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On 5/18/2017 at 1:26 PM, julian.david said:

The REF 10 is arguable the most compatible 10 MHz clock out there and it will work in conjunction with various 10 MHz compatible DACs and clocks by manufacturers - not just MUTEC. It is however a strict 10 MHz clock, so you will need a 10 MHz reference input of some sort. The dCs Vivaldi Master Clock has a 10 MHz compatible reference input and the REF 10 will be an excellent choice to drive this input. But the other Vivaldi products (Upsampler, DAC) only have Word Clock inputs and (unfortunately) don't provide a 10 MHz input. So you would need some other master clock in between (like the dCs or the MUTEC MC-3+/MC-3+USB) to generate and distribute the Word Clock signal to the upsampled or DAC. 

 

Unfortunately dCs doesn't publish any phase noise figures for the Vivaldi Master Clock, which is the figure to pay attention to. But to my knowledge, the phase noise performance of the REF 10 is unmatched at the moment!

 

Had a great time with the new REF 10 today.  Thank you, Julian (and Christian)!

 

Unbelievable noise floor measurements from 10Hz to 10kHz but ultimately, the proof is in the listening.  Even amidst the noise of the MOC in Munich and the fact that the REF 10 was connected to the MC-3+USB which was then connected to a modest headphone setup via a high jitter Toslink optical cable, the impact of this clock over the MC-3+USB's standard clock was amazingly obvious from the standpoint of increased air around voices and instruments and size of the sound stage but also a much smoother presentation.  In comparison, the MC-3+USB's stock clock was almost unlistenable.  Quite surprised by this delta and much much larger than the delta I heard with the Vivaldi and the Vivaldi Master clock.  Steep price and despite my initial skepticism based on experience with other master clocks, I found myself already reaching for my wallet for what I now consider a must have.  Perhaps, the surprise of the show for me and I will probably use this to synchronize 3 SOtM sCLK-EX clock boards.  Amazing performance as well as versatility!

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2 minutes ago, Confused said:

I'm kind of thinking aloud today, but to put my last post a different way, if I could get somewhere close to the performance of the dCS Vivaldi Upsampler + Clock (£24k in the UK), with something like a PC with HQPlayer, sMS-200Ultra, Mutec MC3+USB & Ref 10 Clock, then I would be very happy indeed, and I do actually suspect this is possible.  But is there a better way?

 

I would suggest you get the sMS-200ultra with the master clock option so that you can extend the benefits of the REF 10 to that device as well.  To be able to synchronize that device with the Mutec should result in a very nice further improvement.

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Just now, Confused said:

Thanks romaz, that does look like a very interesting option.  I have been following the 'A novel way....' thread with great interest, so I get the point, however I had not yet worked out the possibility of feeding the sMS-200Ultra with a Ref 10.  That's quite a compelling idea, I wonder! B|  And even if it did not work out for reasons as yet unknown, you would still have a Ref 10 fed Mutec to keep you happy.

 

It will absolutely work with the sMS-200ultra.  I have already verified this with SOtM.  To be able to synchronize the clocks on both the Ultra and the MC3+USB will be huge.

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26 minutes ago, jelt2359 said:

To use this with the SMS200Ultra, would you need to feed it with a signal from the Mutec 3+ USB? Or can you go direct Ref10 -> SMS200 Ultra?

 

Each device requires its own 10 MHz input from the REF 10. Fortunately, the REF 10 has 8 outputs.

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19 minutes ago, Confused said:

Thinking about this, why do you need the sMS-200Ultra, which from what I understand is basically a sMS-200 with better clock?  If the original sMS-200 can be modified to accept an external clock, is this not the best 'bang for your buck' solution?

 

The standard sMS-200 is incapable of a master clock input. The new sCLK-EX board in the sMS-200ultra, however, is capable of such an input although the option to fit it with the necessary BNC connector will cost an additional $200.

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  • 5 weeks later...
4 hours ago, 4est said:

Forgive my ignorance please, but how are these clocks synchronized? I do not understand how say a master clock at 24.567MHz gets sync'd to USB at 12MHz(or whatever it is).

 

I am not a clock expert and so I have no choice but to accept statements made to me by clock designers and manufacturers at face value but according to Lee, SOtM's lead engineer and the gentleman responsible for the sCLK-EX clock board, the 4 clock outputs on this board are capable of independent frequencies but ultimately, the precision of their timing is "synchronized" to the performance characteristics of a single internal oscillator, an "internal master clock", if you will.  A similar analogy would be a computer motherboard's system clock operating at 24.567MHz but providing timing to various buses on the motherboard via DPLL even though those buses operate at frequencies other than 24.567MHz.  There is a language barrier when speaking with Lee since English is not his native language but he used the word "synchronized" repeatedly as the reason for why his clock board sounds so good and the primary reason why any audiophile should consider an external master clock.  Regardless of the exact technicalities involved, what matters more to me is the resultant improvement in SQ.

 

Isolating the impact of a clock in a component isn't always easy but with SOtM's standard endpoints, such as the sMS-200, tX-USBhubEX, or dX-USB HD and their corresponding "Ultra" versions, the only difference is the clock and so it becomes easy to appreciate the difference their new clock makes and the difference is surprisingly large with respect to increased air around voices and instruments resulting in a more 3D presentation but also resulting in the better layering of detail and discernment of subtle nuances that are easily and consistently differentiated with blind testing.  Even when their new clock was applied to an inexpensive $20 network switch, the improvement was shocking and when 4 Ultra endpoints were combined in series, all I could do was shake my head in disbelief with how my music had taken on new life while sounding smoother and more grain-free.  This wasn't any form of artificial romantic coloration that I could detect, it was simply realism taken to another level.

 

Now, when I got a chance to first listen to the REF10 in Munich, I was aware of all the criticism against external master clocks by the nay-saying experts and how the long clock cables involved would surely degrade the signal and based on my mediocre experience with the dCS Vivaldi master clock last year, I was admittedly skeptical but the REF10 has completely changed my mind about how impactful a well implemented external master clock can be.  In the same way that SOtM's sCLK-EX transformed their standard endpoints, the REF10 did the same with their MC-3+USB but perhaps to an even greater degree.   I hope to receive my REF10 and a pair of Habst clock cables in a few short weeks and so I will get a chance to hear for myself its impact in my system.  I will report my findings at that time.

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On 9/11/2017 at 1:44 PM, barrows said:

The DS DAC from PS Audio always operates in master mode for its converter section, as it is an asynchronous DAC by design, that is it resamples all incoming data asynchronously to its single internal masterclock.  This does not matter what input you use.

 

Here is John Swenson's response to asynch USB DACs:

 

"But what about asynch USB, isn't the DAC in control? Overall yes, the DAC has its OWN FIFO and also checks it, but instead of changing a clock frequency it sends a command back to the computer which tells it to speed up or slow down the average sample rate. So even though the local DAC clock is in ultimate control of the sample rate, as far as the MC3+/USB is concerned the USB data stream is in control, it just passes it on down to the DAC."

 

 

 

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24 minutes ago, barrows said:

@romaz  Exactly, but i think you are missing the point I was trying to make:

 

For someone like you with a Digital front end >$20K (blu+DAVE) improvements in your D/A converter are going to be hard to come by, perhaps impossible, so going all out with experiments on source feeding that combo may be justified as your last area to achieve gains in sound quality.  But what has happened over in the other thread (and some here, as they seem to have been linked) is that readers are getting the idea they are going to get "big gains" by doing this.  I am only trying to bring back some sense of scale and relative importance to the discussion.  For example: spending thousands of dollars on re-clocking a router or switch, or a Regen type device is going to be wasted money when much bigger gains could be had to the audiophile who is using a lower level DAC.  People with more moderate DACslike an Ayre Codex or Mytek Brooklyn would eb better served by first putting that money into a better DAC first (even more so better speakers if they do not already have Vivid Audio Giyas or something similar).

Maybe my perceptions of the value of "upstream tweaks" is somewhat ameliorated by the USB source I use;), but I am not going there fully as this is not a Sonore thread.  I do value some upstream tweaks in my own system, like the custom built ultra low noise power supply I built for my router, which we will be using at RMAF, but if you read this thread there are some folks here who fully do not understand even what kind of influence upstream clocks can have (noise as you and I agree, and not better data timing in the direct sense). 

 

In addition my posts are not meant to be any criticism of the Mutec components, for their intended purpose I expect they are excellent, and the clock itself, if it actually meets the claimed specs is excellent for its intended purposes (clock distribution necessary in Pro audio environments), and I agree the price appears to be quite fair as well for a non asian origin product.

 

Barrows, first of all, much respect to you and Sonore.  I am well aware of the great things you are capable of and having met you, Adrian, and Andrew at RMAF last year, and as a proud owner of a microRendu, know that I hold Sonore in high regard.  Having partnered with John Swenson, you guys have some serious IQ over at Sonore and it shows.  Contrary to how things might be perceived, if I have a bias, as an American, it is to see a small American company like Sonore succeed and succeeding, you guys are.

 

Having said that, if my effusive praise of the REF10 seems disproportionate to your perceived value of it relative to other devices in a digital chain, I want to remind you that I am posting in the REF10 thread.  If there is a place to be able to speak openly and candidly about the REF10, this is that place.  Having clearly stated what DAC I use, people can make up their own mind about how it might translate in their own system.  I would like to think that anyone reading this thread who can afford a REF10 and might be interested in a REF10 isn't going to be a fledgling audiophile and isn't going to use it to reclock their iPhone.  Absolutely, the REF10 is not the first component anyone should buy but nonetheless, this is one remarkable piece of kit.  There are many who jump onto this thread and make comments of what's possible or not possible based on theoretical grounds and have never actually heard the REF10.  The perspective that I offer is that of an actual REF10 user.

 

While it appears Sonore has not yet substantiated the impact of clocking in these "spaghetti" devices, I would encourage you guys to give it a go.  Where there's smoke, there's usually fire, and so I believe you will find that the many who have gone down this path are not just listening to placebo.

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11 minutes ago, barrows said:

This is a exactly what John S. is doing.  Hopefully his research will be able to determine the mechanism for any improvements, and then audiophiles can move forward in an informed way.  Of course we are well aware of the value of a better clock in our Ethernet Renderers, we first started experimenting with that long ago, which led to the development (among other things) of the ultraRendu and Signature Rendu SE.

But, if I am going to use clocks as accurate as the Mutec claims to be, they are going in my DAC first as a replacement for the audio clock(s) where they can do the most good.  Of course audio clock frequencies will not have as low phase at low frequencies as a 10 MHz clock can (phase noise scales with clock frequency all other things remaining equal) but using a fixed frequency clock at audio frequency will perform better than a 10 MHz reference clock used to generate (digitally) audio frequency clocks.  There are now some pretty nice audio frequency clocks available at semi-reasonable prices.

 

This is what is so challenging with digital audio.  No one truly understands it all.  What I respect about people like John Swenson and Rob Watts is that they fully acknowledge this and that some phenomena are inexplicable but just because certain observations are inexplicable, does't make them invalid.  Look how routinely Sonore and Uptone Audio get bashed on ASR based on lack of measurements.  It seems some people, purely based on theories or measurements, already know how somethings sounds even before the needle hits the groove.

 

With regards to the clock in a DAC, this appears to be a very complex topic and more than I am willing to tackle with my DAVE.  Here is what Rob Watts has to say:

 

"The issue of clocks is actually very complex, way more of a problem then in simply installing femto clocks. People always want a simple answer to problems even if the problem is multi-dimensional and complex. I will give you a some examples of the complexities of this issue.

 

Some years back a femto clock became available, and I was very excited about using it as it had a third of the cycle to cycle jitter of the crystal oscillators we were using. So I plugged it in, and listened to it. Unexpectedly, it sounded brighter and harder - completely the opposite of all the times I have listened to lower jitter. When you lower jitter levels in the master clock, it sounds smoother and warmer and more natural.

 

So I did some careful measurements, and I could see some problems.

 

The noise floor was OK, the same as before, and all the usual measurements were the same. But you could see more fringing on the fundamental, and this was quite apparent. Now when you do a FFT of say a 1 kHz sine wave, in an ideal world you would see the tone at 1 kHz and each frequency bucket away the output would be the systems noise floor. That is, you get a sharp single line representing the tone. But with a real FFT, you get smearing of the tone, and this is due to the windowing function employed by the FFT and jitter problems within the ADC, so instead of a single line you get a number of lines with the edges tailing of into the noise. This is known as side lobes or fringing. Now one normally calibrates the FFT and the instrument so you know what the ideal should be. Now with a DAC that has low frequency jitter, you get more fringing. Now I have spent many years on jitter and eliminating the effects of it on sound quality, and I know that fringing is highly audible, as I have done many listening tests on it. What is curious, is that it sounds exactly like noise floor modulation - so reduce fringing is the same as reducing noise floor modulation - they both subjectively sound smoother and darker with less edge and hardness.

 

So a clock that had lower cycle to cycle jitter actually had much worse low frequency jitter, and it was the low frequency jitter that was causing the problem and this had serious sound quality consequences. So a simple headline statement of low jitter is meaningless. But actually the problem is very much more complex than this.

 

What is poorly understood is that DAC architectures can tolerate vastly different levels of master clock jitter, and this is way more important than the headline oscillator jitter number. I will give you a few examples:

 

1. DAC structure makes a big difference. I had a silicon chip design I was working on some years back. When you determine the jitter sensitivity you can specify this - so I get a number of incoming jitter, and a number for the OP THD and noise that is needed. So initially we were working with 4pS jitter, and 120dB THD and noise. No problem, the architecture met this requirement as you can create models to run simulations to show what the jitter will do - and you can run the model so onlyjitter is changed, nothing else. But then the requirements got changed to 15 pS jitter. Again, no problem, I simply redesigned the DAC and then achieved these numbers. So its easy to change the sensitivity by a factor of 4 just by design of the DAC itself - something that audio designers using chips can't do.

 

2. DAC type has a profound effect on performance. The most sensitive is regular DSD or PDM, where jitter is modulation dependent, and you get pattern noise from the noise shaper degrading the output noise, plus distortion from jitter. R2R DAC's are very sensitive as they create noise floor modulation from jitter proportionate to the rate of change of signal (plus other problems due to the slow speed of switching elements). I was very concerned about these issues, and its one reason I invented pulse array, as the benefit of pulse array is that the error from jitter is only a fixed noise (using random jitter source with no low frequency problems). Now a fixed noise is subjectively unimportant - it does not interfere with the brains ability to decode music. Its when errors are signal dependent that the problems of perception start, and with pulse array I only get a fixed noise - and I know this for a fact due to simulation and measurements.

 

3. The DAC degrades clock jitter. What is not appreciated is that master clock jitter is only the start of the problem. When a clock goes through logic elements, (buffers level shifters, clock trees gates and flip-flops plus problem of induced noise) every stage adds more jitter. As a rough rule of thumb a logic element adds 1 pS of more jitter. So a clock input of 1pS will degrade through the device to be effectively 4 pS once it has gone through these elements (this was the number from a device I worked on some years ago). So its the actual jitter on the DAC active elements that is important not the clock starting jitter.

 

The benefit I have with Pulse Array is that the jitter has no sound quality degrading consequences - unlike all other architectures - as it creates no distortion or noise floor modulation. Because the clock is very close to the active elements (only one logic level away), the jitter degradation is minimal and there are no skirting issues at all. This has been confirmed with simulation and measurement - its a fixed noise, and by eliminating the clock jitter (I have a special way of doing this) noise only improves by a negligible 0.5 dB (127 dB to 127.5 dB).

 

This is true of all pulse array DAC's even the simpler 4e ones. In short the jitter problem was solved many years ago, but I don't bleat on about it as its not an issue and because it's way too complex a subject to easily discuss.

 

Pulse Array is a constant switching scheme - that is it always switches at exactly the same rate irrespective of the data, unlike DSD, R2R, or current source DAC's. This means that errors due to switching activity and jitter are not signal dependent, and so is innately immune from jitter creating distortion and noise floor modulation and any other signal related errors. The only other DAC that is constant switching activity is switched capacitor topology, but this has gain proportionate to absolute clock frequency - so it still has clock problems.

 

I plan to publish more detailed analysis of this, but from memory all of my DAC's have a negligible 0.5dB degradation due to master clock jitter, so its a non issue.

 

And yes you are correct, the absolute frequency is quite unimportant, so forget oven clocks, atomic clocks etc. Also the clock must be physically close to the active elements,with dedicated stripline PCB routing with proper termination. Running the clock externally is a crazy thing to do, as you are simply adding more jitter and noise and an extra PLL in the system."

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9 minutes ago, vortecjr said:

Barrows has already touched on what is taking place behind the scenes with John. While he is doing this on his own he is sharing the information with us. I also have something planned, but it has to wait until I have time. Anyway, I want to add that John is having to build very sensitive equipment just to prove his hypothesis. If it's this hard to measure then you have to really have faith that it matters as much as you think it does. Whatever is going on, if anything, is not at all obvious while other simple tweaks are. So what we are proposing is that people take care of the obvious things first since we can measure and substantiate them. Hopefully John finds something worth waiving the flag about.  

 

Jesus, fair enough.  Any measurements John comes up with will be trusted as properly done, however, whether they explain what I am hearing or not is irrelevant.  As a proud owner of your products, I will remind you of threads such as these and the conclusions they have drawn based on their measurements:

 

https://audiosciencereview.com/forum/index.php?threads/hardware-review-and-measurements-of-sonore-microrendu-v1-4.1867/

 

https://audiosciencereview.com/forum/index.php?threads/measurements-of-sonore-microrendu-streamer.577/

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2 minutes ago, vortecjr said:

No one is saying that your observations are in valid. What we think is missing a basic understanding about what needs to be considered first and what matters the most. Barrows has been addressing some points of interest already. I only speak for Sonore, but we get blamed for everything and it's never your power supply, your speakers, your amp, or your ears:)  

 

As I have reported my findings, I have kept my power supply, speakers, amps and ears a constant.  I have never blamed you.  Not sure where that is coming from.

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5 hours ago, barrows said:

I would suggest that, perhaps, this already exists.

 

Your bias and your agenda are quite obvious.

 

5 hours ago, barrows said:

Anyway, i have never bashed a component made by anyone.  I have suggested that some approaches, which may be used by some manufacturers are of questionable value: like using an external clock instead of using a good one internally.  Hey, look, Rob Watts, of Chord (a designer I respect very much) said the same thing...

 

Barrows, really, enough is enough.

 

You should know that Rob Watts doesn't think highly of the microRendu at all but at least he is willing to admit he doesn't know everything.  Because of the galvanic isolation he has implemented in his DACs and because his pulse array DACs are inherently immune to jitter, he believes all sources sound the same with his DAVE and that the microRendu is no better than a Windows laptop.  Here is what Rob shared with me last year shortly after I bought my microRendu and I told him I was hearing an improvement:

 

"Hmm, I am somewhat bothered by the idea that the microRendu is better than a windows (it must be windows) lap-top on batteries, as there is absolutely no explanation for why that may be. But "you know nothing Jon Snow" is my favourite quote for good reason; it reminds me that there are limits to one's knowledge."


If you are going to come on to the REF10 thread and publicly question the value of an external clock like the REF10 based purely on your theories, at least have the decency to listen to it first.  No one questions the value of using a good internal clock but sometimes, the components that we buy don't have the best internal clocks and that there are external clocks like the REF10 that are better.  Your continued bleating about how using an external clock is a bad idea makes you look ignorant when those of us who have experienced the REF10 first hand clearly believe differently.  Please, enough with the insults.

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