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In Reply to: Too Much Coffee..... posted by Todd Krieger on March 14, 2007 at 23:50:48:
For reducing jitter? I have a XA777ES SACD/CD player. Please don't flame my player! :) Thanks in advance.
Follow Ups:
"For reducing jitter? I have a XA777ES SACD/CD player. Please don't flame my player! :) Thanks in advance."Since it's a one-box player, you really can't do much "between transport and DAC".... Your best bet, although it's more an RFI solution than a jitter solution, is use an isolation transformer to the AC mains. But be sure you just use for the digital source, so the noise will be isolated from the analog.
...to output balanced 120 volts. Will inserting this transformer to power (only) my CD player act to isolate the digital noise/RFI from the analog sources I have?Steve
The player already does that. It has twin R-core transformers. There's actually not much that can be done on this player, other than perhaps better shielding around the analog "cage".
I seem to recall Stereophile measured it at 178ps peak to peak - this is about half that of most universal players. Offhand, I know of only a handful of players with lower measured jitter (eg. Linn CD12, Sony SCD-1, and these are not that much lower maybe 140-150ps).You will find it a struggle to achieve further significant reduction, ie. less than 100ps.
Worst case analysis suggests jitter as low as 20ps may be audible, but realistically the only way to get anyway close to there is a unit with no moving parts (ie. no transport), something like the Slim Devices Transporter for instance. The problem with these network music players is that they usually require a CPU fast enough to take audio off a network, and decode from various lossy and lossless compression formats, and all these CPU activity can generate jitter (through noise injected back into the power supply and then into the DAC).
Off the top of my head, if you really want to play around - try EMI/EFI shielding the audio cards, but I'll have to issue a warning: once you open the player up, you will realise everything is very tight in there (Sony really went overboard with the engineering) so there's very little room for tinkering.
You could also try to reshape the jitter spectrum (there's speculation that jitter audibility may be dependent on it's frequency profile). Replacing the clock, playing around with power supply regulation, are some potential avenues to explore, but not for the faint-hearted. Personally, I am not convinced that modders who do these sort of things actually result in lowering jitter (yes, they all claim it sounds better, but in reality it may just sound different, and different is not necessarily better). I have never seen empirical before and after jitter measurements as a result of modding a player.
Todd is right - at the end of the day almost anything and everything can cause jitter, so reducing it is not necessarily an easy task.
< < almost anything and everything can cause jitter, so reducing it is not necessarily an easy task. > >True, but neither is measuring it!
In the old days Stereophile used a jitter analyzer made by Ed Meitner. The output was connected to a spectrum analyzer to show the spectral content of the jitter. The input was made by connecting to the word-clock pin on the DAC chip. There were several problems with this system:
a) It required disassembly of the CD player to connect it to the word-clock pin.
b) In some players the word-clock pin was not accessible without dismantling the player to the point where it wouldn't function.
c) Early DAC chips were most affected by jitter on the word clock pin, so this was an appropriate thing to measure. But later DAC chips were sometimes most affected by jitter on the bit-clock pin, and most modern (delta-sigma) DAC chips are most affected by jitter on the master clock pin. So there was a problem of what *should* be measured.
So some years ago, Stereophile switched to the Miller Audio Research jitter analyzer. This has the advantage that it connects to the analog outputs of the unit under test, and so no disassembly is required.
But there are some disadvantages as well:
a) The jitter number as measured by JA cannot be compared to the jitter numbers measured with the Meitner analyzer (typically done by Robert Harley in those days). A good player measured on the Meitner machine might measure as low as 20 - 30 pS, and I believe there was one Meridian model that got down as low as 15 pS. In contrast, the numbers as measured by the Miller Audio Research machine are typically 10x as high.
b) There seems to be some unexplained variability in the numbers as measured by the Miller Audio Research machine. For example, when JA re-measured the dCS Elgar DAC during his follow-up on the Verona master clock, he found that the same machine had jitter measurements approximately twice as high as previously measured, with no explanations available. (See footnote 1 in the link below.)
For the first several years, most of the better players measured around 150 pS with the Miller Audio Research analyzer. But then for a period of two years or so there wasn't a single player that was much below 250 pS. Now in the last two issues there are two machines that measured just under 200 pS. To me it would seem to suggest some sort of systematic variation in the measurement setup.
I have tried to duplicate the Miller Audio Research method, using two different analyzers. With the first, an Audio Precision System One (as used by Stereophile for most of their amplifier and preamplifier measurements), the results for an Ayre CX-7 (that fared well in the Stereophile test) was much poorer. But when I repeated the test using the very latest (and greatest) Audio Precision analyzer (the 2722), the results were much better than either the System One or the Miller Audio Research analyzer.
The bottom line is that it is *very* difficult to compare jitter measurements, even those taken from the same source.
< < Your player already has fairly low jitter measurements. I seem to recall Stereophile measured it at 178ps peak to peak - this is about half that of most universal players. > >
I wouldn't put much stock into these numbers. There seems to be too many other variables to say with any certainty that the Sony 777 really has about half the jitter of "most universal players".
I would agree that jitter is not easy to measure. For example, the Miller analyser is (I believe) based on Julian Dunn's J-test method. The problem is that this does not actually measure jitter, what it's measuring are artefacts associated with perturbing a sine wave with a 1-bit amplitude square wave.Whilst this does highlight the sort of artefacts produced by jitter, it can also highlight artefacts not related to jitter. And some jitter artefacts may not be measurable by this method.
I am less familiar with the Audio Precision instruments, but from what I understand their ADC based measuring equipment does not have sufficient precision to really measure jitter either.
I would however say that if two players are measured using the Miller analyser (which would be the case for all the recent Stereophile reviews), then it is valid to compare the measurements, in the sense that one player generates less (or more artefacts) in response to a specific test signal compared to the other [however, note: as I've pointed out, this difference may not be isomorphic to the underlying jitter differences between the players].
What I do know from examining a Sony XA777ES (note: this is a completely different design than the Sony 777ES) is that there seems to have been quite a lot of attention paid to reduce the effects caused by logic induced modulation, which is a major source of jitter, certainly more so than on a typical universal player. And having an audio clock that is not regenerated from 27MHz certainly helps. So I would suggest that the comparative jitter numbers are not surprising.
My point is that I'm not sure that jitter on this player (measured using any method) can be substantially reduced by tinkering with it. All the obvious areas have already been addressed (apart from perhaps better EMI/RFI shielding), and in any case there isn't any room to do anything really fancy anyway.
If I really wanted to dramatically improve the sound of the player, I would probably investigate replacing Sony's proprietary digital filter. The Triple DAC architecture is also a bit bizarre, it's more likely to increase rather than decrease noise (but I suspect Sony may have implemented it for euphonic reasons).
< < the Miller analyser is (I believe) based on Julian Dunn's J-test method. The problem is that this does not actually measure jitter, what it's measuring are artefacts associated with perturbing a sine wave with a 1-bit amplitude square wave. > >Yes, I believe that the Miller analyzer does use the "J-test" method. This is actually a large-amplitude *square* wave that is overlaid with a 1-bit amplitude square wave of a different frequency. One big problem with this method is that it was actually developed to stimulate the worst-case scenario of jitter introduced by an S/PDIF (or AES-EBU) link. It therefore has virtually no application whatsoever to one-box players, as they don't use the bi-phase mark encoding that is susceptible to the signal used on the "J-test".
< < What I do know from examining a Sony XA777ES ... is that there seems to have been quite a lot of attention paid to reduce the effects caused by logic induced modulation, which is a major source of jitter, certainly more so than on a typical universal player. > >
While this may be true of a "typical" universal player, I can assure you that there are certainly examples of universal players tested by Stereophile where this was *not* true. And yet the measured jitter numbers do not correlate well with what would be expected by comparing the designs. I believe that this is a flaw with the test (or its implementation), and not the result of faulty analysis. We already have one data point that shows 100% variation in the results of the Stereophile test on the exact same machine.
*** I can assure you that there are certainly examples of universal players tested by Stereophile where this was *not* true ***Any examples? :-)
*** I believe that this is a flaw with the test (or its implementation), and not the result of faulty analysis. ***
At the end of the day, what the Miller unit measures is a player's response to a specific test signal. Whether or not this is indicative of underlying "jitter" is debatable.
It's easy to downplay or dismiss these measurements and say they are not relevant. However, the presence of artefacts in any test signal indicate that something is wrong (whether or not that something is "jitter" or not), and if a player does not perform well on this (or any other measurement) I would be worried. If it was my design, I would try and find out why. The answer may be jitter. But even if it is not, it's worthwhile addressing.
*** We already have one data point that shows 100% variation in the results of the Stereophile test on the exact same machine. ***
There are also examples where players give wildly different results depending on whether upsampling is turned on or off, or optical vs coaxial, etc. I would also be concerned if a player exhibited variations in response to a fairly simple test signal. It indicates something somewhere is not quite right. Rather than blaming the "test" as faulty, I would be interested in understanding why this variation occurs.
< < Any examples? > >Sure, the Ayre C-5xe. In that player, there are two independent master clocks, one a multiple of 44.1 kHz and the other a multiple of 48 kHz. Each one is an ultra-low jitter design, and the unused clock is turned off to prevent any interference. All other required clock signals are derived from this audio master clock.
The critical clock pathway it optimized in every way we could think of:
a) The clock is less than 1" from the DAC chip, and the traces are controlled impedances that match the clock output buffer. This minimizes the chances of reflections or degradation.
b) Each gate in the critical clock pathway is a "Pico-gate", meaning that there is only one digital gate in each package. This means there is no ground "bounce" between the different gates, as each individual gate has its own ground pin.
c) Each of these individual gates has its own individual power supply pin, which is fed by its own zero-feedback, discrete transistor power supply regulator, and has its own ultra-low impedance bypass capacitor. This means there is virtually zero cross-talk (LIM) between the gates via the power supply.
d) There are two separate power transformers, one for the transport, controller and all non-critical digital circuitry. The other "clean" transformer is for the analog circuitry, but has a separate winding just for the critical clock path.
There's probably a few more things, but that's what I can think of off the top of my head.
I haven't looked inside the Sony so I can't say for sure how it is implemented, but I would be absolutely shocked if they paid as much attention to detail as does the C-5xe.
Yet according to Stereophile's measurements, the Sony has lower jitter than does the C-5xe. Yet when I replicate the Miller test using the AP 2722 as a spectrum analyzer (instead of the National Instruments PC plug-in spectrum analyzer card used by the Miller system), I get a *far*, *far* cleaner spectrum than does *any* player JA ever measured.
On the other hand, I get a noticeably "dirtier" spectrum if I use the older AP System One as the spectrum analyzer. My conclusion is that what this test is mostly measuring is the noise floor of the spectrum analyzer being used. And what's more, for some reason, the noise floor of JA's test setup seems to have changed over time. Looking at all of the test reports over the years, it would appear that the measurement floor of Stereophile's test setup has varied from around 150 psec for the first few years, to around 250 psec for the last few years to around 190 psec for the last few months.
On the other hand, if you get a really poorly performing player, then differences can be seen even with the Miller system. This would explain why some units JA have tested have shown as much as 1000 psec of jitter. In these cases the jitter level of the player is much higher than the measurement floor of the Miller system.
I haven't seen or heard your player, Charles, so I can't comment on how it compares with the Sony.However, the original topic was whether there were any ways of reducing jitter on the Sony, and I can state with some confidence based on my knowledge of that player that Sony seems to taken reasonable care that department, and the "proof" (for what it's worth) is that it's one of the lowest measured by Stereophile.
Does that mean it sounds better than your player? Not necessarily. Would it measure even better if we used something other than the Miller unit? Perhaps, perhaps not.
It's interesting that you suggest that perhaps the Stereophile tests show a pattern over the years. If that is something of concern to you, you could ask John Atkinson to remeasure your player and see if the results are significantly different. But I didn't think your player measured that poorly - seems to be in the same league as other players based on a Pioneer platform (like the Bel Canto for example).
It would be interesting if all Pioneer-based players do measure roughly the same, because it would imply that the main constraint to jitter performance is upstream (close to the optical drive or media decoder) rather than downstream (close to the DAC). However, I don't have enough evidence, so that's just speculation on my part.
PS - I like your idea of using separate clocks. I'm using the same idea in my player - the only problem is the DAC needs to be "reset" everytime the clock is switched. On my player, this effectively requires a number of other chips to be reset as well, so it's a fairly intrusive (and potentially error prone) operation. On rare occasions (I have only encountered it 3-4 times so far), the clock switch is not successful, and I lose audio output. Have you had any problems switching clocks? Can you do it on the fly? (ie. can you switch between 44.1 and 48kHz on a single disc whilst it is playing (for example, switching between audio tracks on a DVD-Audio)?
Most of the pro audio equipment I have seen with separate switchable clocks require a power cycle to effect the switch. For example, my SD90.
< < Have you had any problems switching clocks? > >No. You just have to spend a lot of time figuring out what each chip needs and at what time.
Everything is controlled by a custom-programmed Xilinx FPGA. The Pioneer MPEG decoder board outputs a flag that signifies if the sample rate is a multiple of 44 or 48 kHz. We determine the sample rate (1x, 2x, or 4x) by comparing the word clock against master clock. Then the FPGA sends all the appropriate signals to all the various chips so that we can change sample rates on the fly.
I hope you don't mind me asking another question.Is the Pioneer MPEG decoder also clocked from a derivative of your master clock (based on your earlier statement that everything is referenced to the audio clock)? So when you change the master clock from say 44.1x to 48x won't it potentially affect the decoder?
Or is the decoder clocked independently?
< < Is the Pioneer MPEG decoder also clocked from a derivative of your master clock? > >Yep.
In this particular case, there are about 5 different frequencies fed to the MPEG decoder. Some are based on 27 MHz (video stuff), some on multiples of 44.1 kHz (for CD stuff), some on multiples of 48 kHz (for DVD stuff), and some that change depending on whether you are playing a CD or a DVD.
Normally these clocks are all generated by a 3-PLL chip that is driven by a 27 MHz crystal oscillator. We remove the crystal and inject our own 27 MHz signal that is derived from our audio master clock by a second custom-programmed PLL that we add. The "N" and "M" divide ratios in the second PLL are controlled by the same flag that tells whether to use the 44.1 kHz based audio clock or the 48 kHz based audio clock. So when we change audio clocks, the second PLL just keeps outputting the 27 MHz to the PLL for the Pioneer MPEG decoder. Everything always stays synched up, but is governed by the audio master clock.
The details are different, but a similar implementation is used in our D-1xe and DX-7e DVD players. The audio clock is the master, and everything else is slaved to that.
I'm amazed that switching clocks does not cause a glitch on the 27MHz line. Good on you, bet that took some effort to get right!I seem to recall you are also using an upsampling filter. Is that true? If not, then presumably you need to buffer the audio signal from the MPEG decoder to filter out the jitter?
If you are (using upsampling), did you consider using ASRC on everything to a single common rate (eg. 200kHz)?
There's been a thread recently that the Lavry DA10 does this (even though the user manual suggests otherwise). Everything is resampled to 115kHz - I was wondering why Dan chose such a low frequency (since that would mean 192kHz gets downsampled), then I realised he's trying to force the DAC into dual rate mode.
< < presumably you need to buffer the audio signal from the MPEG decoder to filter out the jitter? > >Yes but since everything is slaved synchronously, only a one-bit buffer (aka flip-flop aka reclocker) is required.
Also remember that each DAC chip's architecture results in different sensitivities to jitter on different pins. In the Burr-Brown DSD1792A that we use in the C-5xe, the master (system) clock is the critical one.
< < did you consider using ASRC? > >
No, the very idea of changing the data like that gives me the creeps. It just seems like an inherently bad idea. YMMV.
*** No, the very idea of changing the data like that gives me the creeps. It just seems like an inherently bad idea. YMMV. ***Well, the data gets changed in the digital filter anyway. I don't really like it myself, but only because typical ASRC implementations generate a fair amount of artefacts and have non perfect passbands. If there was a "perfect" ASRC implementation, then I would consider it. Practically though, I don't think we'll see one soon (and it will require so much computational power it will actually *generate* jitter hence defeating it's purpose)
PS - the reason I was asking whether you were using an upsampling filter was because it *could* account for your slightly high Miller results, so the unit is actually measuring artefacts generated by the upsampler rather than the underlying jitter.
PPS - Did you see the Stereophile review of the Transporter? The jitter numbers aren't that great - I think 293ps peak to peak in 16-bit mode. So much for Slim Devices claiming the peak to peak would be single digit ps. I have a feeling most of it is probably caused by logic induced modulation.
< < Well, the data gets changed in the digital filter anyway. > >Kind of. Many (but not all) digital filters for audio use algorithms that leave the original data points untouched and just interpolate new samples between the original data points. I don't want to get bogged down in semantics, but to me this is clearly different than an ASRC where none of the original data points survive the process.
< < 293ps peak to peak in 16-bit mode. So much for Slim Devices claiming the peak to peak would be single digit ps. I have a feeling most of it is probably caused by logic induced modulation. > >
And like I've said in previous posts, I disagree. I think the bulk of what is measured with the Miller analyzer is simply the noise floor of the spectrum analyzer used.
Do you have access to a spectrum analyzer? It's very simple to make the test disc. The original paper by Julian Dunn is linked below. Look at section 2.6 to see what the waveform is:
C00000 C00000 4000000 400000 (24 times) BFFFFF BFFFFF 3FFFFF 3FFFFF (24 times)
So there is a square wave at Fs/4 that for part of the time is mostly "zeroes" and for part of the time is mostly "ones". Being mostly "zeroes" or mostly "ones" makes a big difference in the jitter added to a bi-phase mark encoded signal such as S/PDIF, but it shouldn't really do much to a one-box player that avoids bi-phase mark encoding.
And that's exactly what I see when I use the (latest, greatest) Audio Precision 2722 as the spectrum analyzer. The noise floor of this machine is below -140 dB. With a 16-bit test signal, the only spuria I see are the expected ones at the frequency that the LSB modulation is applied. But with a 24-bit test signal even this is below the noise floor of the analyzer. I will be glad to e-mail some sample spectra to you (in PDF format) if you will be willing to host them somewhere so that other Asylum members can link to them and also look at them.
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