following paragraph if you feel its off topic, but I think it shows the direction this technology will be heading and LVDS isn't part of it.

The best possible solution is to place the asynch USB clock(s) at the DAC providing synch reclocking (alignment on the critial i2s line for the dac). Then send this clock signal back to the Xmos via GMR, and the remaining I2S GMR.
This way the GMR's add no jitter, neither does the digital filter, and of course their is no receiver with its intrinsic additive jitter. I think the Wave I/O can be setup this way my a competent modder. This is the future for computer as transport, but obviously we don't want just I2s output, we want it split left and right, right justified binary 2s compliment, up to 768khz, get rid of the digital filter, let the computer do the over/upsampling/filtering/data conversion with more precision than a chip. Proper conversion down from the 32 bit output and compatability with from the classic TDA1541 to the PCM1704 is important. This is how things will end up in 5 years or so I hope.
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Old 27th January 2012, 05:24 PM   #326
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Quote:
Originally Posted by regal View Post
This is the future for computer as transport, but obviously we don't want just I2s output, we want it split left and right, right justified binary 2s compliment, up to 768khz, get rid of the digital filter, let the computer do the over/upsampling/filtering/data conversion with more precision than a chip. Proper conversion down from the 32 bit output and compatability with from the classic TDA1541 to the PCM1704 is important. This is how things will end up in 5 years or so I hope.
If this isn't too off topic I'd just like to mention that all the reports from audiophiles I've noticed say that having the computer do less computation sounds better than with more. So they say things like '.wav sounds better than .flac' and 'I stripped down the OS to only the bare minimum number of processes and that improved the sound'. So why would we want to add more things for the CPU to do? Using a dedicated digital filter will generate less noise than asking a sweaty 65W+ Intel CPU (with attendant local buck regulator) to do it, that's for sure.

Why would we want to tie in our audio systems that much more closely in with our PCs when the PC is already on the way out?
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Old 27th January 2012, 05:35 PM   #327
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Quote:
Originally Posted by regal View Post
This is how things will end up in 5 years or so I hope.
Wolfsin can just imagine the Audio Alchemy designers reading that forecast and getting a good belly laugh. That something as simple as reconstituting digital audio could have taken so many years is testimonial to marketing over engineering. RedBook dates to the early '80s.
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Old 27th January 2012, 07:17 PM   #328
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Default This is...

"The best possible solution is to place the asynch USB clock(s) at the DAC providing synch reclocking (alignment on the critial i2s line for the dac). Then send this clock signal back to the Xmos via GMR, and the remaining I2S GMR."

how competant USB DACs work now (Ayre, Wavelength Audio), OK, except they use optocouplers instead of GMRs.

It would be really cool if Lorien could add this functionality to the Wave IO: it seems a small board (daughter) for placement right at the DAC, with the two oscillators, necessary regulation, and I2S reclocking could be made. Then a new Wave IO which accepts MC input, and can communicate the clock switching necessary to the daughter board for the two SR frequencies. This approach would also allow the I2S lines to be a little longer without problems, giving more flexibility for placement of the Wave IO in the chassis where it's associated RF field can be isolated from the DAC and analog circuitry.
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Old 28th January 2012, 03:15 AM   #329
regal is offline regal  United States
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Quote:
Originally Posted by Wolfsin View Post
Wolfsin can just imagine the Audio Alchemy designers reading that forecast and getting a good belly laugh. That something as simple as reconstituting digital audio could have taken so many years is testimonial to marketing over engineering. RedBook dates to the early '80s.
Funny thing wolfsin is I still own an I2S input AudioAlchemy DAC, believe me when I tell you that they didn't understand the concept, they is so much jitter added in their handling of I2S input to the DAC it is scary, read that theirs were some of the highest jitter measurement DAC's ever made. Still a good sounding little unit PMD100-AD1862, going to get an upgrade soon.
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Old 28th January 2012, 03:38 AM   #330
regal is offline regal  United States
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Originally Posted by abraxalito View Post
If this isn't too off topic I'd just like to mention that all the reports from audiophiles I've noticed say that having the computer do less computation sounds better than with more. So they say things like '.wav sounds better than .flac' and 'I stripped down the OS to only the bare minimum number of processes and that improved the sound'. So why would we want to add more things for the CPU to do? Using a dedicated digital filter will generate less noise than asking a sweaty 65W+ Intel CPU (with attendant local buck regulator) to do it, that's for sure.

Why would we want to tie in our audio systems that much more closely in with our PCs when the PC is already on the way out?
That was before asynch USB when the computer was supplying the clock and there was some validity to accessing memory and cpu resources affecting jitter. If you go to that forum now one of the most popular DAC's for them does the upsampling in the computer to 16x and what's left of the DAC is an "NOS" DAC. It only makes sense to do the digital manipulations with the computer, the horsepower to perform algorthms is there and only makes sense that a PC is going to have more capability but more importantly flexibility. Digital filter programming is the one area in high end audio that is still fertile for development, as it moves to the computer, more programmers, open source projects, imagine a foobar plug-in with code of the caliber of Berkley's Alpha? Plus you get more rf away from your dac, you have the flexibility to use good but forgotten chips like the PCM56k or the AD1865 with the proper data shifts. Just huge flexibility replacing the digital filter with the computer.
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