XMOS-based Asynchronous USB to I2S interface - Page 34 - diyAudio
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Old 28th January 2012, 03:25 AM   #331
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Quote:
Originally Posted by regal View Post
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.
But the problem isn't jitter, its common-mode noise. Async USB is still affected by various software issues (people do report sound quality differences with async) so that's evidence which falsifies the 'jitter' hypothesis.

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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.
I'd not touch it with an infinitely long barge pole

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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.
I don't buy that for one moment. Yes the horsepower is there but the future is lower horsepower, with much much lower energy footprint. Why would you wish to exclude the tablet users of today (and tomorrow) from decent sound quality? Horsepower has been too cheap (a horsepower bubble if that's not too mixed a metaphor ) with the result that software has gotten hugely bloated. This bubble will pop.

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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?
Agreed that filter development will blossom in future, that's great for DIYers. Its not necessary for it to be on the PC for projects to start up using open source.

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Plus you get more rf away from your dac,
Indeed you do get more RF, and its conducted down to the DAC via the cable. So distance isn't really much of an issue.

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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.
None of the flexibility is lost by having the digital filter done locally.
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Old 28th January 2012, 08:49 PM   #332
Wolfsin is offline Wolfsin  United States
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Quote:
Originally Posted by regal View Post
Funny thing wolfsin is I still own an I2S input AudioAlchemy DAC
wannaBuy two more? :-) My parcel from Lucian is stashed in the P.O. until Monday but part of my plan was to lash up an AA Dac first to assure bitzRflowing and then go with Opus. Is their jitter caused by their 'dejitter' boxes?

I am not quite ready to return to the digital world right now but am quite anxious to see how LucianSolution + Dual WM8740 compares against four BB1704s and the Apogee clocking.
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Old 30th January 2012, 05:09 PM   #333
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Default But does it work?

Lucian wraps so that tanks and closely detonated explosives could not damage these little jewels. A friend happened to be at the post office and we had a nice long talk as we struggled to open the "adult proof" packaging

It really is beautiful (my friend, formerly with IBM, agrees) so now I need to get serious and hook it up. 'twas worth the wait!
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Old 1st February 2012, 05:16 PM   #334
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What is the deal with driver updates, will thesycon provide these or how is the
"next" operating system supported?
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Old 1st February 2012, 07:01 PM   #335
Wolfsin is offline Wolfsin  United States
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In the case of Winderz, there is a WHQL process by which hardware vendors get into a loop of testing, approval, digital signing, and inclusion in the next product release drivers. Not sure that is happening but would be surprised if it is not.
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Old 2nd February 2012, 02:23 AM   #336
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Hi Lorien,

I was wondering if you had heard of the recent developments with DSD audio via xmos.

Is there anything in your hardware that limits the ability to apply the firmware changes described here - DSD audio over USB: an XMOS Reference Design implementation
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Old 2nd February 2012, 02:31 AM   #337
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I have heard that DSD was designed for lossless translation into PCM at half the sample rate. Is this true?
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Old 2nd February 2012, 03:46 AM   #338
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yeah as i understand it, the PCM header is just used as a container to describe how the DSD information will be processed thereafter

I can certainly process DSD and DXD on another Xmos based unit
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Old 2nd February 2012, 07:50 AM   #339
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Quote:
Originally Posted by qusp View Post
yeah as i understand it, the PCM header is just used as a container to describe how the DSD information will be processed thereafter

I can certainly process DSD and DXD on another Xmos based unit
Wich one Jeremy?
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Old 2nd February 2012, 10:48 AM   #340
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Quote:
Originally Posted by Wolfsin View Post
I have heard that DSD was designed for lossless translation into PCM at half the sample rate. Is this true?
The frequency response at half the sample rate would be rather constricted, due to the limited slew rate, but that's really just the nature of DSD, not a side-effect of any translation. If you run at one sixty fourth the sample rate then there is more amplitude at the higher frequencies near Nyquist. The reality is that most material that is encoded as DSD is severely band-limited before being sampled, so there isn't really anything there at half the sample rate.

However, DSD can be translated into PCB without loss at any multiple of the sample rate, provided that you have sufficient dynamic range and hopefully some sort of adaptation to find the "zero" in the signal represented by a +1,-1 data stream, otherwise you'll have clipping. DSD is relative and PCM is absolute, so the only tricky translation is going from a relative system to an absolute system.

Looking at Wikipedia, there is an implication that DSD is equivalent to 20-bit PCM at 44.1 kHz (where they say 120 dB dynamic range, that's 20-bit; and where they quote 20 kHz response, that's better than 40 kHz sampling rate). Later in the article, DSD is compared to 20-bit at 96 kHz, but I say that the high frequency response of DSD would be severely limited in dynamic range compared to PCM. The thing to remember about DSD is that it is a significant low-pass filter, because the higher the frequency, the lower the maximum amplitude possible. PCM has no frequency-dependent amplitude limit, although frequencies near Nyquist should ideally be attenuated on input to avoid aliasing.

If you want to run a PCM DAC at 352.8 kHz from translated DSD input, then you'd get very little amplitude at higher frequencies, and practically nothing above 100 kHz.
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