The threshold of audibility of phase noise in ADC and DAC clocks is a fairly contentious issue in the HiFi and audiophile world. Some sources claim that jitter is clearly audible at low levels, and some claim that high levels of jitter are inaudible. The literature describes several tests, many with conflicting results.

One of the chief difficulties in testing the audibility of jitter is that it requires a complex hardware setup, which means that many listeners would be required to be present for an time consuming (and expensive) on site test. Over the last couple of months I have been thinking about organising a distributed listening test to look at the audibility of jitter in audio applications, based on algorithms for simulating the effects of jitter on signals. These algorithms are fairly well described in RF and telecomms engineering literature, and would be interesting for comparison purposes.

The kind of thing I have in mind is this:
Use samples which are accepted to sound good -> simulate jitter -> perform listening tests -> perform more tests at different levels of jitter depending on results

The purpose of this thread is to get ideas of the Hydrogenaudio community about performing these tests. Some of the things I would appreciate input on are:

• Would such a test be useful
• Would the results of such a test be valid?
• What sort of test procedure would be best?

I would suggest you begin with a working definition of "Jitter", which you could post to the HA Knowledgebase. Is jitter "phase noise" ? What does it sound like? What does it do to a sine wave that goes through ADCs and DACs ? Does it matter at all ? What is the order of magnitude of jitter in PCs? and in high-end audio equipment?

There is a good article about jitter @ Digital Domain:

http://www.digido.com/modules.php?name=New...icle&sid=15

I'm looking forward to see such test... How many of you believe in "jitter effect"?

A good point. I would go with a definition of Jitter as the high frequency component of the time interval error in the clock signal. Where the cutoff of "high frequency" is must be chosen, as must the spectrum of the modelled noise. This spectrum is one very hard problem - it's likely to consist of a variety of intermodulation products, PSU noise, other noise, harmonic distortion and vary widely between different pieces of equipment.
Jitter spreads the spectrum of a sine wave out, effectively by convolution with the spectrum of the jitter. I have no idea what it sounds like, except in some extreme cases. And whether it matters at all is one of the questions I am hoping this test will help answer.
I have no idea about PCs, but total jitter figures for midrange HiFi stuff is generally somewhere between 0.1ns and 100ns - a three order of magnitude range.

Having just read the article you linked to, I found it a little too much "audiophile"-minded and not very clear about the technical issue of jitter. Quoting: " ...The sonic results of passing this signal through processors that truncate the signal at -110, -105, or -96 dB are: increased "grain" in the image, instruments losing their sharp edges and focus; reduced soundstage width; apparent loss of level causing the listener to want to turn up the monitor level, even though high level signals are reproduced at unity gain..."

I know quartz oscillators have a small amount of phase noise (jitter).
How much does the phase noise in a 12MHz quartz oscillator driving a DAC at 48kHz affects the final waveform?

100ns sound a little bit high (for a 10MHz quartz oscillator that's one full cycle) but OK, let's work with that.
How would a 10MHz signal modulation of an audio signal ranging from 20Hz to 20kHz affect the signal?
It seems all the intermodulation products would be way above the audibility range, no? Or am I missing something?
I think a typical figure in an PC-grade audio codec would be 1 ns jitter (approx. 1% jitter in a 10MHz clock), which would represent approximately 0,005% jitter in a 44.1kHz clock (if my napkin calculations are right).
My a priori guess is the ABX test would be unconclusive...

(edit) I just checked the ALC882 codec datasheet and the 24MHz bit clock input is specified with a maximum jitter of 2ns.
The S/PDIF-OUT jitter is specified at 4ns max (6.144MHz clock). I think the S/PDIF-OUT clock is PLL generated internally in the codec.

Yes it would be useful. Jitter is the main cause invoked in order to explain the alledged sonic difference between the digital output of a 50 € DVD player and a 10,000 € dedicated digital drive.

The results would be valid if several conditions are fullfilled.
On top of my head I can think about
-The kind of jitter. Some kind might be audible, some kind inaudible. We must choose the kind of jitter that should cause the biggest audible effect.
-The test must be performed on a system whose jitter is small compared to the tested amount of jitter.
-The samples used must be sensitive to jitter.

A full scale 20 kHz sine with jitter introducing a 3.5 kHz artifact should be the most sensitive combination.

The most difficult part in this test, in my opinion, will be to get comprehensive jitter analysis from consumer CD players. Especially if jitter is signal dependant.
Jitter should also be analyzed directly at the clock output in order to account for hardware induced jitter. Some claim that heavy error correction or tracking corrections burdens the power supply, which could in turn affect the clock stability.

I think you meant "measured" in the phrase above. However, measuring jitter is not a trivial thing, whether we are measuring 0.1ns or even 10ns.
I would dare say that none of us HA readers has the equipment to directly measure jitter with any degree of precision.

All three of these are difficult problems - and would need careful test design to handle. Possibly starting with something along the lines of "is 1us of jitter audible? in test samples? in music?" and moving down to the harder problems if that test has a positive result would be a good plan. I don't know what the right solution is.

I do have access to such equipment when I am at University, and hopefully I can get another student to teach me to use it. Real measurements would be invaluable but, as Pio2001 says, if the jitter is data dependent (and it seems to be in many cases) it becomes a much harder problem.

Gigapod - you sound like you know quite a lot about jitter/phase noise - thanks for your input so far. Keep it coming

Cabbagerat, I think it's very interesting if you can measure jitter in audio equipment in a well equipped university lab, and later determine through listening tests if it has any audible effect. I think it's always satisfying to debunk a myth with some solid experimental data, and I am highly suspicious of the whole jitter audiophile shebang.
Actually a long time ago I studied precision clocks and did some superficial documentation gathering on quartz oscillators. I think nowadays even the cheapest CD player (like the ones costing 15 euros that you can buy at the supermarket) uses a quartz clock base, because quartz is so cheap. Typically I think the jitter figures will be very low, because even with 25% jitter (highly unlikely) at 10MHz that's only 25ns, but I have no experimental data to offer to support that assertion (lack of equipment, lack of time and vague motivation).
If I may offer some guidance here, I would say, take a single piece of audio equipment (e.g. a normal CD player) to your university, find the quartz oscillator, and check the jitter at the buffered output of the quartz oscillator (not directly at the quartz leads, obviously). Of course schematics would help a lot, but if you can't find the schematics it shouldn't be too difficult to find the quartz oscillator circuitry.
BTW at 10MHz a few pF will significantly affect the waveform, so I recommend a FET active probe with 1GHz bandwidth.
If you find out the jitter is of the order of 1ns, I wouldn't bother with the listening tests...

I agree with your assessment that the article is too audiophile. Here is a more scientific approach by someone who has designed ADC and DAC for MRI and military devices and makes studio quality ADC and DAC’s.

http://www.lavryengineering.com/white_papers/jitter.pdf

unfortunately jitter is very real.

Thank you, that was an interesting paper.

I agree with you that jitter is real. However, the author of the paper, Dan Lavry, used FM modulation of sampling rates to simulate jitter. This is definitely not an accurate simulation. Jitter is also called "phase noise" for a good reason: it has a noise profile.

Just a note: your inexpensive PC codec uses a 24MHz clock to drive an internal PLL that itself drives the sampling rate of the A/D and D/A converters. It's next to impossible to FM modulate that internal PLL...

The second graph on page 6 is a more accurate representation of real-life jitter effects, and you can see they are below the noise floor of the signal.

Cabbagerat suggested in a post in another thread that most manufacturers don't quote jitter figures for their audio equipment gear because the effects of jitter can ultimately be measured in noise and THD figures. I think this is correct; I would only add that providing jitter figures directly would be a) difficult (because jitter is difficult to measure and where exactly do you measure it ?) and b) meaningless, as you can only hear the side effects of jitter as added noise or distortion.

Nowadays noise and distortion figures hover well below audible levels, so I confess I am not too worried about jitter, even though it's real... in a sense.

The problem here is that THD does not measure how bad the distortion sounds. In general, higher THD is worse, but a tube amp fan will tell you that 2% THD doesn't always sound bad. That's well accepted, but it means that audiophiles (and others) will claim that the distortion introduced by jitter is particularly bad.

Lavry's paper is fascinating. However, I agree with Gigapod that his treatment using narrowband noise and FM modulation is too simplistic to be an accurate representation of what happens in reality - the phase noise on quartz oscillators does not resemble the test signals he used. It's the best audio-specific treatment I have read.

Imho when THD is below 0.01% it matters little whether it has only odd harmonics or only even harmonics: you just can't hear it anyways ("audiophiles" will certainly disagree...). ABX testing of power amps (Transistor x Valve) seems to support this.
Similarly, if the added noise is below the noise floor of a recording. However, note that noise spectrum analysis is required: changing the noise spectrum profile can be audible in some circumstances.

Getting back to the paper by Lavry:
FM modulation of the sampling rate -> I.M. distortion byproducts.
Phase noise (jitter) in the sampling rate -> noise byproducts.
Nothing new, really.

Unfortunately Lavry provides very little in the way of experimental data and perhaps I missed it but I didn't find any mathematical analysis. I am sure one can easily deduce a mathematical formula that will relate the amplitude and frequency of a signal, jitter (or FM modulation) in the sampling rate and noise/IM distortion levels in dB (or %).

I wouldn't be surprised if normal jitter found in commodity-priced audio equipment would have a noise effect at the quantization noise level in 16-bit CD audio. 24-bit audio probably requires a lot more care with jitter. But then 24-bit audio requires a lot more care with everything, doesn't it?

Jitter matters only during slope of the signal. In severe case jitter is causing LSB error. By taking fastest slope, and limiting jitter induced error at 1 LSB, we can find out max acceptable jitter in terms of technical perfection.
For 16-bit system, to keep sample error below 1 LSB, 15KHz signal of 0db can tolerate no more than 324ps of jitter peak.

In other words, we can change the problem into asking how many LSBs in error can we get away with without being audible. Obviously, it does matter whether these errors are white noise-like, signal correlated or alien.

I suggest few papers from:
http://www.essex.ac.uk/ESE/research/audio_...blications.html

C41 IS THE AES/EBU/SPDIF DIGITAL AUDIO INTERFACE FLAWED?, Dunn, C. and Hawksford, M.O.J., 93rd AES Convention, San Francisco, preprint 3360, October 1992

C134 JITTER SIMULATION IN HIGH RESOLUTION DIGITAL AUDIO, Hawksford, M.O.J, 121st AES Convention, San Francisco, October 2006, paper 6864

Whoaaa! Some serious math here. That'll take some time to digest, that one!!!!

Thanks wimms!
Should I send you the coffee bill or are you willing to lead us through this paper?

(edit) Your sig -> Very funny!!!

The second one of those paper is fascinating. It's the mathematical treatment of jitter that I have been looking for for a while. One excellent quote from the paper is:


Much of the complex maths in the paper is derivation of a simplification of the standard bandlimited interpolation formula, with added jitter factors. This optimisation might be necessary for real-time processing, but for offline processing simple truncation can make performance of the unsimplified algorithm acceptable on modern processors. Sections 7 and 8 (one analyzing LPCM and one sigma-delta) are interesting, as are his conclusions.

For all its complex maths, there's an unfortunate lack of audibility data in the paper, or even a summary of previous such data, to indicate how 'important' this ends up being to the consumer .... so what does it do towards addressing the subject of this thread?

Because - unlike for example harmonic distortion - you can't hear clock jitter directly, as explained in the various papers, you can only hear its effects on the data as it gets converted back from digital to analog.

And as I wrote in my rant above (sorry I am repeating myself, must be getting old), the effects (basically distortion & noise) are usually much below others (for example, noise and HD in the microphone that is used to take the recording in the first place, or quantization noise in 16-bit recordings, etc), even in commodity-grade audio equipment.

well, let me offer this 2005 paper for consideration, then


http://www.jstage.jst.go.jp/article/ast/26/1/50/_pdf


Audibility threshold for timing jitter, for 'golden eared' listeners in a two-alternative forced-choice paradigm using their preferred listening environment and samples: 250 ns.

What was the spectrum of the jitter? That's really a key question, you know.

For the Dutch speaking and jitter interested readers it might be interesting to know that the Dutch AES has planned an evening about "digital clocks" (wordclock distribution in the digital studio). Non-members are welcome too in Utrecht on 25/01/2007. More info can be found here.
Jitter is one of the topics that will be discussed and demonstrated.

For tests, why not generate a simple program that can simulate jitter?

The proposed program simulates an analog source represented by in.wav, which is AD converted using a non jitterfree convertor

How this can be done (I think):

command format:
simulatejiitter.exe input.wav jitter.wav output.wav

jitter.wav contains values, that determine how much time-offset should be created for the corresponding input.wav sample. Jitter.wav can be given any shape/spectrum.

For instance. jitter sample=0 equals no time off-set, and jitter sample = 32767 results in max positive time offset (for instance 1 complete sample period)

As an example, output samples can be calculated using formula:
out[n]:=in[n] + (in[n+1])-in[n-1])*jitter[n]/65536


For simplicity, this example uses a simple lineair interpolation for approximating signal value near sample
in[n]. A 2nd order function would be already be more accurate.

Best solution would be infinite FIR/ nyquist lowpass, that can calculate any, in between sample value, but that's not what I call simple!

If out.wav is 24bit format, rounding errors are kept out.

Correlated jitter is probably worse than random jitter

High-frequency jitter is probably worse than low-frequency jitter

Any simulation of jitter for detectability threshold will be limited by the playback jitter performance - typically unknown.



By doing detectability measurements, coupling those to models from good measurements, one should be able to decide this once and for all.

Any progress in finding jitter attenuation in "typical" DAC/surround receivers?

What about inserting a RLC network along an spdif path to provoke real spdif-jitter performance of various spdif receivers?

-k

I have a MATLAB program which does this, based on bandlimited interpolation. I would be happy to post the code when I get my home PC out of storage in a few days.

Does anyone else have any additional actual published papers on this topic of the audibility of jitter or listening tests?

What journals cover this topic, if any?

My focus is to understand:
a) given a synthetic jitter profile, is it audible using DBT?
b) given a real jitter environment, is it audible using DBT?
c) can you DBT the difference between toslink and coax?

I have been looking into the basis of the audiophile belief that toslink is broken due to too much jitter and the implicit belief of the audibility of very small amounts of jitter. Given the intensity of the belief, perhaps where there is smoke, there is fire, even though this belief makes no sense to me based on my understanding of jitter and its impacts in this application space.

As far as I understand jitter impacts two things related to audio reproduction:
a) at the level of the synchronous bit transport, it influences bit error rates
b) at the level of the DAC, it causes errors in the recostruction of the waveform. The outcome of this is essentially higher noise and distortion, i.e. you just get a bump in the noise floor related to the waveform being reconstructed - its correlated to what is being reconstructed, which is a wrinkle - and possibly spectral aliases being created.

Detailed studies of this understanding are also appreciated with actual waveforms & spectrum views. I also want to be clear I understand the techniques to dejitter a clock, including reclocking and most importantly, buffering. The issue is about impact, not about repair or avoidance.

I don't want any more audiophile "received wisdom" on the issue of jitter, I have received lots of that.

Two papers keep getting cited at me in this discussion. One is the 1992 Stereophile article on jitter, which had a rebuttal in the Audio Critic and is not my idea of peer reviewed journal article (but has massive audiophile traction). The other is a 1992 AES paper, Is the AESEBU Digital Audio Inteface Flawed?, 93rd AES Convention, San Francisco. Both of these seem somewhat, well, dated, to me.

The paper previously cited in this thread has actual audibility testing, and appears to set an audibility threshold of jitter of around 250ns.

Papers without audibility studies set the threashold far lower. For example, Dunn's 1992 AES paper claims an audibility threshold of an astonishing 20ps at 20 KHz, based on his 1991 paper "Considerations for Interfacing Digital Audio Equipment to the Standards AES3, AES5, AES11, Proceedings of the 10th International AES Conference, 1991" (paper not yet found online). As another data point, "A Digital Discourse, Dr. Malcolm Hawksford; HiFi News & Record Review Feb,April, June, Aug, 1990" claims a peak jitter threshold of 400ps (also cited by Stereophile). I have not found the actual article yet, just citations and quotes.

Is Dunn's audibility curve an analytic derivation, or an audibility study? Dunn's curve of audibility is widely quoted. Anyone have a copy of this paper?

Others cited, but not yet found (I hesitate to pay the $20 AES paper fee) include "Eric Benjamin and Benjamin Gannon, "Theoretical and Audible Effects of Jitter on Digital Audio Quality", Preprint 4826 of the 105th AES Convention, San Francisco, September 1998" and "The Effects of Sampling Clock Jitter on Nyquist Sampling Analog-to-Digital Converters, and on Oversampling Delta-Sigma ADCs, 87th Convention of the Audio Engineering Society, October, 1989" (also cited by Stereophile).

There appears to be tremendous discussion of jitter measurement, but little understanding of what it actually means. A lot of this appears to me to be very old work and at best analytic, not audability based. None of it considers modern techniques to break the end to end synchronous clocking paradigm although some of them hint at what is now common practice in the telecommunications & Internet space.

Journals that publish in this area or references to further studies would be welcome.

(Zster, thanks for the reference to the Lavry overview paper. It is a useful overview document to help explain the impact of jitter in a well written way, and review some of the more modern methods to dejitter signals.)

A small group of experts on this subject (some from this forum) is currently investigating the audibility of jitter.
The idea is to develop a jitter simulation application to enable testing (listening) without a low-jitter DAC.
Alternatively an ultra-low jitter DAC and ADC are available for DBT.
There's no strict time schedule for the project. I'll post updates in this thread.

If anybody is interested to do a listening test in France, I can provide a jitter modulator JM1 from Prism and a QSC ABX comparator for DBT. Get here the manuals.
We have to verify that the setup and ABX box doesn't add any jitter (I can get jitter measuring tools).
It's not as pratical as a software simulation but could be a complementary approach.

Is this the wrong place to ask "What IS Jitter and How does it affect people with digital music?" And I feel a little stupid asking this question having an EE background. The most simplistic description of Jitter is, "A Time-based error where a digital clock is not quite accurate,"

Now I've never really known whether typically when people say that, they're referring to an instance where a clock signal was held higher (or lower) longer than it was supposed to, or if a clock is operating at 44.2kHz instead of 44.1kHz.

Most of what I read about jitter issues involves ADCs, which although important, isn't all THAT important for people who prefer to listen to their music (as opposed to folks interested in recording it).

For digital transport, jitter can be a serious killer, but it has to be pretty damn serious before it affects this layer. Basically your data lines and clock lines have to be out of sync. By a lot. If people want to simulate this, I'd recommend writing a program to either randomly delete samples or randomly rewrite a sample to the sample immediately preceding it. My guess is that this happens as often as cosmic ray memory errors.

Since most applications for people with digital music is in the playback, only the DACs are relevent, and the question must be raised, "What are typical settling times for DACs?" But I think it also raises another question. Is Jitter at all relevent to people who don't know what the settling time of their DAC chip is?

I've read this and found it largely reflects the findings of many friends. Wonder if this can be repeated exactly with a different group?

Single number jitter specifications don't say much if you don't know the jitter spectrum.
If the effects of jitter are audible at all, it will be interesting to know which part(s) of the spectrum is/are responsible.
Measurements of real world equipment indicate that the jitter spectrum can't be assumed to be flat. Jitter simulation should be able to reproduce this behavior.

If the the sampling clock for the DAC (whatever rate it has) has a high jitter (and I guess this thread
is about what is "high"), the sampling is not equidistant in time anymore, thus the sampling theorem
is violated. This has an effect on the spectrum and the reconstructed signal. Question is, when is it
audible...?
The effect of jitter being so high that actually data is lost is not part of the discussion.

250ns and 400ps are not contradictory. One is saying "we've tested it subjectively - at around 250ns it starts to become audible". The other is saying "from first principles, if you keep it below 400ps, for the most sensitive possible signals, it will have a smaller impact on the signal than the limits of the system itself (i.e. the sample rate/bandwidth and bitdepth)".

These are two different approaches to audio engineering. One says "we can make it as bad as we want as long as no one can hear it". The other says "we will make it as good as we can to the point where this part can never be the limiting factor".

The real world has to sit between the two. You can't engineer something so that nothing is the limiting factor! You have to have some understanding of human ears to know when to stop improving everything (or be permanently depressed that nothing is good enough).

Conversely, you can't make everything "as bad as it can be before it causes an audible problem" because if you chain all these separate things together you can be fairly sure that you will have an audible problem at the end!


I like the idea of an experiment, but I don't see how a typical HA public test can work or be valid. We're all listening with unknown levels of jitter.

It would be like testing the audibility of -120dB of noise while we're all listening with soundcards which add noise at somewhere between -108dB and -60dB. Not hearing the -120dB of noise through these sound cards proves nothing. The same is true of jitter.

Cheers,
David.