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Chords New M -Scaler


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Apologies if this has been covered already, but it amuses me that the scaler thing does not have usb out. Funny how we were supposed to this that s/pdif and its inherent jitter /long term clock matching problem was supposed to be THE issue. And that was at 44.1kHz not 768kHz.

You are not a sound quality measurement device

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36 minutes ago, BigAlMc said:

Just read all ten pages of this thread and not one of you bozo's answered the all important question! Will this thing improve @BigAlMcs system! ?

 

Short answer I suspect is not without significant changes, as I'm not even close to being willing to consider parting with my Directstream DAC and because I have a heavily optimized USB flow.

 

But would appreciate some more expert input on the following. The M-Scaler sounds very impressive but it has USB input but no USB output. Therefore I could potentially put it after my TX-USBultra but not before.

 

So if I have a TX-USBultra (clocked by an SoTM OCX-10 reference clock) providing a very, very nice USB signal to my DAC. In theory would feeding that USB signal into an M-Scaler have potential benefits or be a dumbass idea?

 

Would the M-Scaler take that very nice USB signal and scale it to 386Khz or whatever making it even nicer? Or would the M-Scaler completely reclock/rejig the signal to the extent that the efforts (money spent!) on the TX-USB-Ultra/OCX-10 were rendered pointless or lost?

 

Cheers,

Alan

You have to weigh up the relative benefits of introducing inaudible jitter with the M-scaler as against the very real danger that by not using it you will leave in inaudible images at mulitples of 2x the upsampled frequency of your dac due to its using zero of first order hold approximation in the next stage of upsampling for sd modulation.

which one has the prettier box?

You are not a sound quality measurement device

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Just now, BigAlMc said:

 

Dude I thought you and I were in the same ballpark if not page when we posted at pretty much the same time referring to USB. Guess not! 

 

The Chord defo has the prettier box. It's very cute. 

 

Question stands if anyone wishes to opine with more than sarcasm. 

 

Cheers, 

Alan

 

I think technology is interesting and I sort of like the idea of aspiring towards perfect DA. But how do you compare the relative importance of two rather different ideas of which part of the process to polish to within an inch of its life? If you buy into the cult of Watts then you can probably ignore usb as IIRC he prefers S/PDIF. In which case buy with confidence. But don;t be surprised if there's a 2 million tap model next year.

 

You are not a sound quality measurement device

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7 hours ago, ecwl said:

So this is getting a little more off topic because it touches more on DAC design than Chord Hugo M-Scaler. Rob Watts, the Chord DAC designer thinks that async USB or digital PLL solutions are already excellent for jitter. He thinks most DAC jitter actually comes from the switching noise of the R2R or DSD/PWM or multibit SDM DAC chips. 

....

but in the sense that the same number of elements are constantly switching so that the switching noise is always the same which would control jitter and also eliminate noise floor modulation.

[as for the first thing -he may well be right, albeit selectively] You mean he thinks that the problme is the thermometer dac elements switching on and off. Not sure I follow- that simply is the output signal.

Anyway I'm genuinely intrigued as to what this means.  How can he have constant switching - if you go from min to max you have to switch on all the elements. So how does he ensure the same number of elements are changing irrespective of the signal- does he have all the elements change but have some of them out of circuit?

Presumably this could be demonstrated on a simple  J est - or do we need to refine the J test to involve maximum changes of dac element values (J test was perfected for 16 bit numbers)

You are not a sound quality measurement device

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For mathematicians out there- one of the things which has been puzzling me is that when I have played with using sinc function as a filter, it’s apparent that the windowing function seems to matter more than the width of the window to the frequency response (certainly comparing sensible window short sinc with long rectangular window)  

 

Would a bespoke window function be more promising than increasing of tap length to increase accuracy of reconstruction? Or perhaps an increase in precision of output? 

You are not a sound quality measurement device

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1 hour ago, mansr said:

The point is that it's just another windowed sinc (JAWS) filter. It's not even clear that this is the best choice for audio applications.

Could you expand on your thoughts on the best choice. -assuming we are sticking with linear phase is this just a question of the right windowing function or do you mean that we shouldn’t bother with an actual sinc?  Am I right in thinking that you can make something with more attenuation for a given filter length by not using a windowed sinc?

there is a plausible sounding theory out there that (to be on the safe side)  linear phase filters should not be longer than something to do with the ears’ filter bins possibly the time constant. I think that Jj over at HA and Fokus have both mentioned this point, but I have never quite grasped it. 

You are not a sound quality measurement device

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