Yet another look at Jitter: Clock Extraction

amirm

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I hope you are not getting tired of jitter talk because I have more info to share :).

One of the common arguments made against jitter mattering is that: "the data is buffered and clock regenerated in the DAC so jitter won't be there." This makes all the sense in the world. Once we capture the data and then push it out at our will, there shouldn't be a problem. Well, there is a problem. A serious one. Buffering and clock regeneration do not deal with jitter by themselves. I have explained this in words many times but this time I am bringing in some specific data to hopefully put this myth to bed (yeh, wishful thinking :) ).

Introduction
The way a clock is "regenerated" is to have a local oscillator (clock) that we can change its frequency to eventually match and track the incoming digital stream. As you may know, S/PDIF is a serial digital connection with clock and data intermixed. By using this circuit which is called a Phase Locked Loop (or PLL for short), we are able to extract a clock that is cleaner than the incoming one. This clean up allows us to capture the digital samples reliably.

Our job is not done however. We not only need to extract the data samples but also accurately match the incoming data rate (clock regeneration) without any timing variations in it getting to our DAC. Not doing so causes jitter to appear in the output of the DAC.

S/PDIF Receiver Clock Regeneration
Let's look at a Crystal Semiconductor CS 8416. From its data sheet: http://www.cirrus.com/en/pubs/proDatasheet/CS8416_F3.pdf, we see this measurement of its rejection of incoming jitter as it regenerates the clock using its PLL:



I have annotated their graph as to make it easier to understand.

The measurement shows how much of the incoming filter is reduced in amplitude (vertical scale) at each frequency of jitter (horizontal axis).

Note as I have indicated on the graph there is absolutely no reduction of incoming jitter below 8 Khz!!! Even worse, there is actual *amplification* of jitter prior to reaching 8 Khz to the tune of 2 db. The "peaking" is due to the way the PLL filter is designed and is outside of the scope of this conversation to explain why it occurs.

Even at 20 Khz, we only have a modest 6 db reduction of incoming jitter.

The serious reduction in jitter therefore is from 20 Khz and up. Why? Because it is those frequencies which cause us to not be able to extract the incoming *data* -- the PCM audio samples. That is the "mission critical" application in IT terms. If the receiver can't decode the incoming bitstream, your system breaks and we can't let that happen. So high-frequency jitter is eliminated and with it, a ton of ultrasonic noise and interference.

Measurements
You might say, this is all theory, who knows if it is true in practice. And that would be a fair question. Fortunately I happen to have data on this :).

Here are the measurements of jitter reduction -- i.e. the same as above graph -- for all the devices that I tested for my Widescreen Review Magazine article:



Look at the mass market AVR performance when it comes to jitter reduction. Notice how there is no attenuation or slight amplification of jitter as the above data sheet showed. I show the response up to 10 Khz so the sharp attenuation in higher frequency is not there in my measurements. What is there of course, what we care about: Jitter reduction in *audible* frequency range.

Summary
We see that the common argument of audio frequency jitter being eliminated because we have buffering and clock regeneration simply does not hold water. The common/cheap implementation only gets rid of high frequency jitter so that it can reliably extract digital audio samples. It does little to filter out incoming jitter in audio band which is what we care about.

Of course the problem can be solved using skilled designers and budgets that are measured in tens of dollars as opposed to single digit.
 

LL21

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Dec 26, 2010
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I am NO techie as I must have said a thousand times. Basically a stream comes into the DAC via, say, SPDIF...so the DAC with the 'reclocking' capability 'falls down on the job' because:

- it cannot extract the timing signal from the data signal off the SPDIF properly to give it to the new clock before sending out the data/clock information via analog to the preamp?

Sorry...I'd like to understand this in really simple-man's terms...and your simple explanation is still not simple enough for non-techie me.
 

Phelonious Ponk

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Jun 30, 2010
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I hope you are not getting tired of jitter talk because I have more info to share :).

One of the common arguments made against jitter mattering is that: "the data is buffered and clock regenerated in the DAC so jitter won't be there." This makes all the sense in the world. Once we capture the data and then push it out at our will, there shouldn't be a problem. Well, there is a problem. A serious one. Buffering and clock regeneration do not deal with jitter by themselves. I have explained this in words many times but this time I am bringing in some specific data to hopefully put this myth to bed (yeh, wishful thinking :) ).

Introduction
The way a clock is "regenerated" is to have a local oscillator (clock) that we can change its frequency to eventually match and track the incoming digital stream. As you may know, S/PDIF is a serial digital connection with clock and data intermixed. By using this circuit which is called a Phase Locked Loop (or PLL for short), we are able to extract a clock that is cleaner than the incoming one. This clean up allows us to capture the digital samples reliably.

Our job is not done however. We not only need to extract the data samples but also accurately match the incoming data rate (clock regeneration) without any timing variations in it getting to our DAC. Not doing so causes jitter to appear in the output of the DAC.

S/PDIF Receiver Clock Regeneration
Let's look at a Crystal Semiconductor CS 8416. From its data sheet: http://www.cirrus.com/en/pubs/proDatasheet/CS8416_F3.pdf, we see this measurement of its rejection of incoming jitter as it regenerates the clock using its PLL:



I have annotated their graph as to make it easier to understand.

The measurement shows how much of the incoming filter is reduced in amplitude (vertical scale) at each frequency of jitter (horizontal axis).

Note as I have indicated on the graph there is absolutely no reduction of incoming jitter below 8 Khz!!! Even worse, there is actual *amplification* of jitter prior to reaching 8 Khz to the tune of 2 db. The "peaking" is due to the way the PLL filter is designed and is outside of the scope of this conversation to explain why it occurs.

Even at 20 Khz, we only have a modest 6 db reduction of incoming jitter.

The serious reduction in jitter therefore is from 20 Khz and up. Why? Because it is those frequencies which cause us to not be able to extract the incoming *data* -- the PCM audio samples. That is the "mission critical" application in IT terms. If the receiver can't decode the incoming bitstream, your system breaks and we can't let that happen. So high-frequency jitter is eliminated and with it, a ton of ultrasonic noise and interference.

Measurements
You might say, this is all theory, who knows if it is true in practice. And that would be a fair question. Fortunately I happen to have data on this :).

Here are the measurements of jitter reduction -- i.e. the same as above graph -- for all the devices that I tested for my Widescreen Review Magazine article:



Look at the mass market AVR performance when it comes to jitter reduction. Notice how there is no attenuation or slight amplification of jitter as the above data sheet showed. I show the response up to 10 Khz so the sharp attenuation in higher frequency is not there in my measurements. What is there of course, what we care about: Jitter reduction in *audible* frequency range.

Summary
We see that the common argument of audio frequency jitter being eliminated because we have buffering and clock regeneration simply does not hold water. The common/cheap implementation only gets rid of high frequency jitter so that it can reliably extract digital audio samples. It does little to filter out incoming jitter in audio band which is what we care about.

Of course the problem can be solved using skilled designers and budgets that are measured in tens of dollars as opposed to single digit.

"Tens of dollars." :) High-end.

Tim
 

amirm

Banned
Apr 2, 2010
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Seattle, WA
I am NO techie as I must have said a thousand times. Basically a stream comes into the DAC via, say, SPDIF...so the DAC with the 'reclocking' capability 'falls down on the job' because:

- it cannot extract the timing signal from the data signal off the SPDIF properly to give it to the new clock before sending out the data/clock information via analog to the preamp?

Sorry...I'd like to understand this in really simple-man's terms...and your simple explanation is still not simple enough for non-techie me.
Almost :).

This is the simplified block diagram of the digital to analog converter:



The output is what goes to your pre-amp.

The input is the S/PDIF digital interface. And yes, the DAC must extract the precise timing of the input, while ignoring the noise and other artifacts that may be on it. This is a harder problem than it seems at first.

Think of it as a turntable. The bigger and heavier the platter, the more it maintains its speed and ignores momentary causes of speed change. The drawback is that it takes it longer to get up to speed. In a turntable that is not a problem. In a DAC that can be a problem in that when you change inputs or input sample rate, it will then take the DAC a while to catch up. Delays longer than a few milliseconds tend to be annoying so a trade off is made (in cheap implementations) to "lock" to the input rate fast at the expense of higher amount of input variations travelling to the output of the DAC.

Hope I didn't confuse you more :).
 

jkeny

Industry Expert, Member Sponsor
Feb 9, 2012
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Ireland
Thanks Amir
Another good read
Can I ask why you didn't measure below 20Hz - I don't think jitter frequency is a direct correlate to audio frequency otherwise jitter above 20KHz would be of no consequence. Jitter frequency below 10Hz is known as wander & above as jitter so at least 10Hz should be the lower bound but probably less.

From your graph - what level of jitter attenuation renders the jitter inaudible - your graph really doesn't answer this or indeed give a handle on what level of jitter is audible.

Finally, what tens of dollar solution do you suggest to handle jitter at all frequencies?
 

DonH50

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Jun 22, 2010
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Nice job Amir.

Reliably extracting the data from the bitstream is not that hard relative to everything else. Not that some design might not screw it up... But then there is error correction (which does not help the clock, of course).

As Amir said, the problem is when the recovered clock is applied directly to the DAC, something many AVRs and probably a lot of cheap DACs do. It is probably worth a reminder that jitter is related to the signal frequency; no relation to the clock frequency (to first order).

You can make a very clean clock output, but it costs more money. Not much relative to WBF members, but $1 to $10 added to millions of units is a big number on a product that has very low margins to begin with.

One quasi-interesting thing I found out was that the Pioneer AVR I got a few years ago (SC-27) has among the lowest jitter. Hard to find since it is not a spec most manufacturers report. Some highly-regarded AVRs exhibited very high jitter, e.g. 50 ns or more. If you want a rough idea of how much jitter matters at what frequency and bit depth (resolution), there's a chart in my old Jitter 101 thread: http://www.whatsbestforum.com/showthread.php?1322-Jitter-101
 

BlueFox

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Nov 8, 2013
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Thanks for the post. Interestingly, this is being discussed on another forum where some are saying jitter doesn't matter since once the data is buffered it is gone. Hope you don't mind my giving a link to this thread.
 

Julf

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But this only applies to cheap DACs that use the data clock (retrieved using the PLL) as audio clock. It doesn't apply to more advanced designs with independent audio clock, right?
 

LL21

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Dec 26, 2010
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Almost :).

This is the simplified block diagram of the digital to analog converter:



The output is what goes to your pre-amp.

The input is the S/PDIF digital interface. And yes, the DAC must extract the precise timing of the input, while ignoring the noise and other artifacts that may be on it. This is a harder problem than it seems at first.

Think of it as a turntable. The bigger and heavier the platter, the more it maintains its speed and ignores momentary causes of speed change. The drawback is that it takes it longer to get up to speed. In a turntable that is not a problem. In a DAC that can be a problem in that when you change inputs or input sample rate, it will then take the DAC a while to catch up. Delays longer than a few milliseconds tend to be annoying so a trade off is made (in cheap implementations) to "lock" to the input rate fast at the expense of higher amount of input variations travelling to the output of the DAC.

Hope I didn't confuse you more :).

Thank you...let me try to 'read that back to you' in even more basic terms. The digital information is sent across to the DAC by an analog signal (electrical impulse simulation of the 1s, 0s of the data). This analog on an SPDIF transmission combines the clock information with the data stream. Because it is an analog signal, there can be teeny fluctuations, inconsistencies, and the receiving DAC has to have a way to dealing with this. Lower quality DACs will seek to eliminate certain problems by cutting corners around others...meaning that they inadvertently create their own inaccuracies in the information before it goes out of the DAC. And further, the DAC is creating a true analog signal (ie, the music) to go out to the preamp...so again, there is risk in poor implementations of jitter within the DAC being reintroduced just before the data gets converted back into music.

Any closer?
 

Julf

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Thank you...let me try to 'read that back to you' in even more basic terms. The digital information is sent across to the DAC by an analog signal (electrical impulse simulation of the 1s, 0s of the data). This analog on an SPDIF transmission combines the clock information with the data stream. Because it is an analog signal, there can be teeny fluctuations, inconsistencies, and the receiving DAC has to have a way to dealing with this. Lower quality DACs will seek to eliminate certain problems by cutting corners around others...meaning that they inadvertently create their own inaccuracies in the information before it goes out of the DAC. And further, the DAC is creating a true analog signal (ie, the music) to go out to the preamp...so again, there is risk in poor implementations of jitter within the DAC being reintroduced just before the data gets converted back into music.

There are no "inaccuracies of information" - the actual data gets transferred without errors and without any change. The issue is with the clock - cheaper DACs use a clock that is generated by having a PLL lock to the incoming data clock, and that PLL is not always very good at rejecting jitter. Thus the timing of the DAC conversion can be affected, resulting in jitter-induced distortion/noise on the analog output.
 

LL21

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There are no "inaccuracies of information" - the actual data gets transferred without errors and without any change. The issue is with the clock - cheaper DACs use a clock that is generated by having a PLL lock to the incoming data clock, and that PLL is not always very good at rejecting jitter. Thus the timing of the DAC conversion can be affected, resulting in jitter-induced distortion/noise on the analog output.

Got it. Thank you!
 

DonH50

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But this only applies to cheap DACs that use the data clock (retrieved using the PLL) as audio clock. It doesn't apply to more advanced designs with independent audio clock, right?

Unfortunately, no, not completely, though I would guess that is true in the vast majority of cases... Or rather it most likely eliminates data-dependent (deterministic) jitter but does not mean you have a jitter-free result. It is possible to design a great clock recovery circuit that would pretty effectively reject the jitter even if it is not independent of the bit stream, though getting the corner down below say 10 Hz is a major challenge. It is also possible to use a separate clock and yet still design a poor oscillator or PLL that has its own jitter issues. IMO one of the big benefits of an independent source is you can isolate it from the data (signal), eliminating the data-dependent clock noise that is most likely the largest source of audible degradation. There are other coupling mechanisms so even a separate clock might not eliminate all data-dependent jitter, however.

To go just a hair deeper, low-frequency noise generated by devices like transistors and tubes has a rising characteristic as frequency reduces; the lower the frequency, the higher the device noise. You have to use great care in device selection and clock circuit design to get rid of that LF noise.

Like most things it gets complicated quickly. PLL design is something I have done, though certainly am no expert, and it is one of those things that does not look all that hard in a grad EE class but is very difficult in the real world. For RF/microwave systems like radars and high-performance data links, that LF noise causes much the same problems as for audio. For some silly reason pilots get a bit vexed when an incoming airplane or missile appears to jump about a bit on their radar screen... :)


Bias Disclaimer: I am on the fence with jitter, feeling based upon what I read that for the vast majority of audio applications and using real-world source material it is not audible. However, I have insufficient listening experience to really know for myself, and of course my listening experience is irrelevant to that of anyone else. My personal feeling is that the majority of designs offer adequate jitter performance and many other things dominate the sound. Seems unlikley jitter spurs 60 - 70 dB down are going to be more of a problem than the 0.5 - 10% or more distortion from your speakers, for example.
 

Julf

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Unfortunately, no, not completely, though I would guess that is true in the vast majority of cases... Or rather it most likely eliminates data-dependent (deterministic) jitter but does not mean you have a jitter-free result.

Yes, that is pretty well stated. My comment was in relation to Amir's discussion about the PLL.
 

DonH50

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One thing to keep in mind is that AFAIK digital audio bit stream formats (S/PDIF, DSD, PCM and such, including the bit stream from a CD) do not have a way to synchronize independent clocks like the computer-centric PCIe and SAS standards. If the clock is just a little slower than the data stream, more data comes in than goes out on each tick of the clock, and unless you have an infinite buffer (or at least one as big as the incoming data chunk and time to clear it before the next chunk of data arrives) eventually you overrun the buffer and data is lost. The signals in your computer (and computer networks) use handshaking hardware and special signals to let each end of the link know if there's a problem and to wait and/or resend the data -- a two-way conversation. The bits from your CD stream out and on and there is no way to go back; it is a one-way conversation. This makes staying synchronized more important.

In a digital data link, you can let the recovery (receiving) clock follow the clock from the source, embedded in the data stream or not, since the most important thing is correct and complete recovery of the data. If the clock wanders off it does not matter as long as you capture that data perfectly. If you use that wandering/noisy incoming recovered/received clock as the clock for your DAC, when it wanders around you are now changing the analog output signal as the sampling rate varies (wander, jitter, frequency modulation, etc. -- distortion of the original). So, you want a perfect clock to drive your DAC, but you must provide buffering and/or some type of synchronization so you do not lose data in an asynchronous system. It's complicated.
 

Julf

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So, you want a perfect clock to drive your DAC, but you must provide buffering and/or some type of synchronization so you do not lose data in an asynchronous system. It's complicated.

This is why async USB is a Good Thing.
 

DonH50

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Good, but not easy. Except these days memory is cheap so buffering a CD not a big deal. Buffering streaming data from Internet radio or whatever strikes me as more of a challenge. Chances are data rates are slow enough you can use a little dead time between songs to catch up or something.
 

amirm

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Thanks for the post. Interestingly, this is being discussed on another forum where some are saying jitter doesn't matter since once the data is buffered it is gone. Hope you don't mind my giving a link to this thread.
Please do.
 

amirm

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This is why async USB is a Good Thing.
That definitely does away with this problem. The trick then is to make sure the device itself doesn't inject noise into its DAC. To be clear, my article talks about what jitter can arrive into the box. There are other sources of it.
 

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