Jitter - gone for good?

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audio_tony

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#1
I have been participating in a thread on another forum. In this thread, a poster states that jitter is no longer a problem. He also states it can be eradicated by resampling. Is this correct?

It's been several years since I did my theory on CD playback, and having worked in IT for the past 18 years or so, electronics has become more of a hobby than a profession.

Can anybody shed some light on the subject please?

Thanks,
Tony.
 
#2
audio_tony said:
Jitter - gone for good?
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Nah, I'm still here 😉 .

I have been participating in a thread on another forum. In this thread, a poster states that jitter is no longer a problem. He also states it can be eradicated by resampling. Is this correct?
Click to expand...

I say he'd be wrong. Maybe some forms of jitter may be reduced by reclocking, but any manipulation may introduce new jitter. At least that's how I understand it.
 
#4
The countervailing view is summed up pretty well on the jitter.de website...

http://www.jitter.de/english/how.html
If you use an ASRC as a jitter attenuation device, the jitter at the input of the ASRC will be distributed into the output signal-data, and what was simple clock-jitter at the beginnig is now forever glued your digital audio signal, it has become something comparable to sampling jitter.

(I have to add here, that the designers of ASRC chips try to minimize the susceptibility to input clock jitter, and this is the reason why manufacturers use ASRCs in order to reduce jitter. It may sound a little better if you add an ASRC but the price you pay, is to loose the original sound quality that was contained in the input signal to the ASRC.)

The clock jitter at the output of the ASRC might be less, but the signal is not the same (the data has been altered), it now irrevocably "contains the input jitter" and the initial signal quality is degenerated.

Unfortunately sample rate converters are praised as jitter attenuation devices by some manufacturers, but they are not.
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#5
That's a bunch of horse ****.

An ASRC does not modulate the input clock jitter into the output stream, except below the audible spectrum (< 10Hz). An ASRC only uses the input clock to estimate the IN/OUT clock ratio. This calculation is performed with an extremely low passband. You can't hear it.

It's possible that the input clock will somehow be modulated into the rest of your system, for example through the power supply rails. This, however, is a system design issue and not a feature of an ASRC.
 
#6
Jitter is the current 'bête noire' of that section of the community which in restless discontent seeks to discover less and less plausible reasons not to enjoy what you hear.

Originally posted by Oscar Wilde
...the unspeakable in pursuit of the inedible
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Samples are just a string of numbers, fixed once recorded. You have no control over the jitter in recording.

How much jitter is present at playback time depends on the exact technology employed to deliver the string of numbers to the DAC, and where the timing information comes from. It is almost certainly entirely possible to produce a cheap playback system with current technology where the jitter is below the threshold of audibility.

The arguments are all about whether particular systems meet this criterion. As technology advances such considerations will eventually be squeezed to extinction.

Please, Lord... make it soon.

w
 
#9
I don't know about fetishes, but it is certainly possible to achieve phase noise sufficient low to claim 24 effective bits during playback, and I don't know anybody who believes that the 24th bit is audible. Therefore, in my opinion, jitter is a solved problem.
 
#12
This is very similar to what Dan Lavry claims with his renowned DA-10 DAC. There was an uproar a while back when people found out Dan called his "Crystal Lock" implementation in this DAC using AD1896 SRC, rather than his actual crystal lock hardware cuicuit of his more expensive DAC's.

-David
 
#13
audio_tony said:
I have been participating in a thread on another forum. In this thread, a poster states that jitter is no longer a problem. He also states it can be eradicated by resampling. Is this correct?

It's been several years since I did my theory on CD playback, and having worked in IT for the past 18 years or so, electronics has become more of a hobby than a profession.

Can anybody shed some light on the subject please?

Thanks,
Tony.
Click to expand...

have a look at the Jan posting in Stereophile and you will not think so.
 
#16
ok so the jitter.de site is pushing it a bit. As penance I started reading an old but good thread in the archives on ASRC:

http://www.diyaudio.com/forums/showthread.phps=&threadid=28814&perpage=25&pagenumber=1

Someone posted a link to an older version of the jitter.de article I linked to, and thread starter "werewolf" made the following comments:

http://www.diyaudio.com/forums/showthread.php?postid=342727#post342727
A couple simple, logical points :

1. Does the ASRC completely ELIMINATE all forms of jitter from the incoming (e.g. S/PDIF) stream? NO. It substantially, heavily FILTERS the jitter ... but does not eliminate it altogether.

2. Where do we "find" the residual jitter? Well, it certainly isn't on the output clock ... that's ultra-clean (by definition), derived from a local crystal oscillator. Only leaves one place ... the output data. So must we conclude that, after being heavily filtered, the incoming TIMING jitter is somehow "mapped" or imposed upon the DATA? The answer is yes.

HOWEVER (actually, there will be alot of "however's" in the upcoming posts) ... where I must take issue with the above quote is with it's use of the phrase "simple clock jitter". There's nothing "simple" about clock jitter, if the intention is to imply that clock jitter is "less harmful" or "easier to eliminate" than what ASRC does with jitter.

Want to eliminate jitter completely? Slave the source (transport) to the DAC. Want to eliminate jitter and remain compatible with S/PDIF? Good luck ... it WON'T happen with PLL-based clock recovery, and it WON'T happen with ASRC. However, ASRC is still superior to PLL-based clock recovery, for reasons we shall soon see
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#17
spzzzkt, it is easy to refute this argument with logic. Suppose you have a DAC and transport in perfect word clock synchronization. The incoming data rate exactly matches the DAC local oscillator. However, assume that the DAC oscillator has zero phase noise and the transport oscillator has a huge amount of phase noise.

Now, assume that your DAC is using an ASRC. The long-term IN:OUT word clock ratio will be exactly 1:1, so the data will pass through the ASRC completely unchanged. The output will be identical to the input. Therefore it is not logically possible that the jitter on the input clock is mapped into the output data stream.

The jitter.de people simply don't know what they're yammering about.
 
#18
theAnonymous1 said:
Subjectiveness aside, how much jitter in a "normal" system is audible to a "normal" person? What does it sound like?

Would a DAC consisting of DIR9001/SRC4192/PCM1794A have low enough jitter for a "normal" person to not hear it?
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The audibility of jitter depends on its frequency distribution as well as its magnitude. The music you're listening to also can affect how audible jitter is, presumably.

I'm writing up some MATLAB/Octave code right now which encodes jitter into .WAV files. I'll probably make a thread on it later when it's done, though I'm on vacation right now and trying to avoid MATLAB and DSP as much as possible 😀

Overall, I wouldn't say jitter is gone for good. No matter how you move data from box to box or run it or what digital process you run it through, at some point in time you'll have a clocked DAC performing the D/A conversion. And obviously, if there's any jitter-created distortion in the recordings you already have, you're stuck with that.
 
#19
jwb said:
Now, assume that your DAC is using an ASRC. The long-term IN:OUT word clock ratio will be exactly 1:1, so the data will pass through the ASRC completely unchanged. The output will be identical to the input. Therefore it is not logically possible that the jitter on the input clock is mapped into the output data stream.

The jitter.de people simply don't know what they're yammering about.
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ASRC's apply a lowpass FIR filter to any data that goes through them, even at a 1:1 ratio. So the output data won't be exactly the same as the input.

The presence of this FIR filtering may be objectionable to some. The folks at Pacific Microsonics decided that selecting between multiple FIR interpolation filters would give better sound on transients in music, this became part of HDCD.
 
#20
That is an implementation detail which is not true for all possible ASRC algorithms, but taking it as a given, at least you'd agree that as long as the IN:OUT ratio is stable, the coefficients will not be changed. You could play a fugue on the input clock and it wouldn't make any difference to the output.
 
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