In quest of budget reclocking

[wkhanna]

That is pretty close to what Jim was talking about.
As close to a 'one box' solution as we have seen so far.

[PewterTA]

I'd love to hear that!!!! Bill, we need to figure out how to make a bunch of money for all these toys!!!

[JonMarsh]

I'd love to hear that!!!! Bill, we need to figure out how to make a bunch of money for all these toys!!!

While it's not what I would call inexpensive, even with the power supply upgrade it's in the same general range as many upper midrange products, like the Berkeley Alpha DAC. Considering the feature set, if it delivers commiserate performance, it's reasonable value.

[JonMarsh]

Care package is on the way- should be there next Friday, vis UPS ground.

[JonMarsh]

The key point for me; some guys will get off on the DSD bit, but I'm a little skeptical about the "worthiness" of that feature, but who knows!

Note the bottom left input!

[impala454]

Just a quick newbie question. Where does the reclocking begin in a chain like this? I mean that fanciness is being plugged into a computer and I'm guessing nothing on that side is reclocked. Is it mainly all contained within that one component? Which items in the chain are being synchronized?

[JonMarsh]

Something to keep in mind- re-clocking of one kind or another happens pretty much in any digital chain one place or another.

In a CD player, there is a local crystal oscillator that works through a PLL loop to control the reading of data off of the CD and reclock it relative to the local crystal oscillator. A lot of those oscillators aren't all that great, so for example, one saw the release of a lot of clock kits or super clock kits for players in the late 90s and 2000's.

In an asynchronous USB setup, the concept is that a local clock at the receiver, which is either based on the 44.1 kHz sample rates or the 48 kHz sample rates, is used as the master clock to read in data from the PC; data is requested into a local buffer and re-clocked by the master oscillator before going to the DAC chip set, whatever it is.

Now, the older "adaptive" USB setup uses what ever clock is on the PC or other transmitting device, (it could be a music server, for example), and uses PLL's to generate a local clock that follows the incoming clock rate, and then reclocks that data into the DAC using the output of this PLL controlled clock. (PLL = Phase Locked Loop).

Ironically, though we don't like adaptive USB so much anymore, that is pretty much how S/PDIF and AES/EBU work- that is, the sending device generates a clock which is embedded in the data signal, and recovered at the receiver. Any level dependent variations in timing show up as jitter in the received clock- this is why optical is not so great from a clock viewpoint, as the drivers and receivers have more non-linearities, and generate more jitter. The thing about standard receiver chips is that they have no flexibility, have a very short FIFO buffer, and have to follow the jitter variations of the sending clock pretty quickly or they lose sync. The common methods in many modern DACs to deal with this is then to run the received signal into an SRC and let it deal with the frequency variations and try to filter the jitter. This they often do reasonably well (compared to digital solutions we had in the late 80's and early 90s) but not without adding some HF distortion. These issues with SRC are why there is a small but vocal body of people who prefer NOS DACs that don't do any SRD or oversampling.

In a professional environment, we use word clocks to synchronize the macro timing of all the digital components in the studio- but remember, the word clock is literally just synchronizing the word boundaries of the digital data to be sure each pieces is able to lock directly to a connected pieces without frequency drift or losing lock. Frequency drift in a connected digital system usually manifests in random "click" sounds which occur at lower or faster rates, depending on the degree of frequency drift between components.

Now, for optimum quality of the actual D/A process, we have to be concerned with what you could call the "micro" timing, not just the macro timing- in other words, how well defined each pulse for the digital clock controlling the actual D/A conversion process is, which are several orders of magnitude finer timing than the work clock- this is where the quality of the internal clock comes in, or where we can kick things up a notch further with an external rubidium clock. The latter are more stable and have less inherent noise jitter than crystal oscillators; in fact, you can use a rubidium clock to determine phase and frequency drift of a crystal oscillator just from the mechanical stress of picking up the box or board it's on and waving it around; the acceleration and deceleration causes very small frequency shifts. Putting the rubidium clock input right in the DAC is the best way to do it- all other things being equal, including the quality of the actual D/A process.

In our case, that is the experiments my friend and I have been doing since around 2008, we found that using the Brainstorm DCD8 to provide word clock synchronization from the music server output (Lynx AES16 board) to the WCLK input of the DAC (Antelop audio Zodiac Gold, originally), and using the reclocking function of the DCD8, where the digital signal is fully reclocked to very precise timing not just for the WCLK boundaries but for each pulse in the AES/EBU output (go to wikipedia for an overview) before being routed into the Zodiac gold (on AES/EBU) this improved the sound significantly in terms of the articulation and definition of complex transients, and in terms of imaging characteristics; having a more natural acoustic perspective with more depth and more precise instrument placement, even though the room has no acoustic treatments at all. So, to summarize, three pieces of equipment were being synchronized on WCLK boundaries (normal studio practice) but also the full digital signal was being completely re-clocked by the DCD-8. The DCD8 is capable of doing format conversion as well as reclocking; you can take in Firewire or S/PDIF and output AES/EBU, for example, or S/PDIF from AES or Firewire input. Other units, like the Antelope Isochrone DA (which I have) can do some of the clock stuff, but don't do full reclocking, and don't have the same sonic benefits to our ears (though it does have some- will make a Berkeley Alpha DAC smoother and more natural in the high frequencies- can't say I know why, but I know what we and others have heard.

Now I need to investigate if the Mutec can do the same things with the same results.

[impala454]

Very interesting. I deal with synchronization of data packets all the time (data typically in the 50-300Hz range) and it's a pain enough to get that done in software. I can't fathom all the wizardry it takes to get this stuff right in a D/A. Very cool. Thanks for the explanation.

[JonMarsh]

Very interesting. I deal with synchronization of data packets all the time (data typically in the 50-300Hz range) and it's a pain enough to get that done in software. I can't fathom all the wizardry it takes to get this stuff right in a D/A. Very cool. Thanks for the explanation.

Cool- glad if it helps!


Dan and Bill,

the 2nd LPFRS I ordered arrived this weekend as I found out checking my mail on Sunday. Will try to setup and test Wednesday evening.

[JonMarsh]

I had another idea this morning- since I have a DCD8 here anyway, and only one system at the moment that i'll be doing reclocking in, once I test the Mutec and give it some kind of thumbs up or down, there's no reason I couldn't send it out to you guys to try out and let you see if it makes a difference in your system or not. I would not be surprised if there is some dependency on the performance and design characteristics of the associated DAC. I'll check it out here with the TotalDAC D1 dual, the NAD M51, and the Berkeley Alpha DAC.

Just something to think about- lower the "risk" for you, though if it does work OK and you hear it, then I guess the "risk" will go up- at least for Bill! But jewelry often solves that kind of problem... :W

[wkhanna]

Ha!
One Mutec for me......one Mikimoto necklace for The Wife......
No wonder this hobby is so much more expensive for us married guys!

[wkhanna]

Dear Maestro,

I came across this question & reply at another forum.

Could you please decipher the reply that I can better understand what author, Larry Ho from Light Harmonic Labs is stating?




Question: Would like to know, if I have an asynchronous USB DAC, will a good USB to SPDIF converter still reap rewards? What I mean is, is it any
benefit to convert to spdif prior to the DAC so I can use the DACS spdif input rather than the USB?


Reply: Given the other conditions are the same, asynchronous USB transmission is better the USB-to-SPDIF, due to the fact that master clock location is better, jitter removing is easier and without SPDIF, DPLL won't be needed.

[JonMarsh]

That all depends.... at this point, it becomes a contest between the the quality of the local clocks in either the DAC or the USB to S/PDIF converter. USUALLY the answer provided should be correct, but it's up to the quality of implementation in the DAC, versus the external device. Also, I'm not particularly a S/PDIF fan, anyway.

For example, others have reported better results with fairly high end DACs from Audio Research and Bryston using a Berkeley Alpha USB interface instead of the internal DAC USB interface, which is also supposed to be asynchronous.

The guy who makes the TotalDAC says the reference design for XMOS for the USB asynchronous interface is flawed and doesn't sound all that good- he put a lot of work into upgrading it- but he also recognizes that special configurations such as my friend uses (with the Brainstorm DCD-8 and external reclocking of AES-EBU can sound better.

[JonMarsh]

Ha!
One Mutec for me......one Mikimoto necklace for The Wife......
No wonder this hobby is so much more expensive for us married guys!

Yup! That's pretty much how it works, I hear... :W

[wkhanna]

That all depends....

The more reading I do the more I see there is no definitive answer to what works best.
As is so often the case, even with analog systems, what components & exactly how you integrate & connect them is nearly infinitely variable as are the results.

One thing I have noticed, most feel it is necessary to have the master clock V close to the DAC chip.
But, unless you are building your own DAC from scratch this is hardly doable.

At the same time, most seem to want to POO POO the idea of putting a V accurate clock in series, claiming worries of EMF/EMI interference & induced jitter during transmission.

It is this general protest that seems to be stigmatized for some reason, as V few appear to have actual evidence as to its efficacy.

[JonMarsh]

The more reading I do the more I see there is no definitive answer to what works best.
As is so often the case, even with analog systems, what components & exactly how you integrate & connect them is nearly infinitely variable as are the results.

One thing I have noticed, most feel it is necessary to have the master clock V close to the DAC chip.
But, unless you are building your own DAC from scratch this is hardly doable.

At the same time, most seem to want to POO POO the idea of putting a V accurate clock in series, claiming worries of EMF/EMI interference & induced jitter during transmission.

It is this general protest that seems to be stigmatized for some reason, as V few appear to have actual evidence as to its efficacy.

People often pooh pooh things they haven't tried or don't want to try. I have been duly skeptical about a lot of possible configurations, as well as my friend's wife has been. (She's a pediatric doctor as well as a music lover, and no dummy, and has a good ear I'd say, for not being a musician. )


Sony engineers pooh-poohed adding higher grade clock components to their first gen statement class SACD players (SCD-1, SCD777ES), but those could clearly make a difference.


Let's just put it this way- for the DAC's we're using, it makes a marked audible difference. Now, the DAC's we're using (TotalDAC D1 Dual (mine; balanced DAC driving balanced outputs) and TotalDAC-D1 Dual Mono (each DAC operating only on one channel signal- my friend's) are pretty fair, and in direct shoot out were chosen for overall reproduction quality over the Meitner DAC2x, the dCS Debussy, and the MSB Platinum DAC IV Plus with the Signature Power base. I had gone into that figuring the Meitner or the dCS was going to be the likely winner; the TotalDAC-D1 was something of a quirky long shot that my colleague had a chance to get on home demo at the same time.

I hope this establishes a reasonable baseline for comparison of performance. But the performance is improved with the Brainstorm DCD-8 in the digital path for the TotalDAC D1, just as it was for the Antelope Zodiac Gold.

Now, I'm not selling anything- not in DAC's, or DAC upgrades, or speakers, or ANYTHING. I could give a rat's bun's or a Honey Badger's tail if someone else thinks this can work or not, especially if they haven't actually tried it, much less trying it in the configurations we're using. I agree that the best situation SHOULD be with the governing clock being local and close by. But guess what? First, the only way you're actually using the local clock with S/PDIF or AES/EBU is if you're either doing SRC to a different local frequency OR you are using a large input buffer to load each track into and clock out at the master rate (which the TotalDAC does, and which the PS Audio PWD Mk II does, but almost no other DAC does), OR, like most, you have a PLL receiver and you are moving your local clock around to follow the input signal (yes, most S/PDIF receivers are adaptive, with a very fast response time because they have no buffer to speak of!) and then you have to do SRC to get it to synchronize after a fashion with the local clock. SRC is done either by off the shelf chips from Cirrus logic and TI, or in the better cases, by custom routines running on a powerful local DSP processor. (Berkeley Alpha DAC and many others)

Asynchronous USB or Firewire should be better, in principle; in the case of Weiss and Metric Halo, it's very good for Firewire; in the case of USB, the details are in the implementation and we still seem to have a problematic situation, in that many DAC reviews of high midrange products report better sound on the AES/EBU input, which should be technically inferior... by the common wisdom. But isn't, when you listen to it.


It's very easy to send a 10MHz sine wave around without signal degradation issues- easier than a conventional digital signal. And it's just remotely possible that the quality of transmission components used in units like the DCD-8 or the big Antelope pieces might be just a skosh better than in the typical boutique audiophile DAC or source components.

I mean, just putting an Antelope DA in the signal path between a digital source unit and a Berkeley Alpha DAC improves the high frequency clarity and tones down the zing that unit often has with some source components- I sure can't explain that, other than a buffering and EMI effect. Several others have reported that effect.

I believe that the more stable the incoming digital signal, the less likely there are to be any artifacts in sample rate conversion (AKA "up sampling" ) done inside the DAC using it's local oscillators as the reference. Remember, the majority of these DAC's take all incoming signals and SRC them to one frequency, so we're not really talking about up sampling per se. I'm still inclined to believe that NOT doing that kind of up sampling or sample rate conversion (at the quality level possible with off the shelf chips, as opposed to say, the quality level possible with 64 bit Izotope code) and working with the native rate at all times is better, but then, hey, that's just my current religious belief about that based on NOTHING other than LISTENING experience, with DAC s that support that, versus ones that don't.

YMMV, Blah blah blah blah.

In the end it's all about the music reproduction. I'm sure there are alternative system configurations with out re-clocking like this that can sound very good, or put that reclocker inside, like that new Antelope Platinum DAC, perhaps. However, the Platinum is designed for DSD, which means it's likely doing multi-bit delta sigma conversion for PCM inputs as well, which is a lot of signal processing and manipulation that I'm no longer convinced is a great idea, compared with a NOS R2/R ladder DAC operating at native rate. So I'd certainly be willing to give one a listen, given the opportunity, but I know there are issues I'd be on the look out about.

The one high end DAC I haven't heard side by side against the total DAC is a Scarlatti dCS *$23,999. I'd especially want to hear it with their optional $9,999 master clock. But the next spare $35K I have will be applied to a Model S Tesla, not a DAC likely to give only small incremental improvements over what I'm using now. Oh, BTW, you'll need the upsampler module for the dS Scarlatti if you want to have a USB input; that will set you back another $12,999. Hmm, guess that's looking more like $50K by the time you pay the tax bill....

OK, throwing in my re-clocking setup, I paid for my DAC less than the equivalent of a Honda FIT, NOT the equivalent of a BMW X5. I think that says it all...

[wkhanna]

....I believe that the more stable the incoming digital signal, the less likely there are to be any artifacts in sample rate conversion (AKA "up sampling" ) done inside the DAC using it's local oscillators as the reference. Remember, the majority of these DAC's take all incoming signals and SRC them to one frequency, so we're not really talking about up sampling per se. I'm still inclined to believe that NOT doing that kind of up sampling or sample rate conversion (at the quality level possible with off the shelf chips, as opposed to say, the quality level possible with 64 bit Izotope code) and working with the native rate at all times is better, but then, hey, that's just my current religious belief about that based on NOTHING other than LISTENING experience, with DAC s that support that, versus ones that don't.

YMMV, Blah blah blah blah.

From my meager technical understanding of what DAC's are doing based on the reading I've done, I always liked the idea of non-upsampling (keeping the data in its native form). Now that I'm learning about how clocking is functioning in the system, it seems intuitive that the better timed the data is going into the DAC, the better (less work) for the SCR's in the DAC. And if people are reporting good results just using the OEM clock in the Mutec, then adding the 10MHz Rubidium reference would likely only make things better. Again, this is all assumption on my part, no empirical data. However, the availably & affordability of a surplus Rubidium clock is information few are aware of. And even fewer are probably willing to experiment with at this level.

In the end it's all about the music reproduction.

Once we have the components assembled, we will be testing this out on Dan's system & mine, giving at least two more data points on how other DACs react to the improved timing. Come to think of it, I have two, a Schiit Bifrost & Gungnir, along with friends with DAC's.

[PewterTA]

I love playing and testing new toys!

Jon, the best part of this thread is the fact of how you so elegantly phrase out what is basically in my head. You've got more knowledge than I do, but your thinking on everything is exactly where I (and frankly what I've done for Bill) to try to get the best out of our systems.

Now that we have some of your guidance, it's a little easier for us to piece together what we can afford to do and in what direction we want to head.

All your knowledge and testing has really gotten myself and I can tell Bill excited in furthering out Digital Output and getting the best possible reproduction.

I think Bill and I need to have another get together and talk about what we need to get and when we can get things.

[PewterTA]

oh and we would love to play with a DCD8! That's extremely kind of you!!!!! I don't think I could say, "No we couldn't possibly," because I think Bill and I would love to see what it can do in our system!

That's really where we've had the trouble... we know things are better, but we've also learned that no necessarily does "better" really equate to "better in our systems."

[jim1961]

http://www.digitalaudioreview.net/2011/08/rega-24192-dac-2-x-wolfson-wm8742/

Turning the story in on itself, the Rega DAC can also act as its own digital re-clocker; its coaxial digital output could feed another decoder down the line.

For the Rega DAC, it this feature related to what you are doing Jon?

[wkhanna]

After reading the review, I find myself quite confused.
Is this coaxial digital output from the Music Fidelity DAC the digital signal prior to conversion to analog but after SRC?
This just makes no sense at all to me, so I must be missing some fundamental understanding of what is going on.

[jim1961]

http://www.rega.co.uk/downloads/Dac%20manual.pdf

This is from the manual:

DIGITAL OUTPUT
The digital output mirrors the selected input. For example when input 2 has
been selected the digital output will be the signal present on input 2 but cleaned
up and re-clocked, this can be used to drive an auxiliary piece of equipment if
so desired.

[JonMarsh]

From reading several pieces about the Rega DAC it sounds like it doesn't use SRC (sample rate conversion) but works at the native rate of the incoming signal. Given the reported differences in audible performance between different digital front ends, and that reviewers preference for the Audiophileo, it is "responsive" or "sensitive" to incoming digital quality and timing. From the description on Rega's own site, they use a Wolfson receiver chip with a PLL style local clock to lock on to the incoming signal; if the coax output is reclocked, it would be done, I suppose, with the clock generated from the local PLL (Phase Lock Loop).

We decided not to use a sample rate converter and process the data at the incoming sample rate which keeps the signal processing to a minimum. Jitter was minimised by synchronously clocking the digital data with our receiver PLL (removing any jitter from the input signal).

This is normal, to use the local PLL clock (which is derived from the incoming clock recovered from the data signal and must be locked to it), but unless they have a really large input buffer, (not normally the case, and certainly not in a product in this price range), the PLL must move pretty quickly at times to stay locked to the incoming signal, and this movement in frequency, is guess what boys and girls, "jitter".

The Stereophile review comment is that the adaptive USB digital input is pretty well sorted for that type

From what I've heard in reviews, the Rega performs well sonically in it's class ($1K DAC's) and is often, with a good digital front end, compared with $2K CD players. But $2K CD players are not compared with the NAD M51...

It's not fair for me to bring that up, but when a manufacturer says a unit performs above it's class, my usual reaction is to question, how much?

Let's just look at 16bit and 24 bit noise, and compare the Rega and NAD M51. (I feel mean spirited to even be bringing this up, but the Rega gets a lot of mention here at HT Guide).








24 bit -90 1 kHz sine wave








11 kHz jitter spectrum

Rega


high-resolution jitter spectrum of analog output signal, 11.025kHz at –6dBFS, sampled at 44.1kHz with LSB toggled at 229Hz: 16-bit data via 15' TosLink S/PDIF from AP SYS2722 (left channel cyan, right magenta), 24-bit data (left blue, right red). Center frequency of trace, 11.025kHz; frequency range, ±3.5kHz.

The Rega's performance via its USB input was identical to that from the S/PDIF inputs, other than the restricted bit depth and sample rate. This was true also of the DAC's jitter spectrum with USB data (fig.14), which was marred by the same supply-related sidebands around the 11.025 kHz test frequency.



NAD



NAD M51, high-resolution jitter spectrum of analog output signal, 11.025kHz at –6dBFS, sampled at 44.1kHz with LSB toggled at 229Hz: 16-bit data via TosLink from AP SYS2722 (left channel blue, right magenta), 24-bit data (left cyan, right red). Center frequency of trace, 11.025kHz; frequency range, ±3.5kHz.


Given the similarity of the jitter performance of the Rega with all inputs and the presence of the supply related sidebands in all cases, the first thing I'd check on the design is if there are issues with the digital supply or grounding layout that could be inducing low frequency supply related jitter.

[wkhanna]

Thanks, guys!
That makes it a bit clearer..........

But honestly, I am still trying to figure out what the heck your supposed to do with the 'digital Output' from the Music Fidelity after it has been 're-jittered' by the substandard PPL?

I guess they figure you can then feed it to another MF DAC?

Honestly, I still don't get it! :banghead:

Please tell me this is all just marketing mumbo jumbo, and this 'feature' is basically of little practical use.

[JonMarsh]

Thanks, guys!
That makes it a bit clearer..........

But honestly, I am still trying to figure out what the heck your supposed to do with the 'digital Output' from the Music Fidelity after it has been 're-jittered' by the substandard PPL?

I guess they figure you can then feed it to another MF DAC?

Honestly, I still don't get it! :banghead:

Please tell me this is all just marketing mumbo jumbo, and this 'feature' is basically of little practical use.

Well, the Antelope Audio Zodiac Gold has a similar feature, and maybe if you wanted to make the sane signal available in another room this would be useful? But the Zodiac Gold has a much more sophisticated Antelope clock system, and does AES/EBU re clocking, too.

Still, it's not a feature I've missed never having...

[jim1961]

Jon. I am not trying to tout the performance of the Rega DAC nor am I trying to compare it to a M51. I was merely trying to see if running a digital signal through its digital in/out to still another DAC would serve any purpose.

[wkhanna]

Nor was my intent to dis the Rega (which I mistakenly & repeatedly referred to as a Music Fidelity, sorry).
Like you, I was only trying to understand what is to be gained by using the digital output.
In fact, this Rega DAC was on my short list when I was shopping for my last DAC.

[JonMarsh]

I don't see it as anything other than a possible convenience factor, if you need the signal routed somewhere additional. But I've never encountered a situation where I ever needed that myself.

And the supply harmonic related sidebands in its analog output raise questions in my mind about the reclocked digital output integrity- if that is due to internal interactions with the clock or other digital processing with the power supply, it's possible that the passed on signal might not be as clean as that from a good source like the audiophileo or something like a Brainstorm DCD8.

It's not my intent to pick on the Rega, but I don't want to gloss over things, either.

[jim1961]

Nor was my intent to dis the Rega (which I mistakenly & repeatedly referred to as a Music Fidelity, sorry).
Like you, I was only trying to understand what is to be gained by using the digital output.

Taking what Rega said literally, it takes the signal in, cleans it up, and sends it back out. The question is, does it really do this or how well does it do it. (again, I am not trying to steer things towards talking about the Rega, but I am thinking that the Rega DAC isnt the only one with this in/out feature. it would benefit us all to know what they are talking about I think).

[JonMarsh]

Jon. I am not trying to tout the performance of the Rega DAC nor am I trying to compare it to a M51. I was merely trying to see if running a digital signal through its digital in/out to still another DAC would serve any purpose.

I don't really think so... Other than providing a pass through for convenience to another piece of gear.

[jim1961]

I don't see it as anything other than a possible convenience factor, if you need the signal routed somewhere additional. But I've never encountered a situation where I ever needed that myself.

And the supply harmonic related sidebands in its analog output raise questions in my mind about the reclocked digital output integrity- if that is due to internal interactions with the clock or other digital processing with the power supply, it's possible that the passed on signal might not be as clean as that from a good source like the audiophileo or something like a Brainstorm DCD8.

It's not my intent to pick on the Rega, but I don't want to gloss over things, either.

Are you referring to those spikes at -110db?

[JonMarsh]

Are you referring to those spikes at -110db?

Yes...

This....



(the widely spaced magenta spikes are normal, the sideband spikes around the 11.025 kHz and any widening at the base are artifacts of jitter)


Compared with this for the M51



Now, one might figure that those spikes down at -110 shouldn't make any difference, but this is with a single sine wave; put in butanes of them and they all have spikes, and the inter mod products have spikes, etc. The odd thing is that with program we still hear things that don't sound right about digital, even though listening through speakers or amps (tube, for instance) with high levels of conventional harmonic distortion.

Be sure when visually comparing these that you remember there's a 12 dB or so difference in the noise floor, and so compare what the Rega looks like at -135 to the NAD at the same level, not at the bottom of it's plot.

But frankly, is something I suggest everyone listen for themselves and decide what they think is OK. But don't calibrate your ears just with other digital systems; go listen to a good vinyl setup or a high end tape setup (open reel for high end is making a minor comeback... for good reason- you can beat it with digital, but not at low price points. OF course take isn't cheap either... but 15 IPS half track can sound pretty dang good...)

[JonMarsh]

I can easily imagine there's more than a small amount of incredulity about some of the things we're discussing and proposing. Particularly this thing about how important the clock and clock recovery without SRC and what not may be to the reproduced sound quality. To that end, I'd like to quote some bits from the Computer Audiophile review of the PS Audio Perfect Wave DAC II.

The PWD II has a critical feature to it's performance, which apparently, according to the review, isn't even turned on by default. It's quite similar to one of the measures taken in the TotalDAC D1; that is, the use of a relatively long deep buffer which allows using a slow PLL and pushing the frequency for which the PLL must track the incoming signal to a much lower cut off point.

The new built-in Digital Lens is the single biggest improvement PS Audio has made to the MKII. The digital lens improves sound quality in the MKII so dramatically that I consider the MKI product broken. Honestly I can't imagine listening to a PerfectWave DAC without sending every bit of audio through the sharp focus of the Digital Lens. The Lens works with all inputs of the PWD MKII even the network Bridge that contains its own version of the Lens. The Digital Lens receives all digital audio signals just before the actual DAC chip and regenerates a new reduced jitter signal for conversion by the DAC.

According to PS Audio the Lens accepts any amount of incoming jitter and reduces it to below 1 picosecond. I believe I've heard similar statements from nearly every DAC manufacturer about how jitter is reduced to below a picosecond by one component or another inside their DACs. Marketing for sure, but they have a job to do like everyone else. PS Audio's implementation of the Digital Lens in the PWD MKII is called NativeX and it's selectable via the front panel touchscreen or the remote control. NativeX shouldn't be confused with Native or any of the Sample Rate Conversion (SRC) options available on the MKII. Like the MKI the MKII offers users the ability to convert sample rates up or down to a desired frequency or leave the audio untouched in its native format. Two options leave the audio untouched, Native and NativeX. Native is available on both the MKI and MKII. This mode is pretty straight forward as it simply send the exact data received on the digital input to the DAC chip. Sending CD quality audio at 44.1 kHz to the PWD on any of its inputs using the Native mode bypasses all manipulation and allows the 44. kHz data to reach the DAC chip. The NativeX mode, available only on the PWD MKII, takes advantage of the new built-in Digital Lens. Think of it this way, sending that same CD quality audio at 44.1 kHz to the PWD on any of its inputs using the NativeX setting sends the audio data through the Lens that regenerates a reduced jitter 44.1 kHz audio signal just before the DAC chip then sends it on to be converted to analog audio. The same process is used for audio of all sample rates supported by the input in use.

Upon first receiving the MKII I must admit to not reading the manual. I placed the unit directly in my system and tapped play on my iPad remote control. I was quite underwhelmed with the performance during my first several listening sessions. I had left the PWD MKII in its default Native mode thinking that must be the most pure audiophile method of playback.

Looking at the other modes I knew I wanted to skip the resampling and thought NativeX must be some eXtreme DSP. Given all the uses of the letter X over the last decade, think X Games or Bender from the show Futurama suggesting "Blackmail Is Such an Ugly Word. I prefer 'extortion'. The 'X' makes it sound cool." I figured the X was all about marketing and nothing I wanted to experiment with until a later date. After this less than stellar showing by the MKII I downloaded the PDF manual and noticed NativeX mode and the Digital Lens. Switching from Native to NativeX mode on the MKII the best thing since the Blues Brothers put the band back together.

In Native mode the MKII sounded syrupy reproducing transients with rounded edges and powerful but muddy bass. My go-to HRx 24/176.4 track Passacaglia from the Kansas City Symphony was nothing special in Native mode and I quickly lost interest in the music, checking the CA What's New section every couple minutes. Flipping over to NativeX mode was fantastic. The deep bass immediately tightened up and the high frequencies were brought much more into focus. The biggest impact from NativeX was definitely heard in reproduction of the bottom end.

I continued to bounce back between Native and NativeX mode. Such a comparison should be valuable for those on the fence about upgrading their MKI DACs and readers who may have written off the PWD based on previous MKI listening sessions. In Native mode tracks like Tori Amos' Silent All These Years and Nat King Cole's The Very Thought Of You suffered from a lack of air around the instruments and very tame high frequencies. Nat's vocal has very little gloss to it while Tori's piano appeared deadened from an over damped recording studio. NativeX mode wasn't the cure or panacea that removed 100% of Native's sonic signature in the high frequencies. A slight synthetic character to the high frequencies remained but this was only an issue when compared to some of the best components I've had in my listening room. Nonetheless NativeX gave brought the once dead music back to life.

Once I understood the benefits of NativeX I played some of my favorite tracks in which the Digital Lens could really focus. The first three tracks from Tool's album Undertow, Tower of Power's Diggin' on James Brown, and School from Supertramp's rime of the Century. The lower frequencies from bass guitar on Undertow and School were very tight while David Mann's tenor sax on Diggin' was reproduced with excellent delineation. Nothing can ruin these tracks like sloppy bass. Fortunately NativeX mode cleaned up many of the common ills of reproducing bass and enabled me to focus on the music rather than unmusical playback flaws. For good measure I played the entire Phat Pack album from Gordon Goodwin's Big Phat Band. Through the PWD KMII and NativeX played at Big Band concert levels I had a blast with this one.

BTW, since I should probably have a reputation by now as being something of an old school curmudgeon, I can also reveal that I have a Genesis Digital Time Lens at home, the grand daddy of this technique using a deep buffer for jitter reduction; and though it's limited to 44.1 and 48 kHz material, it does do some nice things in that range....

1990's tech well ahead of it's time.

[wkhanna]

Well, the first egg is broken for this omelet & we are full in on this so-called “budget” level project.
Dan just received conformation on the order for our High-performance rubidium LPFRS 10MHz Oscillators. One for him, one for me.

Time to find out when Macy’s has their ‘Big’ sale on jewelry, I guess……

[jim1961]

I can easily imagine there's more than a small amount of incredulity about some of the things we're discussing and proposing. Particularly this thing about how important the clock and clock recovery without SRC and what not may be to the reproduced sound quality. To that end, I'd like to quote some bits from the Computer Audiophile review of the PS Audio Perfect Wave DAC II.

This is key. Sound quality is such a subjective experience that measurement data sometimes seems to have about as much relevance to what we hear as quantum physics has on driving our car. In a fantasy world, 4 or 5 of us would all live on the same cul-de-sac and routinely visit each others setup and compare notes. In this case, even though our subjective experience would still vary, at least the initial conditions would be the same.

I still hear people claim their is no difference between mp3 and CD or that $300 AVR's sound just as good as pricey audiophile components. I suppose all of us draw the line somewhere, $200 power cords anyone?

Its troubling that science and scientific measurement cant tell us how something will sound, much less whether we will like it or not in a narrow sense. In a broad sense, this same criteria tells us a lot, or at least points us in the right direction.

Case in point, back in the early 80's, I was in the market for new cassette player. Did my research, and compared specs. The DBX units of the time were claiming s/n ratios of 90+ where the Dolby B & C units were in the 70-78db range. Seemed a slam dunk given the other specs were comparable. Upon listening to the DBX unit, I discovered that what no spec told me was that it breathed! 8O

Claims range all over the place to what can actually be heard and not. And sometimes (tube gear), an audible artifact is considered a good thing :W

While better sound is a generally agreed upon goal, the path to attaining that goal is clouded in a myriad of misunderstood facts, myth, omissions and subjectivity.

[JonMarsh]

This is key. Sound quality is such a subjective experience that measurement data sometimes seems to have about as much relevance to what we hear as quantum physics has on driving our car. In a fantasy world, 4 or 5 of us would all live on the same cul-de-sac and routinely visit each others setup and compare notes. In this case, even though our subjective experience would still vary, at least the initial conditions would be the same.

I still hear people claim their is no difference between mp3 and CD or that $300 AVR's sound just as good as pricey audiophile components. I suppose all of us draw the line somewhere, $200 power cords anyone?

Mmm, me, I figure I'll have really made it when I can afford to buy the $1K power cords! Alas, as my career is drawing to a close within the next 4-7 years, I don't see signs of that ever happening.

However, I do have some pricey speaker cables (more than one set) bought on Audiogon, and am contemplating a newer set- but I can rationalize those because of some very specific measured characteristics, which may or may not correlate with the sonics. Unfortunately, I have also heard comparisons with some stupidly expensive speaker cables, Nordost Valhalla, and if the rest of the system is in sync at that performance level, they do deliver better performance- though the value proposition is a real question...

Its troubling that science and scientific measurement cant tell us how something will sound, much less whether we will like it or not in a narrow sense. In a broad sense, this same criteria tells us a lot, or at least points us in the right direction.

Case in point, back in the early 80's, I was in the market for new cassette player. Did my research, and compared specs. The DBX units of the time were claiming s/n ratios of 90+ where the Dolby B & C units were in the 70-78db range. Seemed a slam dunk given the other specs were comparable. Upon listening to the DBX unit, I discovered that what no spec told me was that it breathed! 8O

Well, I think given enough experience and asking the right questions, we can nail down more of the science than we often do. But keep in mind, let's say company X has figured out some really cool stuff- and their subjective testing indicates a disproportionate improvement compared with what they can measure- but they believe they know why and how they accomplished this improvement. How much do you want to tip your hand to the competition by describing it openly, or do you want to just create a plausible marketing story to generate interest in the market, but not tip your hand to the competition?

Yeah, I remember those DBX units from the 70s. We used to sell them, but... Problem is, the only Dolby unit I found I could tolerate was the multi-band Pro Dolby A. The DBX were pretty awful, weren't they? It's like the modern LCD TV's with active backlight control to get high dynamic contrast ratios- the real physical contrast ratio is still sh*t compared with a good plasma, much less my NEC10PG CRT projector.

Claims range all over the place to what can actually be heard and not. And sometimes (tube gear), an audible artifact is considered a good thing :W

While better sound is a generally agreed upon goal, the path to attaining that goal is clouded in a myriad of misunderstood facts, myth, omissions and subjectivity.

While our brains can do some remarkable adaptive DSP (just consider the whole component burn-in phenomena, which I still tend to believe is hearing adaptation, not actual component changes- color me a skeptic on this one) there are some distortions that are more unnatural than others. Digital reproduction distortions take precedence over the relatively simple colorations of a top flight vacuum tube amplifier or zero feedback solid state. That you can hear issues with a DAC that measures 0.002% THD on sine waves, while running it through a JL-1 Signature power amp that is struggling to get down to 0.02 or 0.03% THD are quite audible and remarkably, after auditioning, my colleague/friend in Munich swapped out his Boulder 2060 for a pair of JL1 Signatures, which I heard subsequently in his home, and can testify that they are might fine sounding (closest amp I've heard in the USA in tubes is the Audio Research Reference 150, reviewed in mid 2012 by Stereophile). Does this make sense? Well, not really UNLESS it happens that the types of distortions we have to deal with in a lot of digital gear is stuff that the ear is more sensitive to than simple low order harmonic distortion.

So go figure, listen for yourself, decide what works for you, but every now and then, I suggest trying some things outside your normal comfort zone, just to be sure you are keeping expectations calibrated high enough... so when a real budget buster like the NAD M51 comes along, one doesn't miss the opportunity. :B

[JonMarsh]

A few more random comments on the CA review and the PS Audio PWD Mk II....

• The PWD DACs use Wolfson DAC chips and a Wolfson reciever chip (like the Rega DAC)
• The MKII update with the Digital Lens is available to retro-fit MkI units for $995
• Personally, I would still be interested to hear this unit in a reclocking setup- as the DCD8 improves the subjective performance with the TotalDAC-D1, (every model we've tried it with, including the vacuum tube buffer version), I expect it may improve things with the PWD MkII
• Anyone considering a DAC in the $4K range should give this unit a listen, though I'm by no means convinced it's 2X the M51- and the M51 top end may actually have the edge (Wolfson DAC chips?)

[wkhanna]

Well said, Jim!

And your relentless work with room treatment & its measured effects are envious.
You have measurements that correlate to what you are hearing.

It would seem that measuring the sound wave once it is produced is a task supported by tried & tested science & accepted laws of the physical world. Still, the perceived audible quality of the wave, or put another way, the reproduced wave’s resemblance to the original wave it was created from (assuming it was produced acoustically and not digitally) is still something no instrument other than our individual human auditory system with its psychoacoustic variability is capable of measuring.

[wkhanna]

Dan emailed me today at sometime around 2.5178852191468733412 minutes after 2:00 pm EST to inform me we now have in our possession two LPFRS Rubidium Oscillators.

No, we did not purchase the SRS PRS10.......
Why not......?
For the same reason we did not order the Antelope Isochrone 10M Atomic Reference Generator.
We make lousy jewel thieves.
Besides, this thread is called 'budget' re-clocking, remember?

Now that our goods from the international (China) order have arrived safe & sound, we are ready to get our PS, which being a standard domestic industrial device, should arrive in 5 days or less after ordering.

After a quick look around I found this budget-friendly offering........

Wonder if this is an acceptable substitute?

Link

[JonMarsh]

Looks like the same unit, the only concern is the bit about "condition unknown"; which means I'd suggest hooking up a small load resistor (say, 100 ohms, 10-20 W), and see if you get a nice 24V output. You guys do have a DVM of some kind, right? :W If not let me know, I can recommend some "budget" models.

[wkhanna]

Yep, I have a good Fluke, & I know Dan has a good DVM, too.
I suppose we can take the assembled units to our local shop, Pittsburgh Vintage Hi-Fi, & have Anders put it on his scope to verify the 10 MHz output, too?

There are a few other PS's on Epray, some NIB at under $200, may invest a bit more to insure we get a working unit right off the bat.
Then, there is always the economy model you recommended, also.

[JonMarsh]

If you have someone you can go to to verify the 10MHz output with the SMA to BNC cable, that's a great idea. Much recommended.

In God we Trust, all others we verify.

[JonMarsh]

BTW, if it's possible to find the time, I will be setting up a reclocking listening setup at home, apart from what's at GF's, with the iMac driving a Berkeley Alpha USB and with the iPad in the Cambridge Audio ID100 digital converter; and will be testing using AES/EBU output with 44.1 kHz material using the Genesis Digital Lens, the Brainstorm DCD8, and the Mutec MC-3 Plus. First steps confirming setup and operation for each piece, then hopefully doing some comparisons with headphones (no speakers at home right now, and living room taken over by music land (think Marshall stack, multiple heads and effects, multiple guitars on stands, and keyboards).

This WILL be competing for time and resources for the new midrange crossover build for the Isiris, as well as doing testing at GFs. Oh, and I'd better do some laundry this weekend, too, and take GF out.

[wkhanna]

My humble recommendation is to start with the last item on your list first. :W

[JonMarsh]

My humble recommendation is to start with the last item on your list first. :W

But Bill, we must maintain the hysterical, er, historical progression! And from much prior experience, I know roughly how well the DCD8 works, so verifying that in the current context is necessary to evaluate the Mutec. Now, I'll admit the Genesis is just an historical curiosity which I throw into the mix just to exercise my powers of curmudgeonly historical precedent... :W