On why the Naim DAC is jitter-immune (longish)

Posted by: bhaagensen on 23 January 2010

... or at least why I think it is.

Due to recent discussion threads, I hope to explain exactly why I personally find it difficult to understand how or why jitter-related issues could be an issue in the context of transports for the Naim DAC and thereby perhaps learn where my understanding is wrong. Its as such not intended for deriving conclusions about the sound-quality. This is all about jitter - some would say sound-quality is not at all about jitter Smile

It gotten a bit long, so I've divided into two parts. The first (hopefully) describes the problem and common solutions. The last part is specific to what Naim are doing. Everything written should of course be quotiented by my limited knowledge - it may be so wrong that its not even wrong. If you want the correct version, see the whitepaper Smile

======================== PART I ========================

Firstly. There are many kinds of jitter, and the significance of any given kind also depends on its frequency spectrum. This is of no importance to the final argument of the following. Moreover jitter, of the kind discussed here, is [u]by definition[/u] only created when converting the signal, analog-to-digital (AD) or the opposite direction (DA).

Considering the simplified (and not really correct, but sufficient for this purpose) chain of audio-data, lets call it a signal S, from source to Storage on a computer based device (including e.g. the HDX where applicable):

-------- / CD ------------> CD-Reader (drive) \
Studio -> Online music --> Network ------------> Storage
-------- \ Other ---------> Interface ---------/

When leaving Studio, S is already jittery from any of the AD-stages of creation. Nothing to do about this, but lets call it J0. Since there are no AD or DA stages in the chain from Studio to Storage, the jitter at the latter in S is still J0.

Let R denote the samplerate, which could be any such as 44.1,48,88,96,... Consider the simplified chain when using a streamer and an external DAC connected via S/PDIF.

Storage -> Network -> Streamer -> S/PDIF Transmitter -> Cable -> S/PDIF Receiver -> DAC Chip
------------------------------ | ------------------- | ------ | --------------------------- |
------------------------------ | ----- Clock A ------ | ------ | ---------- Clock B ---------- |
------------------------------ | ------------------- | ------ | --------------------------- |


Again, from Storage to Streamer there are no AD nor DA stages, hence at Streamer the jitter in S remains J0. The clock colums signify that all components in a column are driven by the same clock to which they must be in perfect sync to avoid jitter. Because the jitter at Streamer is J0, so only consider the chain from Stremaer to the final conversion-stage.

When Streamer sends S to the DAC (S/PDIF Receiver + DAC Chip), it does so using the S/PDIF Transmitter at a rate R provided by the clock A. The DAC Chip performs DA at a rate given by the clock B. Let At be the time between the ticks of A, and Bt the same for B. Does it work?

Consider first the idealised dream-scenario, which entails mainly the following: a) There are perfect clocks such that At=Bt. The frequcy for both is simply R, b) the S/PDIF Transmitter, Cable, and S/PDIF Receiver are 100% jitter-free. c) there is no delay, nor jitter, from the S/PDIF Receiver to the DAC-Chip. Given these assumptions the entire system works by letting B synchronise on the first incoming bit*. The system now clearly does not add any jitter at all. The amount of which at at the DAC-Chip is simply the good old J0.

Obviously, none of the above are real-world realisable. Until now most DAC chip producers have solved this by letting B be derived from A which is encoded in S. This is often refered to as: PLL, reclocking, ASRC, jitter-removal, jitter-elimination etc. Lets simply call it reclocking. Since B is now derived from S and dependent on A, lets call that clock B[A]=f(S) to signify that it depends on the signal S, via some procedure f. Pictorially:

Streamer -> S/PDIF Transmitter -> Cable -> S/PDIF Receiver -> Buffer -> DAC Chip
--------- | ------------------- | ------ | ----------------------------------- |
--------- | ----- Clock A ------ | ------ | ----------- Clock B[A] ------------- |
--------- | ------------------- | ------ | ----------------------------------- |


The Buffer is needed between that S/PDIF Receiver and the DAC Chip in order to do e.g. ASRC.

Now how does this setup relate to the dream-scenario: Ad. b) Here jitter occurs, and this is of the kind often refered to by terms such as: s/dpif generated jitter or transport-jitter. Lets call J1'. The [u]crux of the matter[/u] is that as far as jitter goes, any differences between the e.g. the Transporter and the SB3 is represented in J1'. Ad. a) Now because of J1' influence on S, the reclocking function f can in general never guarantee to obtain exactly the clock A from S. It will leave some remains of J1', which can be called J1. Ad. c) Clearly it does not hold in the theoretical sense cf. the speed of light. However the delay is neglible. Nonetheless because of this, and since the three components driven by B[A] can not be 100% synchronised, jitter does occur here. Lets call this J2. So the system is now real-world working, but at the cost of a total jitter at the DAC Chip of J0+J1+J2.


======================== PART II ========================

Now lets try and aim in on the solution used in the Naim DAC. The jitter J2 is not possible to get rid of as such**. So J1 is the target. J1 ultimately resulted because of the insufficiency of the reclocking-design. Lets backtrack and reconsider the first solution, but loose the assumption a). If it worked, there would be two independent stable clocks so J1 would be eliminated. Leaving jitter at the DAC Chip at the best possible result, wiz J0 + J1.

Unfortunately because of b) it doesn't work since no choice of stable clock B can match the jittery (c.f. b)) incoming signal. If A is too slow bits would be missed, and if its too fast there would be no availble bits from the S/PDIF Transmitter. Now lets combine and refine by adding a buffer as in the second solution and use two clocks in the DAC. One reclocking based B[A] for getting data into the DAC, and and another 100% local crystal oscilliator based B for clocking the signal out of the buffer and the DAC Chip. Pictorially:

Streamer -> S/PDIF Transmitter -> Cable -> S/PDIF Receiver -> Buffer -> DAC Chip
--------- | ------------------- | ------ | --------------- | ------------------ |
--------- | ------ Clock A----- | ------ | - Clock B[A] - | ----- Clock B ----- |
--------- | ------------------- | ------ | --------------- | ------------------ |


If now the average time (over some interval) between the ticks of B[A] was matched by B this would then work. More [u]importantly[u/], it would completely eliminate all jitter J1 since B is 100% local to the circuit clocking data out of the local buffer and driving the DAC Chip. Unfortunately such a fixed single clock B that would match all scenarios does not exist. B would very likely not match the average of the clock B[A]. Hence even the buffer would under- or overrun at some point. A helpful analogy is a waterhose with a tap at each end. Unless both taps (clocks) are set to exactly the same capasity, "holes" will occur, or "pressure" will build up. The buffer may be thought of as a finite volume resorvoir connected to the hose.

But this now hints at the final solution. Rather than fixing B at some predefined tick-rate (frequency), one simply makes available a set of 10 slightly different and cleverly chosen fixed local clocks. One of these will likely be a sufficiently close match to the average tick-rate of B[A] for the buffer not to under- or overrun***

Conclusion. The system therefore operationally works without the assumptions a), b), and c) at the lowest possible jitter level, J0+J2. In particular it is independent of the jitter J1 and J1'. It is essentially nothing but a good practical solution to the "waterhose" problem mentioned above.

Bjørn

=======================================================================================

PS If you made it here, please comment.

PSS I have deliberaty not mentioned bit-errors in transmissions. In the above, they are practically possible only from the stage of the S/PDIF Transmitter and onwards. Statistically the likelihood of such errors is extremely low AFAIK.

PSSS It is true that B may end up running at A+E. This is not jitter.

PSSSS There are too many simplifications in the above to detail them all. But please comment if I've made any that could alter the conclusion.

PSSSSS Since the secondary clock in the Naim DAC is truly independent of the clock in the transport I'd rather not use the term reclocking as it, in my book, implies a functional relationship between the two

PSSSSSS I haven't heard it... yet Smile

* This is not really feasible in general, but its a harmless assumption that becomes voided in a few lines.
** This kind of intrisic jitter obviously varies between devices depending on the quality of those. Naim has a good design and probably very low jitter at this stage.
*** I understand from others that such a match is indeed, for all intents and purposes always found, as can be seen by an indicator-light on the DAC.
Posted on: 24 January 2010 by Aleg
quote:
Originally posted by bhaagensen:
... or at least why I think it is.

{see above}



Hi Bjørn

I have read your story, and something does not 'feel quite right', but I can't put my finger on it (yet).

I think it has to do with treating jitter as an independent entity in all this, while in my understanding jitter causes / can cause a change of data which is unrecoverable. It is IMO not just a delay/speed-up, but it is a misinterpretation of the data due to the delay/speed-up.

So when J1' or J1 occurs, i.e. in the SPDIF transmitter there is, during the modulation of the SPDIF-signal, a misinterpretation of the data signal due to some instability in the used clock frequency. This misinterpretation can IMO never be recovered, because: 1) no SPDIF receiver actualy knows this misinterpretation has occured and 2) the specific instabilities of the clock that was used during SPDIF-modulation can never be recreated to detect if a misinterpretation has occurred.

So AFAIK any jitter that occurs before the SPDIF-receiver and that has caused a misinterpretation of the data signal, can never be recovered, by any technique at all.

IMO the 'only' thing that all devices in the chain can do, is to try not to add any jitter to the signal.

-
aleg
Posted on: 24 January 2010 by AMA
Bjørn,

Let me put it in simple way.
When DAC S/PDIF receiver is waiting for the next bit and the next bit is delayed for so long that DAC receivers expires the waiting time limit then the DAC receiver records 0 to buffer which is in 50% wrong.

Now tell me please -- what's wrong with this explanation?
Posted on: 24 January 2010 by bhaagensen
quote:
Originally posted by AMA:
Let me put it in simple way.
When DAC S/PDIF receiver is waiting for the next bit and the next bit is delayed for so long that DAC receivers expires the waiting time limit then the DAC receiver records 0 to buffer which is in 50% wrong.

Now tell me please -- what's wrong with this explanation?


I already said that bit-errors can occur from the point of Streamer. However such errors are statistically unlikely (this is easily tested for by comparing what you put in with what you get out). I have to dig out references, but I seem to recall the likelihood amounting to a handful of bits over the course of 24 hours continuous transmission. This is far to little to explain everyday-use audible differences.
Posted on: 24 January 2010 by pcstockton
I am not following. Doesnt the white paper explain how the jitter rejection works?

Im not sure the point of nor the message behind the above.

what are we arguing about here?
Posted on: 24 January 2010 by bhaagensen
quote:
Originally posted by pcstockton:
I am not following. Doesnt the white paper explain how the jitter rejection works?

Im not sure the point of nor the message behind the above.

what are we arguing about here?


Yes it is explained in the whitepaper. Other than that, I'm not sure what you mean as I in the 1. paragraph wrote:

quote:
Due to recent discussion threads, I hope to explain exactly why I personally find it difficult to understand how or why jitter-related issues could be an issue in the context of transports for the Naim DAC and thereby perhaps learn where my understanding is wrong.


Added: Its a curiosity on my side. I can assure you that there are no hidden motives - if that is a worry.
Posted on: 24 January 2010 by bhaagensen
Hi Aleg,

quote:
Originally posted by Aleg:
I have read your story, and something does not 'feel quite right', but I can't put my finger on it (yet).


Well, whenever something crosses your mind do post if you feel like it.

quote:
Originally posted by Aleg:
I think it has to do with treating jitter as an independent entity in all this, while in my understanding jitter causes / can cause a change of data which is unrecoverable. It is IMO not just a delay/speed-up, but it is a misinterpretation of the data due to the delay/speed-up.

So when J1' or J1 occurs, i.e. in the SPDIF transmitter there is, during the modulation of the SPDIF-signal, a misinterpretation of the data signal due to some instability in the used clock frequency. This misinterpretation can IMO never be recovered, because: 1) no SPDIF receiver actualy knows this misinterpretation has occured and 2) the specific instabilities of the clock that was used during SPDIF-modulation can never be recreated to detect if a misinterpretation has occurred.

So AFAIK any jitter that occurs before the SPDIF-receiver and that has caused a misinterpretation of the data signal, can never be recovered, by any technique at all.

IMO the 'only' thing that all devices in the chain can do, is to try not to add any jitter to the signal.



I'm not sure I understand exactly what you are saying. Its regarding 'misinterpretations'. Isn't the worst error-continue condition that a modulation error caused one of the following: That a bit is flipped? That the resulting encoding is jitter-infected? Or a combination. If so I would refer to my answer to AMA above for the "flipping" case. For the second case, that amount of jitter is accounted for by J1'. Or do you mean something else?

Actually J1' includes the combined jitter from clock A, S/PDIF Transmitter and Receiver, and Cable. Maybe that causes problems for the explanation, though I think the overall conclusion is unaffected.
Posted on: 24 January 2010 by bhaagensen
DELETED

On second thought I don't want people reading this make too strong associations with the whitepaper, as it is purely my very amateurish interpretation.

The 2. paragraph already explains this.
Posted on: 24 January 2010 by fatcat
Bjorn

The S/PDIF interface carries the data stream, plus, it also carries the timing information used by the DAC to construct the analogue waveform. The Naim DAC eliminates the S/PDIF interface induced timing jitter but doesn’t eliminate other forms of jitter.

Jitter Explained

I suspect the 10 carefully selected clock speeds used in the DAC aren’t to accommodate inaccuracies in the transport clock. It’s due to the fact different manufactures use different clock speed.

Ranging from 8.4672 To 45.1584 MHz
Pioneer PD-9700 16.9344MHz
Pioneer PDR-04 16.9344MHz
PS Audio Lambda 11.2896MHz
Technics SL PS670A 33.868MHz
Nad 5425 33.868MHz
Nakamichi 16.9344MHz
Sony X779 ES 45.1584MHz
Posted on: 24 January 2010 by bhaagensen
Hi Aleg:

I thought a bit more about what you wrote, and I think there are scenarios where misinterpretation could be detected at least? E.g. if the (Manchester encoded) signal received does not exhibit a transition roughly at the middle of a bits "live-time".

I'm too tired to see if the correct value can be recovered though... Do you know?

Anyway, I just wrote this as a side, hoping perhaps it could result in me better understanding your post.
Posted on: 24 January 2010 by AMA
quote:
I already said that bit-errors can occur from the point of Streamer. However such errors are statistically unlikely (this is easily tested for by comparing what you put in with what you get out). I have to dig out references, but I seem to recall the likelihood amounting to a handful of bits over the course of 24 hours continuous transmission. This is far to little to explain everyday-use audible differences.

Bjorn,

These "statistically unlikely bit-errors from streamer" are quite numerous and called JITTER. This is exactly what hi-end companies are fighting against since late 80-s. Millions of bits are delayed or overtake every second of music because the transport is not perfect in reading and clocking them properly. It has nothing to do with droping the bits during transmission. This is a jitter which is induced by poor clocking of streamer S/PDIF generator.
What you have to do is to watch the S/PDIF clock on oscilloscope once -- you will be amazed.
You expect to see sinusoidal pulses with perfect time spacing???
Come on -- a white noise with sinusoidal trend lines and smeared in time (because of gap variation).
To clearly separate the bits in 705.6 kHz bistream you have to work out a super-fine clock -- the one which can be found in the most expensive gears. For proper hi-res streaming you have to make up even finer clock. And it's not ONLY about the clock generator itself -- the power supply can kill the quality of the finest clock generator you can imagine. To get the best from clock generator you have to feed it with linear PS. This what hi-end companies do in the top products. Check pink fish forum -- you can find the oscilloscope of SB3 impulse power supply. And now you believe that this jagged power can feed a superfine clock with ultimate precision? What if you need to burst 10 pulses the next moment but PS does not supply enough power at this moment -- in terms of SB3 analogue output (which is poor on it's own) this is not audible, but in terms of bitsream which is then clocked to the top DAC with superfine analogue output you are missing a number of fine detailes, broken soundstage, compressed dynamic range, overdrive and noise. The gaps between bits sometimes extends so much that even re-clocking DAC can not handle them. But the next moment you got a storm of messing bits which re-clocking DAC can not identify because they are so close in time. Is it still bit-perfect bitstream? Yes. Is it time-perfect? No.

As I can see from your header post your line of thoughts is absolutely correct except you underestimate the scale and impact of JITTER and overestimate the abilities of re-clocking DAC. But if you pair super-fine transport with re-clocking DAC -- this combo can perform with almost 0-jitter. This is what I plan to do with my Transporter and Naim DAC which I'm waiting for.

Let me ask you another question -- if Naim DAC is immune to jitter -- how could it happen than CD5XS and CDX2-2 which have superb CD transports and proprietary S/PDIF generators can sound so audibly different? Or you believe that Logitech managed to build a better S/PDIF generator than Naim's engineers?
Posted on: 24 January 2010 by bhaagensen
AMA, I must admit I don't really understand exactly all you wrote. So I'll try and comment and ask as detailed as possible.

quote:
Originally posted by AMA:
These "statistically unlikely bit-errors from streamer" are quite numerous and called JITTER. This is exactly what hi-end companies are fighting against since late 80-s. Millions of bits are delayed or overtake every second of music because the transport is not perfect in reading and clocking them properly. It has nothing to do with droping the bits during transmission.


OK, I should have made clear what I meant by bit-errors. Suppose an audiofile is sent through the chain drawn in the first two figures in the header post. But where the S/PDIF Receiver is located on the digital input on a computer (Instead of the DAC). Run this system and record the data received in another file. Then compare the two files. If they are identical no bit-errors have occured. If they are not, bit-errors have occured. By analysis of the files the exact number of bit-errors can also be determined. This is the number I claim is statistically very low.

quote:
Originally posted by AMA:
This is a jitter which is induced by poor clocking of streamer S/PDIF generator.


Agree.

quote:
Originally posted by AMA:
What you have to do is to watch the S/PDIF clock on oscilloscope once -- you will be amazed. You expect to see sinusoidal pulses with perfect time spacing??? Come on -- a white noise with sinusoidal trend lines and smeared in time (because of gap variation).


I have seen such graphs, and still agree.

quote:
Originally posted by AMA:
To clearly separate the bits in 705.6 kHz bistream you have to work out a super-fine clock -- the one which can be found in the most expensive gears. For proper hi-res streaming you have to make up even finer clock. And it's not ONLY about the clock generator itself -- the power supply can kill the quality of the finest clock generator you can imagine. To get the best from clock generator you have to feed it with linear PS. This what hi-end companies do in the top products. Check pink fish forum -- you can find the oscilloscope of SB3 impulse power supply.


I have not seen the this reference, but I do agree.

quote:
Originally posted by AMA:
And now you believe that this jagged power can feed a superfine clock with ultimate precision?


No I don't belive so. So in conclusion so far, we agree that S/PDIF introduces jitter all the time which can be verified by scoping on the S/PDIF? If so, I have not AFAIK said anything in contradiction to this.

quote:
Originally posted by AMA:
What if you need to burst 10 pulses the next moment but PS does not supply enough power at this moment -- in terms of SB3 analogue output (which is poor on it's own) this is not audible, but in terms of bitsream which is then clocked to the top DAC with superfine analogue output you are missing a number of fine detailes, broken soundstage, compressed dynamic range, overdrive and noise.


Ignoring the analoge case and sticking to the digital case since it is closer to the subject matter, I can not see wherin this, or the above, the arguments supporting such a conclusion is; other than the brief "jitter is teh cause". In which case my reply would be that this is exactly what the Naim DAC avoids. Did I misunderstand?

quote:
Originally posted by AMA:
The gaps between bits sometimes extends so much that even re-clocking DAC can not handle them.


In the previous post it was argued that when the delay exceeds the "timeout" of the Receiver, it will go fail in 50% of the cases. I do not disagree, and I think I replied to this. Assuming this did not happen, the late bit will simply be forwarded to the buffer as always and everything still works. I can only say that the lateness then does not matter because the master-clock in the Naim DAC is not directly related to the gaps between bits.

quote:
Originally posted by AMA:
But the next moment you got a storm of messing bits which re-clocking DAC can not identify because they are so close in time. Is it still bit-perfect bitstream? Yes. Is it time-perfect? No.


OK, here I must clarify what I understand by "re-clocking DAC can not identify". So I'll spend a few moments on this.

I assume bit-perfect entails two things. a) No bit-errors as described in my 1. answer. b) No jitter. That makes us in agreement on the last two questions and answers. Although this is not how I interpret Naim's definition on the whitepapers last page. By the way, on the same page Naim themselves state that the DAC "removes all S/PDIF induced jitter"... of course one shouldn't believe everything coming out of the horses mouth, but this is after all an official Naim statement.

Ad a) This is, by the 1. answer, something that happens very rarely. This, to me, implies that the meaning must be that the re-clocking DAC can not re-clock correctly, so I answer with respect to that.

I agree that no re-clocking system which depends on the bit-stream, and in particular on the time-encoding therein, can work perfectly. But Naim does not use such a system: the clock in the Naim DAC does not have this dependence. It is not derived from the bit-stream and exhibits nothing but intrisic jitter. This is why I do not think that the Naim DAC should be called a 're-clocking' DAC.

quote:
Originally posted by AMA:
As I can see from your header post your line of thoughts is absolutely correct except you underestimate the scale and impact of JITTER and overestimate the abilities of re-clocking DAC. But if you pair super-fine transport with re-clocking DAC -- this combo can perform with almost 0-jitter. This is what I plan to do with my Transporter and Naim DAC which I'm waiting for.


Using a re-clocking DAC (see above for my understanding of a reclocking DAC) I do agree with you.

quote:

Let me ask you another question -- if Naim DAC is immune to jitter -- how could it happen than CD5XS and CDX2-2 which have superb CD transports and proprietary S/PDIF generators can sound so audibly different? Or you believe that Logitech managed to build a better S/PDIF generator than Naim's engineers?


This is really the question that I myself am trying to home in on an answer to, because I do not really know. I may have some ideas, but I do not have the knowledge to use them yet. An interesting observation is also that Naim recommends connecting the DAC using optical rather than coax. Optical always performs worse than coax with respect to jitter since it entails an extra conversion step. Based on this recommodation, one could speculate in that they indeed have great faith in the DAC's ability to reject jitter - or alternatively, they could be of the opinion that jitter is not really that important for sound quality... (I don't really buy the latter).
Posted on: 25 January 2010 by bhaagensen
Let me try and put it in another more pictorial way. Consider the last drawing - the one of the Naim DAC.

Streamer -> S/PDIF Transmitter -> Cable -> S/PDIF Receiver -> Buffer -> DAC Chip
--------- | ------------------- | ------ | --------------- | ------------------ |
--------- | ------ Clock A----- | ------ | - Clock B[A] - | ----- Clock B ----- |
--------- | ------------------- | ------ | --------------- | ------------------ |

Even if the bits where delivered to the Buffer by Royal Mail the Naim DAC would still be immune to any variation in Royal Mails service times. The only requirement is that Royal Mail deliver the bits fast enough on average for the buffer not to empty. I.e. if they need to deliver a total of 35 * K bits a week, they could freely choose to deliver 5 * K bits/day monday-sunday, or 7 * K bits/day monday-friday and take the weekend off.

That the delivery-requirement holds in the real world is demonstrated by the fact that the Naim DAC seems to run in sync-mode all the time.
Posted on: 25 January 2010 by bhaagensen
Hi fatcat (strange greeting someone using that wordSmile

Thanks for your reply. Sorry mine is late: I simply did not notice until now.

quote:
Originally posted by fatcat:
The S/PDIF interface carries the data stream, plus, it also carries the timing information used by the DAC to construct the analogue waveform. The Naim DAC eliminates the S/PDIF interface induced timing jitter but doesn’t eliminate other forms of jitter.

Jitter Explained


Indeed, and this should be covered in my 1. post. I could perhaps add that what lead me to write the 1. post was discussions regarding the difference between digital transports when feeding a Naim DAC - hence the springing point, jitterwise, is s/pdif introduced jitter.

quote:
Originally posted by fatcat:
I suspect the 10 carefully selected clock speeds used in the DAC aren’t to accommodate inaccuracies in the transport clock. It’s due to the fact different manufactures use different clock speed.


Well, all I can say is that they are Smile In fact they compensate for inaccuracies in the entire S/PDIF chain. The matching issue is a springing point. However there is a light on the front of the DAC which lights up when sufficiently good match is found. Several of those who have actually seen/heard the DAC in action reports to never have seen this not on. Also Naim write in the whitepaper that a match is very likely to be found. Obviously, this is configuration-dependant, and I guess the exact match/no match ratio will only become known when the DAC has seen more widespread use.

quote:
Originally posted by fatcat:
Ranging from 8.4672 To 45.1584 MHz
Pioneer PD-9700 16.9344MHz
Pioneer PDR-04 16.9344MHz
PS Audio Lambda 11.2896MHz
Technics SL PS670A 33.868MHz
Nad 5425 33.868MHz
Nakamichi 16.9344MHz
Sony X779 ES 45.1584MHz


I'm not sure I am with you here. Surely a transport feeding the DAC with a 16/44 signal, would also use a clock as close as possible to 44 - alternatively one that can be precisely scaled to that rate?
Posted on: 25 January 2010 by AMA
quote:
An interesting observation is also that Naim recommends connecting the DAC using optical rather than coax.

Bjorn, Naim always recommend to use their BNC DC1 cable because of 70 Ohm impedance matching. Meanwhile they tried to develop an RCA option (on the base of Neutriks I guess -- but not sure) which is still higher than 70 Ohm and still features a detrimental impedance step in connector-cable. They can only recommend for tosslink if transport does not features BNC or transport has floating earth ground which may transmit a hum through the coax ground to the naim amps. But new Naim DAC features the switch to the floating earth which allow using coax with all transports.

But again -- BNC coax is the Naim choice.
Posted on: 25 January 2010 by bhaagensen
quote:
Originally posted by AMA:
quote:
An interesting observation is also that Naim recommends connecting the DAC using optical rather than coax.

Bjorn, Naim always recommend to use their BNC DC1 cable because of 70 Ohm impedance matching. Meanwhile they tried to develop an RCA option (on the base of Neutriks I guess -- but not sure) which is still higher than 70 Ohm and still features a detrimental impedance step in connector-cable. They can only recommend for tosslink if transport does not features BNC or transport has floating earth ground which may transmit a hum through the coax ground to the naim amps. But new Naim DAC features the switch to the floating earth which allow using coax with all transports.

But again -- BNC coax is the Naim choice.


OK, that may well be correct.
Posted on: 25 January 2010 by bhaagensen
Hi fatcat,

Thanks for your reply. I think my earlier reply got sucked into some black hole of moderation - its gone anyway, so here it is again.

quote:
Originally posted by fatcat:
The S/PDIF interface carries the data stream, plus, it also carries the timing information used by the DAC to construct the analogue waveform. The Naim DAC eliminates the S/PDIF interface induced timing jitter but doesn’t eliminate other forms of jitter.


Yes I know this, and it stated in the first part of my header post. Did you read it?

quote:
Originally posted by fatcat:
I suspect the 10 carefully selected clock speeds used in the DAC aren’t to accommodate inaccuracies in the transport clock. It’s due to the fact different manufactures use different clock speed.


Yes they are there to accomodate inaccuracies in the transport clock (but see also my answer below as we are possibly not on wavelength here). But additionally they account for all inaccuracies in the S/PDIF chain including transmitter, cable, and receiver. If my header post does not explain this properly, I can only refer to the whitepaper.

quote:

Ranging from 8.4672 To 45.1584 MHz
PD-9700 16.9344MHz
PDR-04 16.9344MHz
11.2896MHz
SL PS670A 33.868MHz
33.868MHz
16.9344MHz
45.1584MHz


I don't think I understand this? Surely a transport feeding a dac with say 44 material would use an oscillator that matches that value as close as possible? Alternatively one from which 44 can be stably derived?
Posted on: 25 January 2010 by bhaagensen
AMA: As for your Transporter feeding the DAC plans, you might be interested in also this:

quote:
Sean Adams:
"TP uses two different transmission circuits on each of the respective interfaces. The BNC has a transformer whereas the RCA has a capacitor. Capacitor coupled outputs are most common these days, but the transformer has the advantage that it isolates the grounds between the two devices."
Posted on: 25 January 2010 by fatcat
quote:
Originally posted by bhaagensen:


quote:
Originally posted by fatcat:
I suspect the 10 carefully selected clock speeds used in the DAC aren’t to accommodate inaccuracies in the transport clock. It’s due to the fact different manufactures use different clock speed.


Well, all I can say is that they are Smile In fact they compensate for inaccuracies in the entire S/PDIF chain. The matching issue is a springing point. However there is a light on the front of the DAC which lights up when sufficiently good match is found. Several of those who have actually seen/heard the DAC in action reports to never have seen this not on. Also Naim write in the whitepaper that a match is very likely to be found. Obviously, this is configuration-dependant, and I guess the exact match/no match ratio will only become known when the DAC has seen more widespread use.

quote:
Originally posted by fatcat:
Ranging from 8.4672 To 45.1584 MHz
Pioneer PD-9700 16.9344MHz
Pioneer PDR-04 16.9344MHz
PS Audio Lambda 11.2896MHz
Technics SL PS670A 33.868MHz
Nad 5425 33.868MHz
Nakamichi 16.9344MHz
Sony X779 ES 45.1584MHz


I'm not sure I am with you here. Surely a transport feeding the DAC with a 16/44 signal, would also use a clock as close as possible to 44 - alternatively one that can be precisely scaled to that rate?


Bjorn

Well, I am 99.99% sure they aren’t.
Below are more examples of clock frequencies used in various machines.

Naim CD5 16.9344MHz
Naim NA CD3 16.9344MHz
Meridian 500 11.2896MHz
Meridian 506 11.2896MHz
Arcam 70.2/ 70.3 11.2896MHz
Arcam 170.3 11.2896MHz
Sony 761 45.1584MHz
Quad 66 11.2896MHz
Quad 67 11.2896MHz
Sony CDP 497 22.5792MHz
Sony CDP 555 ES 16.9344MHz

There appear to be 10 common oscillator speeds between 8.4672 and 45.1584 MHz.

The Naim DAC identifies the clock frequency of the transport, if DAC contains an oscillator of the same frequency the sync light illuminates.

Pioneer DVD players use a frequency of 27.0000MHz. This is uncommon, I would expect the DAC not to sync. If I could turn on the digital output of my DV717 on, I could confirm this.

I can confirm that a good Rotel CD player fitted with a high performance clock performs better than a Toshiba DVD player costing £300. Even when the DVD player is spinning a DVD Audio disc.
Posted on: 25 January 2010 by bhaagensen
Hi,

quote:
Originally posted by fatcat:

Well, I am 99.99% sure they aren’t.
Below are more examples of clock frequencies used in various machines.

Naim CD5 16.9344MHz
Naim NA CD3 16.9344MHz
Meridian 500 11.2896MHz
Meridian 506 11.2896MHz
Arcam 70.2/ 70.3 11.2896MHz
Arcam 170.3 11.2896MHz
Sony 761 45.1584MHz
Quad 66 11.2896MHz
Quad 67 11.2896MHz
Sony CDP 497 22.5792MHz
Sony CDP 555 ES 16.9344MHz

There appear to be 10 common oscillator speeds between 8.4672 and 45.1584 MHz.

The Naim DAC identifies the clock frequency of the transport, if DAC contains an oscillator of the same frequency the sync light illuminates.

Pioneer DVD players use a frequency of 27.0000MHz. This is uncommon, I would expect the DAC not to sync. If I could turn on the digital output of my DV717 on, I could confirm this.


OK, now I see where the misunderstanding lies. The frequencies you quote are the crystal oscillator base frequencies - note that they are in MHz rather than kHz These are constant in any given device and used for deriving the actual frequencies that a given device require(s). I don't think crystal oscillators with the required low freqency for audio applications even exits - they would have to go down to at least 20Hz for redbook audio.

Note that a device can require several derived clocks. Consider e.g. an integrated cd-player with an oversampling DAC at 384kHz. This device needs 44.1 for feeding DSP (upsampler) and 384 for feeding and driving the DAC Chip and the upsampler itself (the DSP and DAC chip is often integrated these days - notheless both frequncies are needed)

Now the point is that in a separate transport-DAC system the S/PDIF receiver knows nothing about the transports oscillators frequency. It only knows the clock frequency encoded in the S/PDIF stream - which is something like 44.1, 48, 88, 96 etc.* For any fixed of those frequncies this is the one frequncy one of the 10 clocks within the Naim DAC must match - though only on average as explained in my header post. Also note that the crystal oscillator in the DAC does not need to have the same frequncy as the crystal oscillator in the Transport because - well as I said, it is only used for deriving. Of course here the point is that the Naim DAC uses the former for receiving data in its buffer and the latter independent local crystal clock for driving the DAC Chip. Hence (s/pdif) jitter is completely removed.

* Once derived there is no way to reverse the process, i.e. find the frequncy of the oscillator. Deriving is basically done by division and multiplication. It corresponds to if I give you the number 2 and tell you it was derived by dividing and multiplying either, 4,6,8,10. There is no way you could tell which from that information only.

Makes sense?
Posted on: 26 January 2010 by AMA
quote:
Originally posted by bhaagensen:
AMA: As for your Transporter feeding the DAC plans, you might be interested in also this:

quote:
Sean Adams:
"TP uses two different transmission circuits on each of the respective interfaces. The BNC has a transformer whereas the RCA has a capacitor. Capacitor coupled outputs are most common these days, but the transformer has the advantage that it isolates the grounds between the two devices."


Yes, thanks a lot. I know that and I plan to connect TP with Naim DAC through BNC.
Posted on: 26 January 2010 by fatcat
quote:
Originally posted by bhaagensen:

Makes sense?


A little, but that’s my fault not yours. Smile Thanks for explaining oscillator base frequency.

However, are you now stating the Naim DAC clocks are set to 44.1, 48, 88, 96 etc, and not to 10 slightly different speeds.

quote:
Also note that the crystal oscillator in the DAC does not need to have the same frequncy as the crystal oscillator in the Transport because - well as I said, it is only used for deriving. Of course here the point is that the Naim DAC uses the former for receiving data in its buffer and the latter independent local crystal clock for driving the DAC Chip. Hence (s/pdif) jitter is completely removed.



The above being correct, the Naim DAC uses the transport clock when interpreting the data stream entering the buffer. It therefore follows that the accuracy of the data entering the buffer is dependent on the accuracy of the clock information that is transmitted through the S/PDIF. Data that is lost during this process is lost for ever. Reclocking the information in the buffer, won’t recover the lost data.