You miss the point that Mr Ease is making: in audio A-D and D-A conversion, the exact timing of the conversion between analog and digital domains determines the sampling jitter and is a dominant factor in the accuracy of the process. Whether the serial digital data is transmitted and received without bit errors is a separate and less demanding issue in this context, but is what the web seminar was mainly concerned with, albeit at much higher frequencies that are found in audio devices.

Audio converter designs usually employ a divided down version of the over-sampling clock as the data stream clock, but this is not an absolute requirement. Instrumentation converters commonly use independent clocks for sampling and for clocking the serial data stream.

Thank you Boswell!

djmukilteo, post: 356907 wrote: Still just digital data being clocked into a digital converter chip nothing that special!

BTW, if you think that high speed digital data transmission or low jitter clock recovery is just a matter of connecting a few IC's then you are hugely underestimating the engineering required.

Circuits that involve digital audio/video systems involving clocking is still a digital transmission issue nothing more or less.
jitter is just one of many slewing and timing issues....so I think you missed the point and audio is not even close to high speed digital data transmission trust me....in fact the problem is the speed at which it takes for the conversions to take place and the cable design used....and I wasn't looking for an argument over jitter just trying to bring the topic into perspective...

djmukilteo, post: 356928 wrote: Circuits that involve digital audio/video systems involving clocking is still a digital transmission issue nothing more or less.
jitter is just one of many slewing and timing issues....

This is just arguing questionable semantics and I would argue that clock recovery has no part in the transmit/receive process that both conveys and corrupts the clock! If all you are interested in is the data integrity over the transmit/receive link, then I would agree. However, at the receiving end of either an S/PDIF or ADAT transmission, we can use a proprietry clock recovery IC which will output a submultiple of the clock required for the local A-D converters. This clock will need to be cleaned up as much as possible and converted to the required frequency for the A-D converters. This is where the PLL comes in and is what I refer to, together with the IC, as the clock recovery circuit. The PLL performance has nothing at all to do with the integrity of the data transmission.

djmukilteo, post: 356928 wrote: so I think you missed the point and audio is not even close to high speed digital data transmission trust me....in fact the problem is the speed at which it takes for the conversions to take place and the cable design used....and I wasn't looking for an argument over jitter just trying to bring the topic into perspective...

Your last comment concerns me as it directly infers that my posts are lacking in perspective. I certainly cannot see why you think this. Let me explain the perspective I perceive I have used throughout this thread in trying to answer the original question, "What is clock jitter in detail?".

The simple and complete answer would be that clock jitter is a total evaluation of the timing errors in a clock due to both random and systematic noise (pretty much what the seminar discussed!). My perspective is that this would be pretty uninformative for this forum. I have therefore attempted to explain in more detail what, exactly, any clock jitter will mean to the sampling of our precious analog audio. This is what I consider to be the most important aspect to everyone here. If that lacks perspective I do not know why.

Now you originally commented in post #80 with this,

djmukilteo, post: 356865 wrote: "Solely focused on data transmission" is the topic here
I don't see any reason why the Tektronix white paper isn't completely relevant as that's what your talking about.
After all the only thing the clock is syncing is data transmissions...nothing else!

Data transmission may be your topic here but it was never mine. What I have been talking about is how clock jitter occurs and the effect it has on the A-D process. Data transmission noise is just one element of the total clock jitter. I think I have already explained above that clock recovery is not part of the data transmission per se, nor is the A-D process which is the main focus of my posts.

djmukilteo, post: 356928 wrote: so I think you missed the point and audio is not even close to high speed digital data transmission trust me....

Thank you for pointing this out. This is EXACTLY why I said I thought that the Tektronix seminar was not particularly relevant to the main focus of my posts - although it was you who disagreed. I DID NOT say, though, that it had NO RELEVANCE to the topic. What point did I miss?

I'm sorry but it is my opinion that it is you, not I, that is lacking perspective on what I have been trying to achieve here. Trust you? No offence but I would rather trust 30+ years of my own experience thank you.

I thought we were talking about clock jitter not "clock recovery" circuits whatever those are....I guess maybe your describing PLL circuitry I don't know maybe something else...

djmukilteo, post: 356974 wrote: I thought we were talking about clock jitter not "clock recovery" circuits whatever those are....I guess maybe your describing PLL circuitry I don't know maybe something else...

Whaaat! You don't know what a clock recovery circuit is and yet you have twice implored me to trust you!

Sorry, I'm just jesting... :<) No offence intended.

Seriously, I have been doing all of this to try and help understanding of what clock jitter really means to our audio and clock recovery is a very necessary and probably the most significant part of the overall clock jitter picture when a soundcard is externally referenced. It seems you have misunderstood the impact the data transmissions we use (i.e. S/PDIF, AES/EBU or ADAT links) will have. I did a quick check on Wikipedia. To save my time I suggest you look up both "S/PDIF" (look at the biphase modulation) and "Clock Recovery" and you should get the basics of why we need clock recovery.....

So yes, I can confirm we are definitely talking about clock jitter and in particular its effects on the A-D process. If we are slaving a soundcard to an external clock then the "data transmission" aspect is only a small (almost negligible) part of the overall clock jitter picture. The quality of the original source clock and the performance of the clock recovery circuit (including the PLL) will have a far more significant effect.

It occured to me that we already have an example of what can happen when there are problems with the logic of clock recovery circuits. In post #33 (on page 4) the last photo shows a blip in the clock jitter. I am quite convinced that this most likely to be due to timing or logic errors in the clock recovery circuits of the MOTU. Quite a clear example of the effects that clock recovery can have on clock jitter and certainly not something caused by the data link between the two soundcards.

It is interesting to note that the MOTU does not appear to use an "off the shelf" clock recovery chip but use their own logic embedded in a Xilinx FPGA. I would hope they have fixed this in subsequent versions of their products. Sadly I doubt they will be offering a firmware update for my older gear to fix the problem.

Sorry Mr Ease if I sounded like I was being arrogant or ignorant on your thread.
Probably a bit of both.....this is a great topic...and I was not trying to derail your explanations in the least. Sometimes I get passionate about stuff I'm interested in and I know if we were sitting around a table discussing this face to face it would come across totally different...

The thing I've been trying to understand goes back to my post regarding atomic clocks and clock sources that make the audio "sound better" and my skepticism on how or where clock generation, clock sync, word clock and timing systems being designed today affect the analog sound in the bandwidths were talking about.

I feel from a basic perspective we are dealing with two locations at which analog (linear waveforms) are digitized....going into an A/D and coming out a D/A.
I'm confused about how the performance of these conversion which is based solely on internal clock generation being derived from a crystal source. If jitter is a form of timing error offset and we employ clock recovery or clock stabilization circuitry to prevent or minimize this....then isn't that already taken care of as part of the A/D or D/A designs themselves?
If we are talking about synchronizing multiple electronic circuit devices using a transmission medium (i.e. cable, fiber optic, coax) then I would think we are talking about something entirely different (maybe I'm wrong) and we are no longer talking about linear waveform conversions....we are talking about data transmission....Am I all wrong here and if so please feel free to beat up side the head...LOL
Cheers

LONG POST WARNING!

Ok, let me try and address your questions as it seems my earlier explanations have not fully hit home. Say we have a multi-channel A-D and D-A soundcard , something like my MOTU828. Internally this will utilise several A-D and D-A IC's and these will all need the same clock reference. This is not part of the chip itself but is, if you like, an internal master oscillator. As this will usually be a crystal controlled oscillator it should be perfectly OK in stand alone mode (i.e. not externally sync'd). Of course it is possible to get even this wrong but let's assume the designer knows what he is doing. As a side note I have come across many instances over the years where this is not the case, mainly with predominantly "logic" designers.

This clock is distributed to all the chips that need the reference. This is an area that must also be done carefully as it is possible for noise to be picked up on PCB tracks from adjacent tracks with either fast logic transients or high currents. Again we must assume that the designer has done his job properly. From the various companies that offer post market "improvements" we could perhaps deduce that the original designs are not always the best. Bear in mind that these companies also offer mod's to the analog side. What is quite easy to find out is the specification of the particular A-D and D-A IC's being used. The specifications of these chips determine the best we will ever achieve when we use them as there is no scope to improve on these figures. There is however plenty of scope to throw away this performance. Such is life, we choose the soundcard that we feel offers us the best bang for our buck.

So in answer to your first question, if you regard the A-D and D-A designs as the entire soundcard then it is taken care of with the above caveats. If you regard the A-D and D-A design as the chip itself then no, it is entirely your responsibility to make sure the clocks are good and the analog design is top class.

When we talk about synchronising multiple soundcards then we have to use one of the soundcards as master or use an external master clock. This is usually done via the usual digital interfaces (or wordclock) which all distribute the master clock reference at the much lower wordclock frequency rather than the master oscillator frequency itself. This will be done with either coaxial cable or lightpipe. Now what you need to grasp is that the timing explanation in the Tektronix seminar examines the data transmission at both the transmit and receive end. At the transmit end the slew limitations, timing jitter and reference levels are all generated by the transmitting equipment - irrespective of the bandwidth of the transmission. This is NOT something caused by the connection between the transmitter and receiver. When we examine the data at the receiving end then we are looking at the effect of the cable linking the two. For a long cable and very high speeds then we will see slower transients, ringing lower logic levels etc. depending on the cable characteristics and impedance matching at each end. Suffice it to say that at the speeds we use in digital audio and the typical cable lengths (a few metres) you would be hard pushed to spot the difference at each end of the link! What you will NOT see is an increase in clock jitter at the remote end as the cable characteristics do not change with time...

Now if we use a wordclock distribution then the wordclock can be used directly for the slaved soundcards PLL but more normally we will be using S/PDIF, ADAT etc. In this case we will need to use a clock recovery circuit to decode the wordclock to be used as a reference for the PLL. The PLL in all cases has the job of reproducing a facsimile of the master unit clock which is locked to the master clock frequency.

The fact that the master clock is a digital signal and has been distributed via a digital interface actually has no effect on all of the A-D IC's in either the master or slave soundcard. All they see is the clock they are fed with and care not a jot where they came from. What IS important is that the master and slave clocks do not suddenly have excessive jitter. Where might any such jitter arise then? Well there is no reason for the transmitted wordclock master to be degraded from the master oscillator and we will not see a significant increase in jitter from the cable connection. The key elements then are the clock recovery circuit in the slaved soundcard (when using S/PDIF etc.) and the PLL. It is these circuits that will have the most profound effect on clock jitter within the slave soundcard. This is the reason I discounted any sonic improvement being be realised by using an atomic frequency standard - they have no better jitter performance than just a simple, well designed, crystal oscillator and in some cases (i.e. Caesium) they are worse!

So to finally answer your second question, yes the clock is transferred from soundcard A to soundcard B via a digital link. Is that digital link critical to our A-D and D-A performance? No. The only thing we are interested in is the jitter of the master and slave clocks in order to not degrade the performance of our A-D's and D-A's. The basic perspective here is that the A-D's being used in the master soundcard are being faithfully replicated by the slave soundcard with no degredation. Of course if you slave your PC's onboard sound system to a Linx then the onboard sound system is not suddenly going to sound like the Linx! :<).

I hope that this lot has not bored the pants off you all and casts another incremental bit of light on this whole process. Once more I am writing this trying my best not to be overtechnical so that you can all understand. Do let me know if I have failed! I will read back through this at leisure and will try and edit any tatty english...

Mr. Ease: I enjoy the technical discussions here. Besides cymbals and percussion, there are Fourier components of violin strings up near 20 kHz and theoretically beyond. Ever read Prof. Philip Morse's "Vibration and Sound" and work out some of the examples? This text was written before computers but the proofs in the text are time invariant. The complete solution of the vibrating string is contained in the Morse text as well as other textbooks.

I'm glad you enjoy the discussions. I am well aware that all "natural" instruments have varying degrees of harmonics and many extend to supersonic frequencies. What is important to appreciate though is that the fundamental notes never exceed a few kHz so all the energy in our recordings above 10kHz are actually harmonics. While these harmonics contribute to the overall sound of each instrument the biggest factor in being able to identify a particular instrument comes from the transients at the start of a note. This is easy to verify by removing the starting transients of notes on various instruments and slowly bringing up the level. When you do this it becomes much, much harder to identify individual instruments.

As far as this topic goes, what is important is that the level of any harmonic is always considerably lower than the fundamental, even for the extreme cases of square and triangular waves. A far as clock jitter goes this, fortunately, plays in our favour as it limits the amplitude of the very highest frequencies that could cause audible artifacts. :<)

Boswell, post: 346239, member: 29034 wrote: The Wiki article is detailed but general. In the field of recording, all one usually needs to know is that clock jitter is the variation in time of a sampling instant about the ideal time, and apples to signal conversion between analog and digital domains in both directions, i.e. both A-D and D-A conversion. Since the analog waveform is constantly changing, a variation in time translates to a variation in amplitude, and hence an error in the converted signal. This will show as non-harmonic distortion.

MrEase, post: 346465, member: 27842 wrote: First of all I would echo what Boswell said regarding Soapfloats dropouts. Maybe one question arises though, are you (Soapfloats) using interfaces sync'd to master or other clocks? If so the possibility of dropouts due to jitter will increase significantly but still should not be enough to cause dropout problems.

What a stellar thread. Best on the web for us guys.
Thanks again for taking the time on this one.

Don't know if this helps anyone, but in a more practical vein (i.e. what does that sound like?), the folks at Cranesong put together a couple of pages about jitter with downloadable audio samples:

http://www.cranesong.com/jitter_1.html

apstrong, post: 433238, member: 36444 wrote: Don't know if this helps anyone, but in a more practical vein (i.e. what does that sound like?), the folks at Cranesong put together a couple of pages about jitter with downloadable audio samples:

http://www.cranesong.com/jitter_1.html

That's a very useful link, AP... thanks for posting it.

After listening to the 4 recordings of the exact same performance, make sure to check out Page 2 of the link as well, which isolates the jitter of each sample file through the use of phase cancelation of the primary sound sources.

The result is that you can actually hear what the various forms of jitter sound like. I think you'll be surprised at what you hear, and in how much different they all are from each other.

Also... check out the "setup" page ( link below), which diagrams the routing used as the "control" foundation through which all the test samples were routed:

http://www.cranesong.com/JITTER_TEST_SETUP.pdf

The article is very upfront about the fact that its aim is to educate listeners to what jitter sounds like, as opposed to being a tightly controlled scientific test.
There are situations where the gain of certain phase cancelled files was increased to make the jitter more audible.
So, it's not as much a tightly controlled A/B test, as it is a way for people to become educated to what various forms of clock jitter sound like.

I was amazed at what I heard on Page 2.

FWIW

-d.

Wow, it's nearly three years since this thread was active and I see it's now over 16000 views!

AP, that is indeed an interesting test. However I don't think there will generally be a particular "sound" associated with jitter as it will depend very much on the jitter characteristics. Hence, I doubt if you'd ever be able to hear a particular "sound" and say conclusively "that is due to clock jitter!"

The reason I say that, is that jitter can arise from many sources and the spectrum of the noise causing the jitter will change the jitter induced artifacts. As I think I mentioned back in the thread somewhere, white noise sourced jitter should be the least intrusive (which should sound like an elevated noise level) but any jitter arising due to a discrete modulation (which does happen) would affect the sound in different ways depending on the modulation frequency and would also be more noticeable.

Having said all that, I don't have time (or a decent sound set up on this PC) to try the tests for myself. If I get the chance though I'll try to find out more about the jitter they have induced on the tests. A quick read through didn't make it clear to me whether discrete modulation frequencies were used or not.

MrEase, post: 433242, member: 27842 wrote: Hence, I doubt if you'd ever be able to hear a particular "sound" and say conclusively "that is due to clock jitter!"

I agree. More to the point, and being totally honest, I wasn't able to hear it at all on the samples they provided - until I went to the second page and heard the various types isolated through phase cancellation.

I think that this is kind of the point to the exercise - that it is hard to recognize, and that it never really "sounds" the same, either.

It's not really an identifiable sound, something that you could point to and say unequivocally, "Oh, yeah... no doubt, that's " _________" , as would be the case with something like SMPTE Time Code chatter, which is immediately recognizable.

So... I'm curious - because there was only the one instrument used for the various samples - I think that it begs the question, "so what happens when you have that jitter happening on 24 different tracks in a mix at the same time?"

It leaves me wondering if - much like noisy preamps can do - it would become more obvious as the various jitter issues would accumulate/stack up, track by track? That would seem to be the case, but I don't know... which is why I'm asking. :)

I dunno listening the the tests I couldn't hear much if any difference between them, The phazed examples were nuts that's a big difference.
I think in the practical world jitter is just going to become a part of your sound like everything else. Preamps, eq, mics etc.

Is there any product out there with absolutely no jitter? Any eq or preamp with zero noise? At what point is it measurable but not noticeable?

I loaded the files into my Daw B is the control the rest are out of phaze with B to show the difference. There's is definitely stuff there and a difference but the level is unhearable unless you crack your speakers or headphones to a crazy level. i can;t see how in the real world that's going to be a factor. Unless you are recording ants walking across the floor I can't see how you'd reach/need that level of gain boost or compression to actually hear it.

Chris Perra, post: 433246, member: 48232 wrote: I dunno listening the the tests I couldn't hear much if any difference between them, The phazed examples were nuts that's a big difference.
I think in the practical world jitter is just going to become a part of your sound like everything else. Preamps, eq, mics etc.

Is there any product out there with absolutely no jitter? Any eq or preamp with zero noise? At what point is it measurable but not noticeable?

I think part of your comment in regards to not hearing anything to be concerned with, may also be something some of us know effects the greater of what isn't realized on the surface. Accumulating ?
an interesting pov from both you and Donny .

Could this fall into personal hearing. Examples: loss, ability, awareness on an individual level ... why some of us freak out when someone scratches a black-board, or why some of us can continue working when a baby is screaming.

Tolerance to freq or less bothered by, does it really matter ? ...

I suppose some people would have the ability to hear the difference. I would suspect most wouldn't. For me I can hear my fingers rubbing together sitting in my studio while listening to those examples out of phase. I can hear my fingers but not the jitter without cranking the snot out of the gain. For me,.. that's enough not to worry about jitter.

It's definitely there though.... I just can't hear it with those examples.

Here's B vs A 40 tracks of each cancelling each other out. This is audible. I wonder if Jitter stacks like this or not though It's pretty noticeable.

[MEDIA=audio]http://recording.or…

Attached files A and B difference 40 tracks.mp3 (1.3 MB) 

What I notice most problematic in music,

• why I choose to replace specific tracks in order to improve the imaging of busy mix,

• what has to do with consistent acoustic reflections throughout all the tracks summed
• what we tend to sacrifice for volume on a track per track.Example moving volumes from track to track and forgetting about the silent sections we think is dead and useless information....
• what happens when we overdub and break tracks into pieces, I'm seeing the image of invisible sky scrapers of dead space left in a mix. So what we have are all sorts of levels of air, we get sloppy with.
• digital editing....

This is imho, why music today sounds boring, dead and worst case, all swirly.

Inconsistent track to track acoustic space (room sound, reverberation, bleed from track to track, bouncing, overdubs) all effects the sum image. This is the "accumulating" that depicts a "good, better best, stellar" sounding recording. Which I assume why people invest in $6000 master clocks to try and help sync their problematic rats nest of external gear, why we invest in better, more stable converters, which I would go so far as including, why a 2 channel DA AD apposed to 8 or 16 or 32 one rack converters are more stable.

I'm thinking PSU and the ability for multiple channels to remain stable starts looking why most of the top end converters don't go past 8 channels on a single PSU.

Most of us only listen to the face of an instrument and pay less attention to the dead space around it. I think its safe to say, the dead space has a lot of detail that gets us in trouble. Kind of like have higher ceilings in one track , opened windows in another and mixing them all up at various levels, all out of sync.

I'm thinking this is related in this thread somewhere?

This 40 track example shows what happens if you were to use the same material and run it round trip 40 times. I'm not sure who does that wanting to keep the track pristine at the end. In a practical situation you would be round tripping to outboard gear with the intent to change the sound. So unless you hear horrible things from the hardware you are using, I'm not sure it would be a concern.

I also wonder if say you recorded a whole band all at once whether or not the jitter would stack the same way or be the same.

I suppose if you had 24 tracks going. 3 sets of 8 channels per Interface then you would have only 3 potential jitter differences.
If you recorded say drums over 2 units A and B and used A for the rest of the overdubs you'd only have 1 jitter difference.

I'm not sure if jitter in that situation would be an audible concern.

It is important to recognise that clock jitter causes slight errors during the sampling process on each individual track. Once digitised these errors are fixed within that track. Provided each individual track does not have excessive errors that are audible then there should be no cumulative effect whenever mixing in the box as each track is independent whatever processing is used. Compare this to a track where you add some overdrive to a guitar. You would have no expectation of this overdrive affecting other tracks within the mix or causing a cumulative effect.

Clock jitter only becomes a concern whenever you convert A-D or D-A as this is the only stage where jitter can have an effect. For instance, if you mix outside the box then you are passing the mix through D-A converters and then back into (possibly another DAW's) A-D. This process will add two stages of clock jitter error to the equation. Balance this with the fact that several members here are convinced that mixing OTB and using an independent DAW to record the mix yields improved results and we must conclude that either there is something the OTB mixer adds or could it possibly be that the clock jitter helps??? I'm not drawing any conclusions here but really I'm pointing out that this really shouldn't be an issue.

Just a point about the tests given in the Cranesong links. Whenever you have to significantly boost difference signals, you are also losing resolution. In another forum some years ago I contributed to a thread about dither where I intentionally limited the resolution of a sine wave to just a few bits - which sounded awful when boosted. Using dither on the same process produced a much more acceptable sine wave at the same resolution. This is really to point out that the boosted difference signals are not necessarily representative of the "true" distortion from clock jitter and will easily add extra distortion due to limited resolution. Does anyone remember the older digital synths with 8 or 10 bit resolution? They generally sounded OK(ish) but I frequently found the "buzzy, distorted" tails on some decaying sounds really annoying! Again, that was just down to limited resolution.

So, unless something is totally noticable noise, buzz etc, jitter isn't a concern?

MrEase, post: 433334, member: 27842 wrote: Balance this with the fact that several members here are convinced that mixing OTB and using an independent DAW to record the mix yields improved results and we must conclude that either there is something the OTB mixer adds or could it possibly be that the clock jitter helps??? I'm not drawing any conclusions here but really I'm pointing out that this really shouldn't be an issue.

For the quoted reasons you point out I will share some brief insight into my madness:
I choose to use two uncoupled DAW's for a variety of reasons.
These would include:

• tracking @ higher samples rates than the destination, monitoring SR conversion in real time,

• increasing computer performance (dedicated DAW's to do specific tasks),
• avoid bouncing,
• monitoring strategic order of specialized outboard gear,
• monitoring sum and master
• monitoring the harvesting and export of audio, especially when you are in the creative process of a mix/sound replacing/ sound designing/master and you are flipping between two DAW's .

The list of why I do what I do is long and little to do with the simple change that may happen (add value) using a D A D pass, but, it is one of the reasons as well!

Personally I find ITB sounds bigger and fuller to what any analog console does for me. Once ITB, stay ITB. The less converter, the better.
But, One box kind of sucks in comparison to 2. My madness is all about the workflow and monitoring the workflow.
DAW1 is for tracking and mixing, DAW2 is for capturing the sum. DAW 1 is designed only for tracking and mixing, DAW 2 is designed for summing, mastering and export.
To my knowledge, this is not possible on one computer, at least not to the level I require.

I do hear an improved sum when I track at example: 96/24 and capturing the mixdown sum to 44.1/24 apposed to bouncing on one box like the mass do.

Hi Chris (audiokid),

I wasn't making a comment as to why you might do something a particular way and certainly not saying I disapprove of your methods or whether you are mad or not! :)

Hi Chris (Perra),

Chris Perra, post: 433339, member: 48232 wrote: So, unless something is totally noticable noise, buzz etc, jitter isn't a concern?

That's sort of putting words in my mouth but in essence, true. What I was really trying to say is that clock jitter only comes into play whenever you move from analogue to digital or vice versa. If you're happy with the recorded sound of individual tracks then the mixing process will have no impact whether tracks had high jitter when tracked or not. There should be no accumulative errors due to jitter and the mixing process. Note that when mixing, you are constantly playing back and the playback clock jitter comes into play. If you really had a clock jitter problem then it should be noticeable then, even if the tracking was "perfect" (which it can never be, sadly).

I guess it's worth noting that even if you're just listening to a single track, you have two lots of jitter involved, once when recorded and again whenever you play it back.

To all, I only came back here as I was interested in the samples provided in the link given earlier and had some comments I thought people should be aware of. I hope it helps!

MrEase, post: 433430, member: 27842 wrote: Hi Chris (audiokid),

I wasn't making a comment as to why you might do something a particular way and certainly not saying I disapprove of your methods or whether you are mad or not! :)

no worries, but I am Mad for getting into the business. Somehow I'm able to function remotely well on the outside world. :)

I have more questions for you, just need to think about the wording. Thanks as always for your contribution here.

For me.. listening to this examples nulled and hearing super exaggerated jitter I'd treat it like part of your sound like any other part of the chain. You either like or don't notice the sound or notice and dislike what you hear. If it sounds good it's good if it sound bad it's not.

Up to now I've treated this whole thread almost entirely from the (electronic) engineering standpoint. I agree with Mr. Perra though.

The bottom line is that I've re-recorded so many things in the past due to a botched performances (frequent in my case!), trying different mic's & positions to get better sound and lots of other reasons. I have never ever re-recorded because I thought there was too much jitter! Other than swapping master's on my two sound-cards, there's nothing else I could do anyway. The key here is what I said earlier - I doubt anyone can clearly state what clock jitter actually sounds like as it can arise from such diverse sources.

I doubt anyone hereabouts can ever say they re-recorded any track solely due to jitter problems. Drop-outs sure but not clock jitter per se.

Can anyone here say for sure that they have ever encountered a problem with clock jitter?

MrEase, post: 433447, member: 27842 wrote: Can anyone here say for sure that they have ever encountered a problem with clock jitter?

Unequivocally no. Until I heard the examples posted, I had no idea what clock jitter even sounded like. I knew from reading various articles over the years that clocking issues could be a concern, but I've never re-recorded or re-mixed anything because I heard clock jitter.

That's not to say that I might not have heard something amiss as a result of clock jitter, I suppose that it's possible that this has happened - but as far as positively identifying it or knowing for a fact that those problems were caused by jitter, no... I've never once heard something "off" on a track or mix, and said "Yep... That's clock jitter!" And, taking into account that it never seems to sound the same, presents an even greater difficulty in identifying it. We heard, what, four examples of intentionally represented jitter? And, no two of them sounded alike; so how many other possible audible examples of jitter are there? 50? 100? 1000? Maybe even limitless? If it never sounds the same, if it never has the same sonic fingerprint, if it's always "random" and presents itself differently every time, then how can it be identified as such in a mix? I'm not telling ... My question isn't rhetorical, guys ... I'm sincerely asking here...

I've certainly re-tracked and re-mixed due to certain issues I've heard - things like maybe too much "room" on a vocal track, or phase issues, or other noise(s) - times where I heard noise due to low output mics and a cheap pre gained-up in response, or hum on a guitar track, or sibilance, etc. Those issues are identifiable. You know phase issues when you hear them, you know the sound of a noisy pre, or the sound of too much "room" on a vocal or on an acoustic guitar track...

But never was it because I was able to actually hear and definitely attribute the problem to jitter.

IMHO of course. :)

-d.

Sorry for the slow reply Donny. The simple answer to your non rhetorical question is that you're highly unlikely to ever have jitter induced noise that you could ever hear. If you look way back in the thread I gave calculations of the worst possible case scenarios for jitter induced artifacts which (I hope) show how unlikely this is to ever be a problem.

If you consider the Cranesong tests, they only serve to confirm that. They have deliberately introduced very large and discrete signals to the jitter and then had to enormously ramp up the gain on the difference files to even give a demo of a jitter "sound". The real world is quite different and even a relatively high jitter clock would normally be made up of predominantly pink(ish) noise (once again with caveats regarding poor design!). This would not have given Cranesongs much to go on in trying to demo the "sound" of jitter as it would only impart a pink(ish) noise to the recorded signal at very low levels. Ramping up the difference gain on this would only yield pink noise that should be indistinguishable from "normal" noise.

All told this means you have no way of having a generalised "how can it be identified". In all honesty I sincerely believe the "prophets of doom" in regard of clock jitter have got it wrong. All our A-D, D-A and DAW processes rely heavily on mathematical procedures. It follows that the same procedures can show why clock jitter should not be a problem in our audio systems. This is what I have attempted to show in this thread giving some real numbers to work with. It makes no sense (to me at least) to discount the simple maths I've presented here while inherently accepting that all the other maths our signals are processed with are OK.

Jitter-induced artifacts and jitter-induced distortion are two separate things. As MrEase said, the artifacts are almost never encountered on a properly clocked system, as they would manifest themselves as double-sample, spikes or similar very audible (and visible) things.

On the other hand, distortion due to sampling uncertainty is present every time you convert from the analogue domain to the digital domain or back again, as the sampling instant is never precise, at least at the picosecond level. A shift in sampling instant results in a change in converted amplitude of an audio waveform, and this shows up as a distortion in the conversion process. Any digital processing assumes the waveform is sampled at exactly the clock instant, and when it is not, the sampled value is wrong by a small amount. This is not visible at the DAW level.

What you have to evaluate is whether the sampling uncertainty caused by the inevitable clock jitter in a system results in an acceptable level of audio distortion. This type of jitter may not cause any audible artifacts, but the waveform does deteriorate, and, as the jitter increases, you will start to hear it first on certain types of programme material, solo piano being a common example.

It's not easy to generate demonstration files of this type of distortion. Maybe the same piano recorded using a low-end converter and a top-quality low-jitter converter would illustrate something, but it would depend heavily on the listener's reproduction equipment whether any difference between the two could be detected, and that would have to be after other more obvious differences in the quality of the conversion itself had been allowed for. Simply degrading the external clock going into a top-end converter would not produce the required effect, as modern high-end boxes use an incoming clock as a long-term frequency correction of their own internal conversion clock rather than clocking the conversions directly from it.