Some additional thoughts about measuring jitter

Great post JR_Audio. I hoped Steve would chime in as well. I also recommend reading Matt Newport's short paper on Word Clocks http://www.blacklionaudio.com/Word_Clock_Myths.pdf He has done extensive testing on jitter himself and his results are interesting. Not to mention this coffee is giving me jitter.

Your findings are dead-on IMO. It's really the frequency content, and furthermore how this is related to the signal that matters than the absolute magnitude. The magnitude tells you very little about the audibility of jitter when you get down to reasonably low levels. For instance I use two clocks that are both rated at 2psec RMS jitter, but they dont say anything about the spectra in the specs. The two clocks sound noticably different in my system.

A useful jitter characterization standard should be created for audio. The current measurements tell you nothing.

Steve N.
Empirical Audio

From Dr Chris Smith Roke Manor (Divison of Siemens)

Ash,

Reading through that CA thread on jitter, it occurred to me that no one had bothered to check out the science.
So here it is. The effect of jitter on a sampled signal may be expressed as a signal to noise ratio :-

SNR(jitter) = -20*log(2*pi*tor*freq) dB

Where
tor = the jitter in seconds.
freq = frequency of the sinewave test signal in Hz.

So the degradation is proportional to frequency, and the worst case would therefore be freq=20kHz. However, as the ear's sensitivity has rolled off by at least 20dB at 20kHz, a more realistic test would be ~4kHz.

If we set the SNR target due to jitter at 100dB at 4kHz, so as to be masked by the 16 bit quantisation noise, plugging the numbers into the equation above gives us the absolute minimum audible jitter = 400ps. This is way above the level deemed to be necessary by the armchair experts. Maybe you could run it past Martin.

Chris.

DAC Chip manufacturers quote jitter figures and effectively dictate how you build a DAC. All modern A to Ds and D to As (the only places jitter occurs) are well with in this specification and can be further improved with SRC if necessary.

Jitter is yesterday's news IMO and not really an issue.

Ash

Hi Juergen, allow me to chime in.

Although I am much more at the beginning of the measuring process, people may know I am in the very same boat, and btw have read parts of Julian's writings as well (and have it at hand as a guide).

Personally I do not know what it is we hear for changes, and while it is obviously the most logical it would be about changes on jitter or the jitter spectra indeed, I wonder whether this really is so. But do we have another explanation ? I'm afraid we have not.

One explanation could be that the way we measure jitter is not right, and all tests applying it, flaw. Could that be ? hmm ...

I am ignorant enough to look for reasons where I possibly should not because of a lack of knowledge, but let me begin with one question I have, and with possible interesting answers :

I must excuse myself to those who read it for the 100th time, but I am very much dedicated to "non oversampling" which means nothing less than what goes digitally in for whatever waveform, must come out like that on the analogue side. This, opposed to oversampling DACs (think delta-sigma here) which explicitly do not. The story about a pure square turning into a pure sine ...
Now if someone could tell me :

When you imagine the most precise digital wave of random type going into the DAC, some wave coming out of it, but due to heavy oversampling it is impossible to have stayed the same (take that square and you will understand), how in the world would a measuring device be able to measure jitter by means of comparing the digital input to the analogue output ? Still these devices do ...

Going a step further on this ...
Knowing what it takes to ever pertain the digital wave, and then looking at the DAC chips needed for it, including the NOS principle and some filtering, or no filtering if you want ... and now compare this to the A/D chips and used principle on the input side of the measuring device and *knowing* that no chips exist other than sigma-delta to do it, hence oversampling and a different kind of filtering applied to sort it all out ... how in the world jitter measurement with reliable results can come from this ?

I say it can't. Not without further understanding.
But, assuming that there isn't anything to understand, I'd dare claim that something must be wrong, because jitter can still measure as a few ps, while the signal has deformed to a high extend and there is no way this is off a few ps only.

I know, I'm talking apples and oranges here, because jitter should not be compared with a resulting signal as such, but when I'd be talking apples and apples things don't fit, and there won't be such a thing as jitter measurement at the analogue output.
But measurement still suggests it ... (including Julian Dunn btw).

The only real conclusion I can have about this - in this stage of knowledge - is that the presented jitter figures can't be right. From this must follow that with the right figures at hand (but who has them, and who can measure them) we might see something completely different. The results might be in the hundreds of ns, and who knows the one software may show 100ns less opposed to the other.

Then, my earlier suggestion about what we're perceiving is not jitter but "something else" is indeed my thinking, although it would always be a derival of jitter. I mean, suppose jitter is a phenomenon which allows multiplication further down the line (like amplification does), the intrinsic jitter of a DAC might be a few ps, but right after that the error is amplified (I/V stage) and possibly is then very ready to be perceived.
Again this is the same apples and oranges, but assuming that jitter just cannot be measured once being in the analogue domain, what's left is at the incoming side of the DAC chip and any variation (jitter) there will be amplified right after that. Btw to my imagination this "ampllified" is always to be seen as resonance, or IOW repeating patterns, for the exact same reason dither was invented.

Lastly for now :
If it indeed would be so that jitter is to be measured at the analogue Out of the DAC (vs. the digital In), it is my suggestion to capture the analogue stream and compare it with the digital. Take a normal music file !
I don't think it is necessary to tell that hardly anything comparable will come from it, and even after careful level adjustment (the amplitudes of the output have to be calibrated with the amplitudes of the input) you won't get anywhere. Use a delta-sigma DAC and you won't even find compareable points, but with an NOS you'll still be shocked. But of course, nobody is even thinking of starting up a test like this (but I did) because we all know what comes from it : mess. This mess for sure starts at the delta-sigma capturing, but also will emerge at the (in-chip if so) I/V stage.

And still our expensive measurement devices are able to come up with figures at the ps level ?
How ??

Peter
(now tell me how many mistakes *I* made hehehe -> just trying to understand !)

Ashihara did an experiment indicating that random jitter is not audible below the 250 ns: http://www.jstage.jst.go.jp/article/ast/26/1/50/_pdf
According to Ivar Løkken: We can see that the audibility threshold decreases from 500ns at low frequencies to as little as 20ps at 20kHz. Especially when using formats or converters with high sample-rate this will be a major issue.
http://www.iet.ntnu.no/courses/fe8114/files/Report_audiodac.pdf

Roseval, thanks. And somehow so logical.

My problem is : I am quite confident that general jitter has more impact on the LOWER frequencies.
I am breaking my brains on the why, but all is in that direction.

This is similar to my observation that highres material (up to 24/192) discerns in the bass area (for the better). And I can't think of why (logic tells me, like Ivar Løkken's telling, that it should be the higher frequencies).

Peter

PS: Does anyone have a clue why I was contacted today by a dutch Audio Precision salesrep ? coincidences exist of course, but an out of the blue "let me help you with the right analyser for your problem" ... neah ...
:-)

"If it indeed would be so that jitter is to be measured at the analogue Out of the DAC (vs. the digital In), it is my suggestion to capture the analogue stream and compare it with the digital."

This is actually being done now, more and more. This is what John Atkinson does. Usually, a single frequency is put to the DAC and then the overtones and jitter contribution is measured with a RF spectrum analyzer. This is useful, but still no correlation to audibility.

Steve N.

Does anyone have a clue why I was contacted today by a dutch Audio Precision salesrep ? coincidences exist of course, but an out of the blue "let me help you with the right analyser for your problem" ... neah ...

But in the end maybe not so OT at all. What happened ?

The email I got from the salesrep was responded by me with the question how the man got hold of my name. After a few emails back and forth he could prove that I asked information about AP products on the AP website. I had a confirmation email of it ... hmm ...

Looking at the website, I did not recognize it at all, and besides that, this was from last January 18. I would have known *and* I just have an audio analyzer. So why ?

Then it sprung to my mind that the "book" from Julian Dunn only came with some kind of registration first. So it is this what I must have done on January 18. Haha, I was not making things up about reading mr. Dunn.

Well, by indeed pure coincidence the salesrep contacted me today, never knowing that I registered only because of getting that book. So yes, coincidences exist.

In the mean time I kind of told the man that J.D. hopping from Prism to Audio Precision, did not help AP much, because users don't get it. Yes, I fed him with Prism users (me) don't get it either. And thus my next question to the kind man (apparently still working after 11pm which was the time overhere 40 minutes ago) : .... "but might you be willing to let Julian Dunn contact me ... yes please !".

Now let's see what will come from that.
Maybe nothing, but if so I will let it know of course.

Peter

"Ashihara did an experiment indicating that random jitter is not audible below the 250 ns"

This would be interesting, except that random jitter is not the problem IMO, it's correlated jitter that is more audible. It's a brain thing, not an ear thing.

Steve N.

Julian Dunn died of leukemia on 23rd January 2003.
http://www.nanophon.com/tributes/julian_dunn_obituary.htm

If this is the case we better pay the shrink instead of buying your products

Ok, now I feel stupid.

On some kind of "positive side" this may explain why AES17 still is as it is. It wouldn't be so that he was the only one capable of doing such things ? But if so, something is quite some years behind now and can't catch up ?

I just want to add one point into this discussion to help, that we speak the same “language”.

I am well aware of the math about jitter etc, but we have to state clear about what numbers we are talking about. This can be either RMS or Peak-Peak value (depending what numbers you are taking for full scale signal), up to what bandwidth this values is measured, is this measured from bit cell to bit cell or frame start to frame start, or is this a FFT of the jitter. So 500 ps alone does tell nothing.

About the audibility of jitter, most of the different AES (Audio Engineering Society) articles I have read, attend and listen to, went into a similar direction (not all the same, but similar). Malcolm Hawksford has hold an very good AES session about jitter, where he played back the music with different kind of jitter and also removed the original content and left only the jitter content to listen to, in deed, very interesting.

Most jitter numbers are given as RMS values (so also with the school book orientated calculation), but the numbers I have given at the above mentioned post (this post came from) with for example 2.5 ps are the FFT values. And for all the maths out there, you can only compare these numbers with the RMS number if you take the conversation factor (with window scaling and at most the bin width) into account. (I used 44.1k sampling, 16K FFT with equiripple window (2.63 bins scaling)). So if I would have integrated over all of the 16K bins, I would end up in about 900 ps Jitter RMS when measuring in broadband up to 100 kHz.

But as I told at the very beginning of this post, only the value of this number does give me no insight about the kind of jitter, so I gave up just looking at this number. I am doing an FFT analysis and try to understand, what correlates with the sound. And I use the expression “try to understand”, because we are far from saying, that we do understand what we are hearing, we are all trying to learn to understand.

Juergen

PS: Sorry for my bad English, but I am not native English and also have not the time to read and correct my writings, until it is fluid.

... Juergen - thank you for bringing clarity to the discussion of jitter here on CA. Your english is very understandable, clear and concise. Don't worry about it too much, OK? .... Please just keep doing what you are doing! You have my full attention.

Thanks again!
-markr

You are perfectly readable

Gang,

Just a few things as I have spent allot of time over the last five years looking at this.

Another good article is the TN23 article also by Julian Dunn, available on the AP site as well. This tells you how to measure side band info like John Atkinson uses and develop a number based on these.

Presently we have 4 companies who make High End Audio testing equipment. Audio Precision, Miller, Prism and Stanford. Only Miller and Stanford actually give use a numerical value. Prism and Ap basically say it is too hard to give a number by measuring side bands.

The other way is open the box and use a Wavecrest DTS system which is the unit that every oscillator company in the world uses to measure jitter.

About 8 months ago I got the Wavecrest and have had the Prism for a couple of years. I bought the Prism for two reason's it has the best and deepest FFT analysis of any of the above and also because it is the only device capable of natively testing computer dacs and adcs.

With the Wavecrest you can basically measure any clock for jitter down to about less then 0.5pS.

An interesting thing happened when I was testing with both units. I found you could not fix jitter. For years you heard of people removing jitter from SPDIF inputs and now Firewire and USB with the use of upsamplers or reclockers. Well actually both of these methods merely reduce the jitter they cannot remove it and the more there is the more that get's through.

I first thought this true with some of John Atkinson's testing when a dac had multiple inputs. Most of the companies said that their jitter removal made all the inputs the same. Well in testing that did not prove to be the case.

I did some tests with my dacs and some of the others sitting around with the Wavecrest centered on the Word Clock WCLK. We use the Word Clock as the reference because it will have the worst jitter component of any of the 3 clock signals of Digital Audio (Master Clock MCLK, Bit Clock SCLK/BCLK). This is mainly because with each division of any clock the jitter goes up and the Word Clock is derived from the Master Clock/256. Ok... I planted the Wavecrest on the input and the output of some of the upsampler chips and sure enough with some SPDIF inputs the jitter (all done at 44.1) were between 512ps (Cirrus/Crystal) and 627ps (AKM) using the one with the best results. [FYI SPDIF streamed from Prism dScope III output] The output of the upsampler was only 69.02ps. This because it was derived by a new clock which would have better jitter than say one derived from an SPDIF input. I did the same using the TAS1020 and Adaptive interface which used the internal PLL generated clocks and we ended up with a result of 2838pS. [USB using the Prism dScope III software Vista Pro 2.4GHZ Core 2 Duo, 4GB Memory, SSD128GB] The output of the upsampler was again 69.02pS. So now we move over to the Prism and using the TN23 from above the SPDIF Jitter came out to 173pS using sidebands but the USB ended up being 296ps of jitter.

WHY? in both cases the Word Clock jitter into the dac chip was only 69.02pS.

I think the damage was done and not recoverable on the input to the upsampler. This would be the same for any anti jitter device.

To further look at this we sent some details out and measured some devices using Phase Noise. This is actually a better aspect for testing jitter than using the Wavecrest but it get's hairy as you need some real quiet environment. But you can actually test jitter at specified frequencies. The lower the better when it comes to true jitter. Like if someone says they have 2pS of jitter at 1KHz... well that's no big deal. If they say that it has 2pS at under 10Hz then they have something to say. The results found that actually the jitter using Adaptive mode was mainly caused by the protocol itself and the phase noise would not correlate to a linear jitter. The jitter from the SPDIF was correlated and mainly due to the rock solid output of the Prism dScope III and would not be the same for some transports.

In looking at the Adaptive mode output using spectrum analysis (12bit 10MHz sample rate Pico 3224 v6 software) we found that three things came into play for it's performance: 1ms timing of Adaptive mode, PLL generated clock, power supply noise. In the TAS1020 code the output clock is changed every four 1ms periods. This was pretty evident by the spectrum analysis. The PLL noise was actually allot higher and I feel this jitter would be removed by either an upsampler or a reclocker. The power supply noise especially what we call the 1/f noise will always lead to jitter in the system.

SPDIF part 2... Ok what about the jitter coming out of the computers. I tested a MacMini and a MacBook TOSLINK output into the Prism dScope III along with a spectrum analyzer and found these to suffer allot in the area of PLL generated clock noise and also what looked to be noise similar to the Adaptive jitter. Jitter from the Mini was 1607ps and from the MacBook 1103pS. I almost think the drop considering they have the same chip set was due to the speed and capabilities of the computer (MacBook 2.4GHz, 4GB, 128SSD it's a dual boot same as used above, Mini is a 1.6GHz, 2G, 320gB).

Anyways the point of all this is that we need to keep the jitter down from the beginning and not try and fix it later. That just does not work.

Thanks I have to get back to work now.
Gordon

Excellent references and explanations, everybody, this is a most interesting thread. Now I feel as though I'm learning something. :)

Hi Gordon, nice that you dig in with some additional comments. I originally didn’t want to go too deep into these points at this forum, and I wouldn’t have gone so far, if I wasn’t asked by “The Well Tempered Computer”. But you continued ever deeper into it, so I just want to drop my agreement.

Back in the early years of measuring jitter in digital audio I had begun with the LIM Detector from Ed Meitner, which gave same first insight. Then I continued with the HP Frequency Counter, similar to Standford Research, which gave some insight in the time distribution of different clocks.

But for me the most progress was done with the Miller Suite, using the J-Test Signal from Julian Dunn, and analyzes each sideband. This is on the analog side, IMO the best method to measure the Jitter. This can also be done with the HpW Works software, but without the numerical results.

But as I told in this post, I (or we) are still learning. At this moment for me I was looking to detect weather a digital receiver has a fractional clock divider (as the Wolfson 8804 has). This sounded a little be like SRCs do sound, but SRCs I can detect and they aren’t bit true, but fractional clock divider, I can’t detect and they look but true, because the “working” only on the master clock.

Let’s see, what the future brings. One other thing, I would like to get in contact with you directly concerning a different issue. So will you get my email, if I contact you via the main email address of your Wavelength Audio Company or what would be the best way to get in contact with you?

Juergen

Gordon, great that you shared. Thanks.

It would at least prove that my own DAC, rated at 2ps or so after the SRC indeed sounds a hell of a lot better with I2S opposed to SPDIF.

It would also prove that when the jitter is variated at the input, it is audible at the output anyway.

So, nothing new for observations I guess.

Gordon, any chance you can get me on the starting tracks for the abouts with phase noise ?
The way you did it, is it written down somewhere perhaps ?
Test files to load into the PC ?

I own a dScopeIII just bought for these matters, and I am ready to invest a whole of a lot of time in it.
May it help, I developed a 3D local positioning system (like GPS) for ultrasound, as well as simulation software for the same with lightwaves (0.1mm precision). So I'd know everything about phase shifts and what to do with them. But this is the other way around and it is not (happening) in electronics.

Please let know whether you rather do this by email, if at all.

Anyway, again, thanks for sharing !
Peter

using the TAS1020 and Adaptive interface which used the internal PLL generated clocks and we ended up with a result of 2838pS

I'm very curious to know what the jitter specs are when Adaptive mode is replace by Asynchronous.

Gordon - and Juergen,

It may seem strange but the geek in me (and I think, lot's of the rest of the CA population) is really into this thread. I cannot thank you two enough for taking the time to expand on this subject. No pressure. Just let it flow, when it flows. I want to LEARN! -- What Roseval asked.....

- markr

"I'm very curious to know what the jitter specs are when Adaptive mode is replace by Asynchronous."

This is entirely depend on a lot of things:

- The PC clock
- The TAS1020 clock and loop filter
- The implementation of board, power supply, transmission-lines and terminations etc. etc..

There is no simple answer to this. Either one can be very good IMO. I have async using my Pace-Car and synchronous using my Off-Ramp.

Steve N.

We are talking technology so implementations.
You can’t measure if there isn’t a full implementation.
I’m pretty much aware of it.

We are talking technology so there are many ways to Roma.
In this case I’m interested in the asynchronous way.
It is a new way (Gordon and dCs are the only ones who have developed it up to now) and I’m curious to know how it compare to adaptive mode.
The switch from synchronous to adaptive mode was made to reduce the jitter and it worked. Maybe asynchronous mode offers likewise improvements.
If it does, it will make adaptive mode as obsolete as synchronous mode

BTW: one can do the comparison in an almost identical setup: http://www.usbdacs.com/Concept/Concept.html

FYI : Here is another : http://www.bd-design.nl/contents/en-us/d168.html

I can't recognize that it sounds better because of less jitter. It sounds very different though than its predecessor (the TwinDAC+) which is basically the same DAC.

I'm afraid that DACs whith really less jitter at the output, will have it superimposed on certain frequencies (hence less randomly spread), which may be worse for net result than more jitter, randomly spread.

This is by no means based on measuring, but how my ears perceive it, combined with that it should be better ...

Peter

Edit :
To put this in some kind of perspective : By no means I wish to imply that this CrazyT is a bad DAC (it is top of the bill actually). The person behind it compared it with the Wavelength (I did not) and it is his perception that the Wavelength sounded "less good", FWIW. Again FWIW I trust that observation. The point here is, that the same person is as ready to tell that my own DAC is better again, and by no means this is a low jitter DAC at the output when run in the best sounding mode (which is around the SRC).
Now what ...
But there are so many parameters involved ...

...discussion is going nowhere. I don't think anyone has ever proposed that jitter doesn't exist or that it can't be measured. The only question, really, is whether or not it is audible in most gear, therefore necessitating its further reduction, and justifying that reduction's cost.

Have any of the manufacturers/developers/enthusiasts in this discussion ever conducted objective (ie: blind) listening tests that clearly demonstrated that a difference could be heard, beyond the margin for error, in exactly the same system with any of this advanced jitter reduction technology being used? No subjective judgment, no evaluation, just the simple clarity of discovering if the blind listener can identify the device in the signal chain.

Before you object with the comfort of the listening conditions and the sophistication of the listeners arguments, have you conducted such tests under the conditions you think are appropriate, with listeners you believe are sufficiently sophisticated? Do you have plans to do so? Will you publish the results of those tests here and on your websites, regardless of the outcome?

Tim

The only question, really, is whether or not it is audible in most gear
As I don't own 'most' gear and being a bit egocentric I’m only interested if it is audible on my gear.

Another implementation of asynchronous USB indeed.
But at the expense of using dedicated player software and limiting you to WAV/FLAC at 16/44.1 only

We've built many DACs and nine different CD Players since the early nineties and more recently we've compared SRC, which removes jitter for an extra two quid with digital receivers that reduce it to insignificance and we've not been able to hear or measure significant difference. This is as predicted by the chipset manufacturers and by science.

For these reasons, I quoted the maths (which was ignored) to illustrate at what level jitter noise would be worse than 100 dB, a figure that is much better than most preamps and power amps and miles better than the noise in passive speakers. These came from Dr Chris Smith of Roke Manor, which I thought good because he has no commercial interest and great deal of knowledge as Research Engineers tend to.

So to sum up; Jitter produces noise at very low levels that can be measured, we know how much produces more noise than -100dB and we know that most people's hi Fi systems already have more noise than this. So the discussion is probably of no value in a practical sense, although it has obviously been enjoyed. Tim is probably in a better position with headphones to hear problems that people with a separates system and so are we because our active systems have much a lower noise floor than the older technologies.

I think I ought to stress that there are not significant choices available to someone wanting to make a DAC, all he can do is follow the Chipset manufacturers applications notes. Some provide a PCB layout and all an evaluation board with measurement points etc, so that companies like us can avoid making mistakes and compromising performance.

As I've said many times, the problems have always been ultra sonic spuriae and RF. Stuff above 20 kHz that upsets some amplifiers. Years ago this was only about -30 dB but now with a good DAC, it's better than 100 dB, however some are out there with -75dB and there are still amplifiers that are profoundly affected. Therefore DACs can sound different with different systems, more different than anything caused by jitter, but for some reason and despite this being the biggest single factor, no one seems bothered by it.

Ash

"Most? Mine!

The only question, really, is whether or not it is audible in most gear
As I don't own 'most' gear and being a bit egocentric I’m only interested if it is audible on my gear."

Agreed, but I evidently wasn't clear. I was looking for a way for manufacturers to substantiate their claims, something I'd expect them to welcome. Something quite useful to the audio shopper. And the question wasn't audibility ON most gear, but audibility IN most gear. Or any gear, really.

This would be a piece of cake for someone manufacturing outboard equipment designed specifically to reduce jitter -- same source, same DAC, same system, same listeners...magic box in/magic box out. Blind test. On the most sophisticated high-end system or a Yamaha AV receiver. Or all of the above. All the discussion of picoseconds would be moot.

It would be less straightforward for the DAC manufacturer who dares to build jitter control into the core device instead of building another groovy box for us to buy, power and interconnect, possibly creating much greater opportunities for noise than the box could ever eliminate, but it isn't all that hard: ABX the old model DAC, without the supersonic jitterphonic bat-ear reduction circuit against the new one. Can a difference be identified in a blind test more often than you'd get heads in the toss of a coin? No? Back to the drawing board. Yes? NOW is the time to judge the difference subjectively and reel out the poetry and the statistics.

It's really very simple, in concept and execution. It is also very inexpensive. The fact that it is obviously not being done (obvious by evidence of its complete absence from these long jitter discussions) speaks volumes.

Tim

... those last 5 posters were either Gordon or Juergen, were they?

sheesh :-(,
markr

It's a shame. As a couple have already suggested, they'll probably continue the conversation in private and we'll be left with the usual merry-go-round of drivel, until the thread dies or Chris kills it. Ah, well, maybe I can buy a book.

IMHO, YMMV, etc, etc

.. Juergen didn't really want to expand on this here anyway. I don't blame him - the web is the place to just post a paper, not really one to discuss it, UNFORTUNATELY.

yeah, IMHO, YMVV, ad nauseam....

- markr

I think the saddest thing is that this thread is just about getting prepared to measure beyond current possibilities. So ...

So I think these two guys should really stop nagging about some positives here, and most of all they really should stop interfering with every other thread with their by now long known positions, and stop spoiling the fun for everyone.

Trolls ... (which is what it really comes down to after 1046 of such posts, sorry ...)

Peter

Originally I started this post to just answer one question, where I wanted to clarify a little bit more specific, where the jitter numbers came from, when I compared different software drivers, which leads to different jitter and also different sound.

Justifying jitter is a little bit like “fighting against the wind”, because there are so many people out there who just believe in math and the thinking: what can’t be measured can’t be heard. That was also my thinking when I finished my electrical engineering.

I am happy that besides being an audio engineer for over 25 years, I also make music for over 30 years and actually do some classical recordings and own a small recording studio (for 5 years) so besides the learned math and the ability to use measurement instruments a have at least a small ability to be sensitive to music.

So I am playing around a lot with sound and music and most times it is the hearing of something as the starting point, followed by endless “research” to try to answer the question, why is this audible and what is is the most appropriate measurement tool to find an answer.

And also with my classical recordings, I am surprised and also nervous, that when I have the raw recordings, to what extent some points are audible. Really, this sometimes totally pushes away what I was told during my engineering education, what would be audible and what not.

So really, I originally do not what to open this long thread, because discussing about jitter is a similar long story as discussion about THD, but in both field I wished, that if a number is raised, one should really specify the conditions.

So I wish all of you a nice day and follow your heart, when listening to music or justifying audio quality. I have better results, in doing this, than following my “brain”, when hearing to ABX (ob man, I wish this explanation is clear (because I am not native English)).

Juergen

.. Thank you Juergen (AGAIN). This is what I thought we might hear at the end of things. I am really sorry that some here felt it necessary to advance their personal agendas ( you SHOULD know who you are.........) and I have to say that I totally respect your position. I pray that you have total success with your adventures on and on into eternity. Thank you again for taking the time here to delineate some your experience here!

When using playback software with native EMU ASIO support (Wavelab, Cubase, Reaper, JRiver Media Center, Foobar with ASIO Plug-In), I got bit perfect true data. With theses software I got jitter values of random nature with approximately 2.5 ps

This type of in formation is rare.
Most people simply claim they hear a difference (big of course, being blown away seems to be an absolute minimum).
I’m glad we have somebody on the forum who is able to establish if the software allows for bit perfect data and can measures its impact on jitter.
What we need are ‘facts’ especially from guys like you who know that a number is a fact but not an explanation.
Keep up the good posting (and stop complaining about your bloody English).

There are a couple of manufactures nuking any tread with promotion of their products.
These are the brands that will never make it on my short list.

PeterSt wrote:
So I think these two guys should really stop nagging about some positives here, and most of all they really should stop interfering with every other thread with their by now long known positions, and stop spoiling the fun for everyone.

I totally disagree. I originally joined this forum believing it would be beneficial to a beginner approaching computer based music systems and provide helpful and sensible advice in getting the most out of such a system. tfarney, Ashley James and a few others here, have done just that, providing sensible, well intentioned advice that won't break the bank and provide for a well rounded and well constructed computer based audio playback setup.

In these times of hardship for the many and the fact that being frugal is a necessity for the majority, sensible and practical advice is what's required. This, both Tim and Ashley have offered in spades. The positivity found here at CA, IMO, comes from THEIR posts and not that of others, who continue to ramble on about the same old, no doubt driving away any newcomers to the site who already find the whole experience of computer audio daunting.

Basically, through all the fog of jitter, Tim and Ashley have been a beacon of light that has allowed me personally, to find my way without too much issue. The amount of threads that have been started in relation to jitter make me question what the actual agenda is here? CA is going to suffer if it continues IMO. :(

PeterSt wrote:
Trolls ... (which is what it really comes down to after 1046 of such posts, sorry ...)

THAT comment is TOTALLY out of order. :(

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djp