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Review and Measurements of Empirical Audio Synchro-Mesh

GoMrPickles

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One observation that I will make here is that the jitter around the tone is much less with the Synchro-Mesh. The baseline noise level is higher, but this is still in the noise and I would argue not audible. The difference in the hot pink plot and both the blue and flesh colored plots that concerns me about the Gustard is the widening of the tone spike at the bottom. I have correlated this shape to degradation in audio quality in the past. The Synchro-Mesh plot has virtually no widening at the base.

If you believe my premise that it's the correlated jitter that is more audible than random jitter, then the widening of the base of the fundamental in this plot is precisely that, jitter that is frequency modulation close to the the frequency of the fundamental. The Synchro-Mesh plot contains zero of this.
Hi Steve, all,
Are you aware of any studies that look at this? I did one quick google and found this:
https://pdfs.semanticscholar.org/d9bf/d506271a8c38cf0f77e6edfbffebf5e368b6.pdf
but I'm well outside my area of expertise here.

I'm not sure what an experiment to test your hypothesis ("correlated jitter ... is more audible than random jitter") would look like, but I'd be interested in the results.
 

amirm

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Perceptual masking does not allow that bit of random jitter to be audible:



Masking threshold is a shadow on each side of any music tone that stops close-in sounds to be heard.
 
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@amirm This you have to explain: “For the next test, let's instruct the Audio Precision analyzer to generate a jitter tone that is in the form of sine wave at 5 kHz which has a "magnitude" of 1 nanosecond.” Assuming that “magnitude” here means the length of the signal, I.e. it feeds in the 5kHz signal for one nanosecond. How do you define a 5kHz frequency in a 1 ns long signal, and how does the electronics identify, measure the frequency? I use “measure” the same way it is used in quantum mechanics, i.e. how does it have a specific effect? Now, you say only that the sample is 1 ns long, you don’t say how often it repeats. But it seems to me that trying to represent that signal with 24/96k sampling, you would get zero with a probability of 0.99999 and any number between -2 million and +2 million, each with a probability of O(-11). And if the ns signal repeats often enough that you can somehow identify its frequency, certainly chopping it into ns fragments would introduce a lot of GHz hash.

Time precision and frequency precision are inversely related, similar to Heisenberg.

So please explain.
 

SIY

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1 ns is the magnitude of the jitter.

IOW the magnitude of the jitter goes back and forth from zero to 1 ns at a rate of 5 kHz.
 
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1 ns is the magnitude of the jitter.

IOW the magnitude of the jitter goes back and forth from zero to 1 ns at a rate of 5 kHz.
Ah. Not length at all. I misunderstood.
So the jitter varies between -1 ns and 1 ns, sinusoidally, with a frequency of 5 kHz?
 

Blumlein 88

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Okay, My wife and I both did this, at different times. She is always busy...

Result: I picked 4 out of 4 correctly.

My wife picked 3 out of 4 correctly.

The thing that we both noticed is that in at least 2 of the tracks, the 8X version sounded more focused. This made it difficult, because the altered track actually sounded better than the original, but didn't match the reference. Usually, we listen for the better sounding track, not trying to match two tracks, so we had to resist this tendency.

I have a theory about this improvement in focus in the 8X tracks:

As I believe I mentioned in the other thread, some of the early re-clockers used several PLL receivers in series to reduce jitter because one was not good enough. They went from receiver to transmitter to receiver etc.. S/PDIF to I2S to S/PDIF to I2S and on and on. In this case, we are going from S/PDIF or I2S to analog to S/PDIF or I2S to analog and on and on. I think the effect of the series PLL's might be the same, reducing the jitter in the end result and thereby improving focus. Because the altered tracks actually sounded better than the originals indicates perhaps that the compression and harmonic distortion added by the multiple A/D and D/A is less obvious than eliminating some jitter. This reinforces my desire to see the relevance of these different distortions studied with DBT tests. It's not unlike amplitude sensitivity. If you cannot hear it, it's not important.

Steve N.
All was USB Asio no PLLs involved.

Correction the DAC was USB and the ADC was Thunderbolt. Still no PLL's in use. DAC with asynch USB has free running clock and so does the Antelope audio ADC.
 
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pkane

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iFi SPDIF iPurifier is a fraction of the cost and said to be bit-perfect. If one is in the market for a re-clocker why pay more?
Since @Empirical Audio was so enamored with iPurifier, I thought maybe I didn't give it a fair shake when I first tried it. I went back and measured it with a couple of DACs: Emotiva XDA-2 and Gustard X20pro. iPurifier was connected via Toslink at 24/96kHz for all tests.

In the case of the jittery XDA-2, iPurifier actually made things worse with a couple of large new spikes (blue = iPurifier):

xda2-nosrc.png



In the case of X20Pro, with no internal reclocking (auto clock mode) iPurifier seemed to help quite a bit. Blue = iPurifier again:

x20pro-auto.png


But once internal reclocking was engaged (clock = normal) on X20Pro, iPurifier didn't seem to make much of a difference:

x20pro-normal.png
 
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Frank Dernie

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Ah the cool DAS-702es or the 703es? I'd forgotten about that. They were interesting alright. Sony pairing D/A equipped CD players with a matching but separate D/As. Didn't makea lot of sense to me. The later 703 used the Philips TDA-1541. Strange period where Sony was using Philips' D/As while they waited for the BB PCM-58 and their own PWM converters.

What happened to the transport? I hope you still have it- it can be resurrected.
Mine is the original, the DAS-702ES, not sure what happened to the CD player, my son used it.
 

Empirical Audio

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All was USB Asio no PLLs involved.

Correction the DAC was USB and the ADC was Thunderbolt. Still no PLL's in use. DAC with asynch USB has free running clock and so does the Antelope audio ADC.
Ah, then it is a mystery unless somehow the USB interface improves jitter a little each time. It cannot improve other types of distortion can it?
 

Empirical Audio

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Since @Empirical Audio was so enamored with iPurifier, I thought maybe I didn't give it a fair shake when I first tried it. I went back and measured it with a couple of DACs: Emotiva XDA-2 and Gustard X20pro. iPurifier was connected via Toslink at 24/96kHz for all tests.

In the case of the jittery XDA-2, iPurifier actually made things worse with a couple of large new spikes (blue = iPurifier):

View attachment 23824


In the case of X20Pro, with no internal reclocking (auto clock mode) iPurifier seemed to help quite a bit. Blue = iPurifier again:

View attachment 23825

But once internal reclocking was engaged (clock = normal) on X20Pro, iPurifier didn't seem to make much of a difference:

View attachment 23828

I don't use iPurifier on my music system, only on movie/TV audio. I think the jitter from my Samsung TV and my cable box is probably huge, so it helps. My direct measurement showed 10X the jitter of the Synchro-Mesh.
 

garbulky

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Do you have a way to determine if it's in fact bit-accurate as claimed?
I'm still trying to figure out what's the impact of resampling CD audio 44.1 khz to 96 khz in an ASRC. Doesn't that affect a bit perfect signal and also doesn't it affect the timing or perhaps even the frequency accuracy? Twox of 44.1 khz is 88.2. 96 Khz is not an exact multiple. (I'm referring to the Syncro-mesh here).
 
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Empirical Audio

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Just tested iPurifier at 24/96k and it is indeed bit-perfect (although not jitter-free, as demonstrated in my previous post):

View attachment 23859
I don't think it would pass Dolby Digital and DTS if it wasn't bit perfect. The output waveform is actually textbook perfect into a 75 ohm load. Best I have ever seen and about 800psec risetime.
 

Empirical Audio

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I don't think it would pass Dolby Digital and DTS if it wasn't bit perfect.
I'm still trying to figure out what's the impact of resampling CD audio 44.1 khz to 96 khz in an ASRC. Doesn't that affect a bit perfect signal and also doesn't it affect the timing or perhaps even the frequency accuracy? Twox of 44.1 khz is 88.2. 96 Khz is not an exact multiple.
I don't believe that the iPurifier does ASRC. It's another technique.
 

gvl

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Nice, thank you. Is this sample by sample check
I don't think it would pass Dolby Digital and DTS if it wasn't bit perfect. The output waveform is actually textbook perfect into a 75 ohm load. Best I have ever seen and about 800psec risetime.
Couldn't it just detect a compressed stream and enable pass-through?
 
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