Superclocks

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17 hours ago, bibo01 said:

Is it worth considering the LT3045 regulator for it?

ADM 7150 with good implementation is actually the better regulator for an XO.

 

17 hours ago, AudioBang said:

Just curious, is your DAC using an ES9018 Sabre or ES9038PRO?

 

On a marketing sheet, ESS states that the 9038 is pin compatible with previous versions [9018] and I was wondering if the 9018 could be swapped for a 9038 PRO without any software considerations....

Just me contemplating more Hi-Fi stupid things to do 

No, 9038 Pro has different requirements of the I/V stage, you can however direct swap in the ESS 9028 for the 9018, but you will need to change the control software as well.

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2 hours ago, AudioBang said:

Half of my post for whatever reason got edited out....

My point was that in my system, the masking of detail, blurring, compression at certain frequencies/transients etc., from ineffectiveness of interconnect cable shielding severely handicapped DAC comparisons and minimized the audible effect of replacing the stock timing source in the DAC with a higher performing [10dB lower noise] OCXO.

 

By opening XLR pin 1 [ground] at the receiving ends and inserting a rare earth metal EMI absorber between pin 1 and ground at the source end, everything opened up significantly.

Sounds like something is wrong with the design of your components or your interconnect.  With a true differential amplifier/receiver, and properly designed balanced cabling, no shields currents will matter to the differential signal pair, as that boise would be common mode and cancelled out.

Here is a great paper to study, which includes discussion of this, among other things.  (and I ma actually a Bybee believer in general)

The G Word.pdf

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7 minutes ago, Miska said:

 

In many cases I've seen, jitter performance of DAC output doesn't change much by changing the clock part. Because the problems are more in the PCB layout design and other parts rather than the clock module itself. There's not much joy about fancy clock if it is spoiled before it reaches the DAC chip pin... And even if it reaches the DAC chip pin unspoiled, the it can still be spoiled by noise coming to any other pin of the DAC chip.

 

So I'd recommend to stop looking at datasheet plots and start looking at jitter measurement output of the entire DAC device instead. Because the two don't often correlate much...

 

Jussi, the interesting thing is, swapping to a better specified clock has always worked for me in terms of subjective listening, with a clear improvement, but the J-test spectrum looks the same.  I suspect the J-test is insufficient to see what is really happening, after all the J-test was developed basically to to test (the rather poor) SPDIF interface performance, we need a better measurement.

My experience is with changing clocks with the same implementation as far as clock distribution to the DAC chip goes.

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9 hours ago, Miska said:

Problem in changing clocks is that it is hard to place new clock properly in regards to PCB layout. For example if a DAC is originally using the NDK clock part and then you would like to swap to Crystek, no way it is going to fit there. If you put any wires there, then you are certainly big time spoiling the PCB layout and likely adding a lot of jitter. So at least the performance needs to be always verified by measuring, to check that it doesn't actually get worse.

Yes, I totally agree with this.  My experience with swapping clocks is in using the same interface for each clock, with impedance controlled u.fl micro coax as the clock line (same length for every clock <6cm) and same ultra low noise/ultra low impedance power supply to each clock.  So, no unshielded flying clock leads etc.  And I have heard differences/improvements with XOs featuring lower phase noise at 10 Hz and below.  But with the J-test, the spectrum generally appears perfect, even with a relatively "standard" XO...  

For a Sonore project a few years back we had a batch of NDK clocks measured and sorted for phase noise by a guy here in the US who does this for a few companies, so we had a range of performances of the same model clocks to try.

I wish I had access to an analyzer which can measure the phase noise, it would be very interesting to measure XO phase noise at the input pin of the DAC (chip), but the expense of that tool is not justified at this point.

 

One other note, I have a pretty good idea what lower jitter (measured), and lower phase noise XOs sound like, as back when I worked at PS audio we did some work correlating lower jitter (measured), with listening tests.  From that experience, I can say that in some system contexts it is possible that, sometimes, a listener might actually prefer the sound of higher jitter-so I could certainly see someone doing a rather poor "clock mod", using a better XO, but resulting in higher jitter (due to implementation), and then claiming way better sound.

 

One thing I saw recently, on a not to be named audio review site, was a very expensive Ethernet switch a company was selling, which included an "upgraded" OCXO.  they showed pictures of the internals and I was appalled.  The clock signal from the new clock board added was carried on a single unshielded wire, for some distance, and then soldered to the input pad of the prior clock, yikes!  This is a commercial product being sold for a not insignificant upcharge over the basic switch, and it looked like a poor DIY implementation.  Of course the review said it sounded amazing and lifted many veils, etc....  And of course, i have always been wary of companies offering external clocks to improve performance for a DAC, while this can be necessary to sync multiple devices in a studio environment, it is not a good idea for home audio: beware of these companies folks.  Best way to implement a clock in a DAC is to have it and any re-clocking, directly adjacent to the DAC (chip), with proper PCB layout, and not use an external clock going through a wire and at least two connections, etc.

 

I can have the J-test measurements you suggest made with different XOs, but the AP is in Florida and I am in CO so it takes some time.  I may get the AP sent here if/when I get an even better clock for comparisons (Pulsar).  At that point maybe I can get the AP sent here for some testing...  But from prior work, I suspect I will see a clean J-test with (excluding some very low level power supply related spikes) every clock I try.

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13 hours ago, AudioBang said:

Deleted

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12 hours ago, AudioBang said:

Not sure if I want to rattle the cage yet with power cord discoveries.... 

I would be interested in discussing some of your mods including power cabling etc...  Perhaps in a dedicated thread.  I am open minded about such things, but also believe that we can often be fooled by listening tests alone, and we need to be somewhat vigilant in understanding technically what may be going on.

RE the balanced cabling, a lot of audiophile XLR cables are made "wrong", as per Mr. Putzeys paper I posted, you should have a twisted pair for pin 2 and 3, surrounded by a  braided shield which then connects to pin 1, and then pin 1 goes to chassis (which can also be considered shield).  Lots of audiophile XLR cables have three wires plus shield, and this construction would likely result in uneven sharing of ground currents on the actual signal wires (pin 2 and 3), some wires also have "drain wire" added which could also result in uneven noise currents on the signal wires.  I check my cables, and my Nordost cables (Frey) are correct, with a symmetrical construction and only the braided shield carrying ground currents.  I also make sure pin 1 of the XLR shell connects direct to chassis.

 

I suspect that all the audiophile "grounding" products we see are "working" because they compensate for incorrect engineering of cables and components which is common in audiophile products (or to correct for problems inherent in single ended, RCA, wired components).  If the components are true differential circuits and wired correctly with pin 1 direct to chassis and the circuit ground to chassis at one other point, and we use correctly designed XLR cables, then there would be no "need" or advantage to the specialty grounding blocks, etc.

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11 minutes ago, Miska said:

No reason to bother with such, because it would be irrelevant. Only relevant thing is what comes out of the DAC's analog output. Even if you have perfect clock in the clock input pin, but if you have USB packet ticking and a bit of extra thermal noise leaking to the reference voltage pin the entire result is still spoiled.

Well, i would say we disagree on this.  And checking for noise on the DAC's voltage reference is pretty easy, so easy to rule out as a source of problems.  USB packet noise is 8 kHz, I do not see it, there is pretty good isolation from that noise in this DAC with dedicated ADM 715x regs for each DAC Vin, and isolation between USB receiver and DAC I2S input.

BTW, I am not modding commercial DACs with alternate XOs, I am building DIY DACs and evaluating different XOs with the same implementation (for clock distribution).

As for "perfect" J-test results with regards to jitter spectrum, OK, perfect is in quotes!  But take a look at the Stereophile measurements of commercial DACs, these days a lot of them have virtually perfect results.  And a lot of these days have far from "perfect" XOs, or XO implementations.  That said, I still find audible differences going from, for example, a Crystek 575 or 957, to a NDK SDA series XO (not the SD series)(and depends on XO frequency, of course) in the same implementation.  Have not measured on the AP (yet as mentioned) but i highly suspect I would see virtually no difference on J-test.

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I'll be in CT over the holidays, I know many may not have time for audiophilia then (I might not!) but I would be interested in checking out your stuff, shoot me a PM if you would like.

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33 minutes ago, Miska said:

And there are other funny things, like for example ESS' wandering noise bumps around the noise floor likely coming from the ASRC. These are actually hard to catch because they are time-varying and very low level.

I use ESS chips but the ASRC is not running.  Also I reduce variables as much as possible by having same clock distribution when evaluating different clocks and there is only 1 clock running (on the isolated side) of the DAC for 44.1 base sample rates only.  My understanding is that with the 9028/9038 designs they have eliminated much of the low level artifacts you mention although trying to figure this out from what ESS actually makes public (or even in the datasheets) is difficult.

Also I have DSC-2 balanced in progress, but not fully implemented (yet), very interested in this approach and your design.

 

But of course, in general, you are entirely correct that there are many aspects of any design which can be screwing up results and leading to wrong conclusions.  But, I have heard too many differences which were obvious, repeatable, and without question occurring, which do not show up in a standard set of measurements taken at the analog outputs of a DAC to agree that one can learn everything just from that measurement.  Do you not hear differences in various HQPlayer filters which produce the "same" analog output (in the audible bandwidth) in standard measurements?  Just curious. 

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1 hour ago, Miska said:

I have no problem analyzing/measuring differences between HQPlayer filters and modulators...  It is not right/wrong, just different.

You have measured differences within the range which would be audible?  That is <20kHz and above -110 dB?  Really?  I mean, I get that an early rolloff filter could be considered a measurable/audible difference (although not theoretically for my hearing)-but what about minimum phase vs. linear, for example...

 

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2 minutes ago, mansr said:

Then again, so is making one that doesn't

That is what i mean.  HQPlayer has many filter options, some of which could be shown in measurements restricted to the audible bandwidth O(and level) to have differences and some not, yet many have their favorites. 

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My POV is that all "should" be measurable at the analog output of the DAC, and perhaps it is, if we just know exactly what measurements to use..  But I have heard too many very audible differences which do not show up in what is traditionally considered the audible elements of the traditional measurement sets: J-test, THD/noise, FR, IMD, etc.  once you hear these differences they are no longer un-hearable, so to speak.  I also do not believe in "magical" audio, but it does appear that we are in need of better measurements to describe all aspects of audio performance.

Interestingly, to me a least, is that Jussi prefers to see alias products at extremely low levels, where most engineers would be comfortable with filter designs with -120 dB suppression of filter alias artifacts, Jussi appears to prefer them to be much lower although clearly anything at -120 dB should not be audible in room (never mind that these artifacts are above 20 kHz as well).  I know Jussi to be a really, really smart guy, so I intrinsically accept that there is probably something to this, but I am at a loss as to how there would be a difference between an alias at -120 dB and -160 dB?

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2 minutes ago, Miska said:

 

I don't want to specify what figures are "audible", I don't have clear answer for that. Not for clock jitter/phase noise either. I just know what is measurable or detectable in digital domain analysis. For example filter transition for RedBook begins somewhere below 22.05 kHz when it begins to roll off from 0 dBFS. This is certainly measurable in frequency response. Minimum phase vs linear phase is certainly measurable in the phase response too.

 

of course I understand that, but is it audible? certainly a measurement showing the rolloff above even at 19 kHz is not audible, But still people hear the difference in filters-I would conclude that something else about the filter's response is what we hear, but we are not measuring that thing.  This is the point which I was trying to make.  Same thing with linear phase vs. minimum phase, the phase shift at a very high frequency which we can measure, is also not considered audible by humans (unlike low frequency phase shifts), but still people report differences in sonics of these two filter types (overly simplified here, as we can have intermediate phase filters as well).  We can measure the pre-ring/post ring distribution with an impulse response test, but is that ringing audible?  Very questionable that it is.

I have seen posts where you advocate for very high attenuation of alias products, to levels many times below an audible threshold, indeed many times below what the analog circuitry is capable of delivering, is that audible: well it cannot be as far as what one measures, but perhaps there is some other aspect to it which is audible, otherwise, why do you advocate for it.

In no way am I suggesting you are wrong, and I have huge respect for your work and expertise with digital filters/oversampling, I am just trying to illustrate that the measurements we currently use are inadequate to describe all aspects of the sound of (in this case) a DAC, and that for similar reasons, perhaps the higher performing clock, not showing any difference in an analog domain J-test measurement, is still affecting some aspect of the sound.  I also am not trying to suggest that we cannot measure these things, just that we do not measure them all with the current set of measurements traditionally done-I think we need better measurements, I am really interested in what those might be.

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@miska

 

OK, I get that the measurements you use to develop filters are the necessary and applicable ones too show that the filter design is doing what you want from a technical perspective, and I have no doubt that many of your filters do sound better/different.  I do not have HQPlayer (yet, unfortunately) because I do not have a good machine to run it on.  But I have tried quite a few different approaches, and the differences are usually small, but meaningful in an audiophile sense.

I have a hard time saying that a measurement of something down at -120 dB is audible though, in room, through loudspeakers, given the real world dynamic range of systems, and I use Ncore amps with very low floors, and the DACs here are very good in this regard as well.  And I have hard time saying that a measurement of something above 20 kHz is  audible, given human hearing limitations.  Phase relationships at high frequencies, I do not know enough of the psychoacoustics of to know for sure, but my understanding is that it is not audible, I will do some reading on that to try and come to a better understanding.

Clearly you appear to believe that current measurements are enough, this is where we depart a bit.

So your take away is that humans, apparently, can hear things which most research says we cannot.  I certainly cannot hear a single tone at 20 kHz in room, but you are suggesting that perhaps I can hear something at that frequency and higher?  Hmm, I am aware of some research suggesting our bones conduct these frequencies and we may be able to "sense" them somehow, I ma to sure how that would be related to our perception of reproduced music...  Then there are those folks who claim to be sensitive to things like WiFi, bodily.  Of course a lot of those same people also claim to be able to "see" the invisible aliens walking among us, oops, I digress...

As to clocks, for sure I am no believer in external master clocks at 10 MHz (atomic or not) which them have to be carried on a wire to a component, and then converted by a DDS to the actual needed frequency.  Although the recent improvements in DDS does make this a little better proposition.  Give me an XO with -120 dBc/Hz phase noise at 10 Hz, at the audio frequency, right at the DAC and flip flops please.  And as I said before, i would like to have a phase noise analyzer to be able to measure that performance right at the input pin of the DAC to be sure.  But even without that, with clocks using the same implementation, I do hear the difference if a swap to an XO with a bit better (6 dBc/Hz or more) low frequency phase noise performance.  And, I consider this difference larger than what I hear via most different digital filters (but I need to try HQPlayer as well).

I do not see much difference in my advocacy for a better clock (lower low frequency phase noise) and yours for HQPlayer filters.  Both have measurements to show a difference, and both have the anecdotal evidence (which you rely on from your customers as mentioned).  Maybe I can get the AP here at some point, some XOs to test, and take a really close look at the 24 bit J-tests measurements.

As an aside, I know it takes a super computer to run HQPlayers single pass filters (I need to go to DSD 256 and even 512), and quite a bit less power to use the 2 stage filters, how much of the HQPlayer "goodness" can I get via the 2 stage filters?

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1 hour ago, Jking said:

The Mutec with Rubidium clock use to be popular on these pages.  

 

 

The problem with an external clock like this is two fold:  First you have to carry the clock signal on a wire, with at least two connections, and then some internal connection inside the component are virtually unavoidable as well.  So you get clock degradation there.  Then, you have the wrong frequency for an audio DAC, so the DAC then internally has to convert that 10 MHz frequency to one of the audio frequencies (like 22.5792, for example).  This is generally done using a DDS chip, this also adds more phase noise, so by the time your clock signal gets to the DAC ir has been degraded quite a bit.

It is much better to put a really good clock in the DAC, right at the DAC chip, at the correct frequency.

External reference clocks like the Mutec are designed for recording studio use where often multiple components need to be clocked together to a single reference, in a home audio playback system we do not have multiple components to clock together (OK there are some rare exceptions to this), so it is better to not use an external clock.

But do not take my word for this, ask @marce, or at @JohnSwenson.

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9 minutes ago, Miska said:

I personally have problem with metal dome tweeters that have resonance frequency somewhere between 22 and 28 kHz. Listening to such in long term is like listening to dentist ultrasonic teeth cleaning tool. If you know what I mean, that cleaning tool sounds just like telephone line modem handshake from the 80's-90's. Of course the tweeter is not as bad, but you get sort of headache kind of annoying feeling in back of your head. And it is much worse if the DAC has leaky filters. You often notice this at shows/fairs where you suddenly enter listening room and get a feeling of the sound in couple of first seconds before you even register what you are listening to.

This is a great observation.  Most metal domes are a problem for me as well, although I have heard some beryllium implementations which are very good, they appear to have their resonance higher up.  I have oldish soft revelator dome here.  I ma actually flirting with building some speakers, with the help of an established speaker designer, looking at the Seas soft domes, Crecendo, etc.

 

And of course I agree entirely on the implementation of the rest of the DAC, analog stage, filter implementation.  I am certainly not advocating for just changing the XO in a DAC with poor design overall!  I am advocating for better clock when the other aspects of the design are (near) "perfect".  I consider DAC analog stage design (including analog filter) the most important aspect, followed by power supply implementation (given we are not talking ancient DAC chips, etc), then digital circuitry and clocking.  I would only optimize the XO after the others are addressed.  I have gravitated away from leaky (digital) filters and minimum phase types lately, as DAC/system performance improves it appears I prefer apodizing filters and linear phase...  I wonder if a lot of folks who prefer slow roll-off filters may be experiencing a kind of masking effect (similar to dither?) which is, perhaps, covering up system "problems".

 

I am almost ready to start listening to my DSC-2 DAC in my main system, it will be an interesting contrast to my ESS 9038 based DAC, I will be feeding both DSD 256, and perhaps 512 later on.

[barrows]

Hi Alex, yeah, I know, I have to get going with HQPlayer eventually, it will happen-I have a DSC-2 version going here, which is pretty cool, but I still need to add an analog volume control (Muses chip option on the way), then I can listen to it in my set-up vs, the very formidable ESS 9038 DIY DAC I am using now.  I have also played around a lot with A+/Isotope filters, ending up preferring an apodizing, intermediate phase option, which rolled off a bit early, to allow a not too steep filter slope, while still reaching -125 or so at Nyquist.

Do you find the 2 stage filters in HQPlayer to be good enough?  I know the single pass ones require a supercomputer.  I need to be able to do DSD 64 to at least DSD 256, and all PCM rates to DSD 256.  Right now I am using ROON, and am very surprised how good it sounds going to DSD 256, considering the little processing power it uses (it does not break sweat doing this even on my I5 Mini!), but I expect HQPlayer to be significantly better.  My ESS DAC is set up for DSD 256 in terms of all the filter settings and the analog filter, and is running just a single 45.1584 XO in synchronous mode (no ASRC or DPLL active).

 

I expect the DSC-2 to be a bit more able to differentiate between OSF settings, considering it is just a discrete converter with no modulator operating, even with the minimum way I am using the ESS chip it is still running its (SDM) modulator.

 

Good for you getting out pedaling, I have been a bit lax in that area this summer, it is amazing how hard cycling is when your not getting out 4-5 days a week like I usually am.  Oh well, backcountry snowboard season is nearly here and I will be skinning uphill like a maniac soon!

 

Best!