Don't think so. Newer high quality DACs are very immune to incoming clock jitter from cables or from any other source.
Here is what one DAC manufacturer says about it: "If we want to remove jitter it may seem logical to try and measure the jitter-induced timing errors on a sample-by-sample basis. Unfortunately this technique would have errors exceeding 1000 picoseconds and would produce very poor results. A much better technique is to measure the frequency ratio between the incoming jittery clock and a stable clock that is used for D/A conversion. Over a sufficiently long period of time we can calculate the frequency ratio of the two clocks to a very high precision. The incoming samples can be buffered and then shifted out at a rate that is determined by this precision ratio. If the ratio is locked down and not allowed to change, all of the jitter will be removed, but if frequency of the incoming clock drifts, we may eventually run out of space to store incoming samples, or run out of samples in our buffer. On the other hand if we allow the ratio to track any slow drift in the incoming sample rate, we can manage our data buffering requirements, but some low-frequency jitter will be encoded into the audio waveform as it passes through to the output clock domain. The key is to only allow very slow changes in the ratio calculation. To do this, the DAC2 has a ratio tracking filter with a corner frequency set to about 1 Hz. All interface jitter above 1 Hz is rejected, while interface jitter below 1 Hz is encoded into the output waveform. The 1 Hz corner frequency was selected because jitter frequencies below 1 Hz are far too low to cause audible problems. The magnitude of interface jitter below 1 Hz is also very small."
Which leads to a jitter specification of "Maximum Amplitude of Jitter Induced Sidebands (10 kHz 0 dBFS test tone, 12.75 UI sinusoidal jitter at 1 kHz)
< -144 dB"
Inside the DAC2 - Part 2 - Digital Processing - Benchmark Media Systems, Inc.
Yea, pretty much the whole premise of the thread was wrong, the test was wrong, the test organiser has no notion of what he is doing & the participants & some of the posters & readers of this thread have fallen for the charade (it can only be called a charade because the o/p resisted all advice given to him about the faults of the test).
Anyway, all that seems to be over now & some sense & sensible voices have returned
Happy New Year, all!!
Anyway, all that seems to be over now & some sense & sensible voices have returned
Happy New Year, all!!
It's easy to be 'apparently' immune to jitter if you mask it with other distortions/noise or if you overlay jitter on all sources - as was probably the case in these blind tests where an optical feed to the DAC was being used.
Jitter isn't the only factor that determines how a DAC sounds
About the Benchmark DAC - Here's the thing with all ESS DAc chips - even though their jitter reduction ASRC is said to be state of the art, if you bypass their ASRC by feeding a synchronous clock to the DAC the sound noticeably improves!!
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Some types of jitter are known to create euphonic distortion which is not more accurate, although it may sound subjectively better to some people. Not what I want, myself.
Are you saying the ES9018 DAC which is used in the Benchmark DAC creates euphonic distortion, then?
Using your subjective experiences as basis to prove that some DAC-s sound better is such a disaster.
Imagine if people relied on just on their experience to determine which objects "falls down faster"(with air resistance), heavy ones or lighters ones. Even today people assume that heavier object fall faster based on their intuition and perception! Mathematics and physics have precise tools to describe reality and what is going on resonantly accurate.
Humans perception can be shaped and change based on experiences and suggestion they get. Humans suffer from biases, all humans. We have to try to minimize those in order to get closer to the truth.
We do need:
-more sophisticated testing methods
-more scientific testing methods
-models that do not invent new weird theories completely based on what we hear(more like what we think we hear)that defy physics and keep in mind that Electromagnetism is established part of physics
Imagine if people relied on just on their experience to determine which objects "falls down faster"(with air resistance), heavy ones or lighters ones. Even today people assume that heavier object fall faster based on their intuition and perception! Mathematics and physics have precise tools to describe reality and what is going on resonantly accurate.
Humans perception can be shaped and change based on experiences and suggestion they get. Humans suffer from biases, all humans. We have to try to minimize those in order to get closer to the truth.
We do need:
-more sophisticated testing methods
-more scientific testing methods
-models that do not invent new weird theories completely based on what we hear(more like what we think we hear)that defy physics and keep in mind that Electromagnetism is established part of physics
Not normally. If you disable its jitter rejection technology, then it potentially might depending on the spectrum of jitter it receives.
Ah, I see what you mean - I forgot to say l in my original post ESS DACs sound better when their ASRC is turned off only when fed with a lower jitter source - in other words the ASRC is masking the better sound quality when a lower jitter source is the input. With higher jitter input it improves the sound.
My takeaway is that there's a certain masking happening with the ASRC which hides the jitter & makes all inputs sound the same. Hence apparent immunity to jitter may actually be a result of masking - making all inputs sound the same.
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"Masking" is probably the wrong word. "Jitter rejection" or "jitter attenuation" is what it does. The way it works is that the clock that runs the DAC is an ultra low jitter clock. The incoming data from a CD player or computer is clocked separately, and is referred to as the transport clock. The ultra low jitter DAC clock and the transport clocks are two different things. Incoming data is reclocked into the DAC using the low jitter DAC clock. Any jitter from the transport clock is essentially thrown away and has no effect. In order so as not to run out of memory or incoming data, transport clock jitter occuring at less than 1 Hz is allowed to come through, but transport clocks and cable jitter have very little jitter activity below 1 Hz, and it is too low frequency to affect the sound of music, therefore it does not create any problems. If it did cause problems, they could just lower it to 0.1 Hz or whatever, but 1 Hz is low enough.
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Yes, that is the ESS marketing copy alright but it doesn't stand up in practise as I just explained & as I think Dustin Forman also admitted to somewhere on this forum.
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I don't think you explained anything exactly, you suggested that some kind of masking was involved, but you didn't explain the what and the how of it. What is masking what and how?
Also, do you have a link to Dustin Foreman saying what you claim about masking?
I believe I said it clearly in my previous post "ESS DACs sound better when their ASRC is turned off only when fed with a lower jitter source - in other words the ASRC is masking the better sound quality when a lower jitter source is the input. With higher jitter input it improves the sound."
That is my & others experience. The how & why of it, I don't really know - it would take deep knowledge of exactly how the ESS PLL works - stating Any jitter from the transport clock is essentially thrown away and has no effect."is just marketing talk & doesn't add much to understanding
I'll try & find the Dustin Forman post
Several weeks ago I read several pages of this thread. It started off pretty interesting and kind of went downhill...
From the original post:
That is pretty interesting. Not rocket science, not the level of testing a potentially-fatal drug, and doesn't seem to claim that there is no DACs anywhere in the universe that can't be ABx'd. (Granted, the original post has been edited, so maybe the claims were more extreme at some point.)
132 pages later this is where the thread is at...If you listen to the right recording, with the right amplifier, on the right speakers, train yourself what to listen for, in the right listening position, with the moon in the correct phase, then you can tell some DACs apart? That is NOT interesting.
Here is what matters...on the amplifier in my living room, with my speakers and my digital source, can I tell the difference between DACs?
I don't know the answer because I don't have multiple DACs and am not about to go buy a $3,000 dollar one. But, thanks to JonBocani, I do know that in one particular instance, one small group of people with high quality gear could not tell the difference. That is valuable information to me and affects my priors.
From the original post:
That is pretty interesting. Not rocket science, not the level of testing a potentially-fatal drug, and doesn't seem to claim that there is no DACs anywhere in the universe that can't be ABx'd. (Granted, the original post has been edited, so maybe the claims were more extreme at some point.)
132 pages later this is where the thread is at...If you listen to the right recording, with the right amplifier, on the right speakers, train yourself what to listen for, in the right listening position, with the moon in the correct phase, then you can tell some DACs apart? That is NOT interesting.
Here is what matters...on the amplifier in my living room, with my speakers and my digital source, can I tell the difference between DACs?
I don't know the answer because I don't have multiple DACs and am not about to go buy a $3,000 dollar one. But, thanks to JonBocani, I do know that in one particular instance, one small group of people with high quality gear could not tell the difference. That is valuable information to me and affects my priors.
In my search for Dustin Forman's quote I came across this post from Bruno Putzeys which explains the problem with all SRCs Hypex Ncore
"@Arthur I think it would take us off topic to discuss in detail. ESS's claims regarding the SRC pertain to the actual interpolation process. They have done an absolutely splendid job there. The thing that causes problems is the ratio estimator (the bit that works out exactly what sampling time to interpolate). The bandwidth is too high which lets through time quantization errors. The circuit samples the incoming clock signal using its reference clock. This results in the addition of jitter with a peak-to-peak value of one reference clock period (e.g. 25ns for a 40MHz reference clock). Next the number of ref clock periods in one input clock cycle are counted and this constantly changing number is fed into a low-pass filter which outputs a cleaned-up version of the ratio between the reference and input clocks. This ratio is then used to space the "virtual resampling points" calculated by the interpolator. Of course the low pass filter doesn't output pure DC. The spectrum of the counter output consists of mix products between the two clocks. The filter can only attenuate those. The attenuated spectrum shows up as close-in FM sidebands exactly like jitter. DNR and THD measurements ignore those. The SRC successfully removes high-frequency jitter, thus guaranteeing good SNR, but it adds low-frequency phase modulation of the signal that wasn't even present in the input clock. All SRC's do this but the bandwidth of the low-pass filter determines whether this is an issue or not.
The attached two measurements were made on a standalone test chip for the ESS SRC which was never issued as a product, but the actual SRC did go on to be used in the Sabre DAC."
And this Hypex Ncore
"I also worry about the SRC on the ESS not being defeatable (if it is the same one I tested 4 years ago it's not that great) but it becomes transparent if you lock all clocks together. "
Can't find Dustin's comments about this but it is all over this forum that "synchronous master clocking on ESS" DACs brings benefits - just do a search on the quoted text
"@Arthur I think it would take us off topic to discuss in detail. ESS's claims regarding the SRC pertain to the actual interpolation process. They have done an absolutely splendid job there. The thing that causes problems is the ratio estimator (the bit that works out exactly what sampling time to interpolate). The bandwidth is too high which lets through time quantization errors. The circuit samples the incoming clock signal using its reference clock. This results in the addition of jitter with a peak-to-peak value of one reference clock period (e.g. 25ns for a 40MHz reference clock). Next the number of ref clock periods in one input clock cycle are counted and this constantly changing number is fed into a low-pass filter which outputs a cleaned-up version of the ratio between the reference and input clocks. This ratio is then used to space the "virtual resampling points" calculated by the interpolator. Of course the low pass filter doesn't output pure DC. The spectrum of the counter output consists of mix products between the two clocks. The filter can only attenuate those. The attenuated spectrum shows up as close-in FM sidebands exactly like jitter. DNR and THD measurements ignore those. The SRC successfully removes high-frequency jitter, thus guaranteeing good SNR, but it adds low-frequency phase modulation of the signal that wasn't even present in the input clock. All SRC's do this but the bandwidth of the low-pass filter determines whether this is an issue or not.
The attached two measurements were made on a standalone test chip for the ESS SRC which was never issued as a product, but the actual SRC did go on to be used in the Sabre DAC."
And this Hypex Ncore
"I also worry about the SRC on the ESS not being defeatable (if it is the same one I tested 4 years ago it's not that great) but it becomes transparent if you lock all clocks together. "
Can't find Dustin's comments about this but it is all over this forum that "synchronous master clocking on ESS" DACs brings benefits - just do a search on the quoted text
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Sometimes people think that -60dB 3rd harmonic sounds "better" than -110dB 3rd harmonic. It sounds more clean, detailed, and 3-dimensional, they say. I don't want that. I would rather have the one that measures better if it is more accurate. So, when people throw out that something sounds "better" is that including if it is more distorted but the distortion sounds good? Also, how was it determined that low jitter sounds better with ASRC off? Blind testing of some kind? Measurements to correlate with preferences? Any science in what was done?
Otherwise it means almost nothing to tell a story with nothing behind it that could be replicated in scientific study. It's like all other old research out that may be bad because we don't know all the details of how it was conducted.
Ah, yes we can all be skeptical of anything that isn't scientifically & rigorously proven. Not everything can be & personal experience & evaluation become the yardstick. So I tried this as did many others including Bruno above & others who have specialised in ESS DAC boards & worked closely with Dustin & ESS but there is always a reason that these opinions can be rejected & ESS marketing copy believed instead
ESS9018 - try new, try more...
I guess my previous post from Bruno didn't pique your interest enough to look up the link - with his text "The attached two measurements were made on a standalone test chip for the ESS SRC which was never issued as a product, but the actual SRC did go on to be used in the Sabre DAC."
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I would agree that how interpolation is done matters a lot. It is possible to do it extremely well with enough computational resources, but I don't know all the details of how Sabre does it. Currently Sabre DACs are measuring jitter sidebands down at -144 dB. That's pretty darn low. That being said, I still don't think data converters I have ever heard are completely transparent. But, they do keep getting better, and the better implementations of Sabre DACs, such as Benchmark, are quite good and probably about as good as anything out there by most accounts (perhaps excepting one of Bruno's Mola Mola type DACs which I will probably never hear).
Looks like in the same post Bruno said, "All SRC's do this but the bandwidth of the low-pass filter determines whether this is an issue or not." Currently the low pass filter corner is set to 1 Hz, and it starts attenuating at around 0.1 Hz. That makes the low pass filter *very* low, and as Bruno said the filter bandwidth determines if there is a problem or not. Looks to me like they have set the filter bandwidth to make it *not* a problem in the current family of DACs.