Why is the nDAC so cheap?

Posted by: Andy S on 04 May 2010

Serious question.

Have Naim scored an own goal? Using a cheap PC and optical to DAC on it's own is such a massive boost over my old CDS1 it just isn't funny and a mate is selling his CDS3 head end as the PC/DAC/XPS is as close as you could get to a CDS3. Not only that, I can connect up a number of sources and get benefit - the TV sounds SO much better through it.

Don't get me wrong, I'm not complaining since I've just bought one (the demo only lasted 5 minutes in all honesty - the distance was that big), just curious...
Posted on: 07 May 2010 by Andy S
quote:
Originally posted by ghook2020:
Just read through the white paper...again.

Was looking for any hints about what may cause transports to sound different, and found three comments.

[QUOTE] 1. As with all Naim designs, the influence of vibration-induced microphonic noise has been minimised in the Naim DAC

Minimized, but not eliminated. So perhaps different transports have different vibration characteristics?
Talking about two different things. Microphonics generally applies to the analogue signals and you can reduce their effect by mechanically isolating the PCBs carrying the audio circuits. Unless the DAC is on the same surface as the transport, assuming the transport reads bit perfectly every time (which a streamer would) it doesn't affect sound quality.

quote:
2. CD, DVD player – connect via S/PDIF (watch out for the switch mode power supplies of cheap digital players and the quality of S/PDIF, which may suffer RF noise)

Different transports have different power supplies, so couldn't they contribute different amounts of RF?
Yup, but if you connect via optical, you've isolated this anyway (other than back through the mains which you have to deal with in any case).

quote:
3. Coax can sound better but optical has the advantage that it prevents ground loops and isolates the ground system of the source, which may be noisy, from that of the Naim DAC.

If different transports have different ground schemes, could not some be more effective than others at preventing noise from traveling through digital coax connections?


Yes, totally, and that's why Naim has fitted the float/chassis switch...


quote:
So while the white paper does suggest that there are differences when connected via BNC and/or RCA S/PDIF, I did not see anything to suggest why different transports connected via Toslink could sound different. But even this contradicts my own experience...

Anyway, if we accept (as the white paper says) that transports connected via digital coax can sound better than those connected via Toslink, then must we not also accept that vibration and/or RF noise via digital coax may have a role in perceived sound quality? If so, then could this help to explain why different transports may not all sound alike?

Hook
Does the white paper quote that Coax can sound better with the nDAC Confused I think the generally held opinion is that it does on DACs been tried up to now, but why would it with this implementation?

Anyway, the reason for the expanded thread is to understand how the differences are possible...
Posted on: 07 May 2010 by AMA
quote:
AMA - I give up.

Andy, I'm trying to understand, possibly to learn.
But I like they way you build a professional sequence of objections rather than emotional posts. I think we came very close to the key point where our perceptions of the bitstream transmission diverge. The reference to self-clocking of SPDIF did not work for me as recovering the clock itself from the bistream is the main issue. Before you recover a clock -- you have to secure that you read a word (a consequent number of bits) correctly. If you can't read the bits correctly -- how can you be sure that you recovered a clock correctly or decode the PCM protocol correctly?
That's where our perceptions differ.

Can you clearly answer my questions so I can see your ground and possibly change my mind:

1. Do you confirm that in high jitter PCM bistream bits often come very close to each other -- say less than 100 ps?

2. If you agree with 1 -- do you agree that SPDIF will not resolve them?

3. If you don't agree with 1 -- do you claim it never happens or it happens so rarely so it's not audible? Can you suggest the proof -- a kind of your personal oscilloscope observations etc.

I'm not a digital engineer -- if you can provide a technical background I would appreciate and willingly change my mind.

I'm looking for the truth -- not for the "spending time in chat". I'm not so arrogant to stick on my point if someone suggests a clear picture of my mistake.

BTW this has nothing to do with that I clearly hear the difference in transports.
This is only about my explanation of this phenomenon -- which can be right or can be wrong.
Posted on: 07 May 2010 by Hook
quote:
3. Coax can sound better but optical has the advantage that it prevents ground loops and isolates the ground system of the source, which may be noisy, from that of the Naim DAC.

....

Hook
Does the white paper quote that Coax can sound better with the nDAC Confused I think the generally held opinion is that it does on DACs been tried up to now, but why would it with this implementation?

....

The quote "Coax can sound better..." came from the Naim DAC White Paper.

I do not see how you could interpret the quote as referring to anything other than transports connected to the DAC.

You seem to be implying that Naim is trying at best to be clever, or at worst deceitful. Instead, and until proven otherwise, I think I will continue to believe that Naim is a company I can trust.

As to exactly why the DAC sounds better (or worse), the white paper does suggest that poor ground schemes and/or RF noise could play a role. So clearly the DAC's circuitry is very sensitive. If it were not, then there would be no reason for Naim to suggest using Toslink for connecting computers.

Of course I do not have the technical ability to discuss exactly how these sources of noise could effect the DAC circuitry and resulting sound quality, but if they didn't, why would the white paper mention them at all?

Again, hopefully we receive some clarification from Naim engineering. In the meantime, would hate to see Naim's reputation tarnished by assumptions that are potentially based on incomplete information.

Hook
Posted on: 07 May 2010 by Andy S
quote:
Originally posted by AMA:
quote:
AMA - I give up.

Andy, I'm trying to understand, possibly to learn.
But I like they way you build a professional sequence of objections rather than emotional posts. I think we came very close to the key point where our perceptions of the bitstream transmission diverge. The reference to self-clocking of SPDIF did not work for me as recovering the clock itself from the bistream is the main issue. Before you recover a clock -- you have to secure that you read a word (a consequent number of bits) correctly. If you can't read the bits correctly -- how can you be sure that you recovered a clock correctly or decode the PCM protocol correctly?
That's where our perceptions differ.

Can you clearly answer my questions so I can see your ground and possibly change my mind:

1. Do you confirm that in high jitter PCM bistream bits often come very close to each other -- say less than 100 ps?
I think I'm beginning to see your misunderstanding (and I'll try one last time Winker)

Bits ALWAYS come next to each other in PCM streams - there can be no gaps in a system that has two states - 1 or 0. The important thing is that the transition point between bits is referenced to a clock - NOT when the state changes from one value to another. This is done as it is an impossibility to go from state 0 to state 1 (or the other way around) instantaneously and as a designer, you have to deal with that to get the data back out otherwise you can't guarantee the system will work! The way to ensure this is typically done by clocking the data half way through the cycle to allow for everything to settle.

Imagine a stream of bits - each one taking 100ns of time. Also lets suppose that the physical characteristics of the cable carrying them means it takes 10ns to go from one state to another. This means that you have 90ns of "steady" state followed by 10ns of change time. So, to make sure we always reliably get this data, we sample (i.e. clock) the data half way through the steady state - that is we wait 40ns before we capture the data. If the position of the pulse is out by a little, it doesn't matter. It has to be out by A LOT before you get the wrong result (i.e. tens of nanoseconds). There is no concept of the bit moving a little bit and not being able to capture it. You can reliably recover the bit because you wait for it to settle before you look at it and the bit can't be out by a lot otherwise the system will be completely unstable and unusable.

Basically, a digital system has to be stable and transmit all bits reliably, otherwise it can't be guaranteed to work.

quote:
2. If you agree with 1 -- do you agree that SPDIF will not resolve them?

3. If you don't agree with 1 -- do you claim it never happens or it happens so rarely so it's not audible? Can you suggest the proof -- a kind of your personal oscilloscope observations etc.


Extending this to SPDIF - it just uses a more complex clocking system to pull data out of the bitstream (as you have encoded the word clock, bit clock and data into a single stream). It's still 100% reliable as you use the encoding to be able to sample the data when it is at it's steady state (if you don't believe me, read how the systems are encoded and how you pull the data out of the bitstream). You have to have a system which is SERIOUSLY out of balance not to work. Unfortunately though, as you are recovering both clocks and data, you will get small but measurable TIME errors in this process due to the upstream factors (quality of output, cable type etc...). These are ONLY timing errors and NOT data errors - because the way the system works guarantees you get the data.

As to the effect of jitter in all this - it is non existent in pulling out the digital data. The amount of jitter we're talking about is in the PICOseconds range which is 100-1000 times less than the length of time of the data high or low period and won't cause you to lose ANY information if this is in the data clock.

Where it will cause you issues is when you convert that signal back into analogue. If you have a perfect DAC that always gets to exactly the right voltage and you time it perfectly, you have a perfect reproduction of what went in. Unfortunately, if the clock is just slightly non stable you introduce distortions onto the waveform as the perfect voltage is reached sometimes a very small amount before it should be, sometimes exactly when it should be and sometimes a little later than it should be. This is where jitter becomes apparent as it is these very small but audible changes which cause the loss of fidelity in what you hear back through the speakers.

quote:
...BTW this has nothing to do with that I clearly hear the difference in transports.
This is only about my explanation of this phenomenon -- which can be right or can be wrong.
Yup... Agree. I hope I've explained why once you have the data off the disc, you don't lose any data anywhere in the system and the effect of jitter is heard only at the DAC stage which, if you've reclocked, should be consistent for all sources.

Crikey, I feel I've written a short novel over the last few days!!
Posted on: 07 May 2010 by Andy S
quote:
Originally posted by ghook2020:
quote:
3. Coax can sound better but optical has the advantage that it prevents ground loops and isolates the ground system of the source, which may be noisy, from that of the Naim DAC.

....

Hook

quote:
Does the white paper quote that Coax can sound better with the nDAC Confused I think the generally held opinion is that it does on DACs been tried up to now, but why would it with this implementation?


....

The quote "Coax can sound better..." came from the Naim DAC White Paper.

I do not see how you could interpret the quote as referring to anything other than transports connected to the DAC.
Depends who wrote it and what they were trying to communicate. I can read it both ways - "generally you get better sound with coax" or "we've found that we get better sound on the nDAC with coax". It's especially confusing as the second part of the sentence refers to optical being more advantageous for ground loops (something which could also apply to all DACs or this one in particular).


quote:
You seem to be implying that Naim is trying at best to be clever, or at worst deceitful. Instead, and until proven otherwise, I think I will continue to believe that Naim is a company I can trust.
No, I'm trying to imply that the document was written by a set of guys who deal with this stuff on a daily basis - you only have to read the detail of the I2V conversion and even my eyes glaze over - and they will have a knowledge base about this which is world class. The way they write will reflect that and they will leave out small but vital information as they fill in the gaps as they just know it... and people like you and me will analyse it to death. FGS look how many people make a living interpreting Shakespeares plays for instance!

quote:
As to exactly why the DAC sounds better (or worse), the white paper does suggest that poor ground schemes and/or RF noise could play a role. So clearly the DAC's circuitry is very sensitive. If it were not, then there would be no reason for Naim to suggest using Toslink for connecting computers.

Of course I do not have the technical ability to discuss exactly how these sources of noise could effect the DAC circuitry and resulting sound quality, but if they didn't, why would the white paper mention them at all?

Again, hopefully we receive some clarification from Naim engineering. In the meantime, would hate to see Naim's reputation tarnished by assumptions that are potentially based on incomplete information.

Hook
Yup.
Posted on: 07 May 2010 by james n
quote:
Crikey, I feel I've written a short novel over the last few days!!


Yes - and very interesting its all been too. Just try a decent transport, report back and then we can all go home Big Grin

James
Posted on: 07 May 2010 by Andy S
quote:
Originally posted by james n:
quote:
Crikey, I feel I've written a short novel over the last few days!!


Yes - and very interesting its all been too. Just try a decent transport, report back and then we can all go home Big Grin

James
LOL.. I've been listening again to USB vs optical from htpc - still nothing to report - and I've been trying hard too (in between writing my novel!).
Posted on: 07 May 2010 by nap-ster
quote:
Originally posted by AllenB:
Too much theory on this thread Roll Eyes


As soon as the OP said that he couldn't hear a difference between transports I lost interest.
Posted on: 07 May 2010 by AMA
Andy, reading your objections I start understanding the weak points in my story (as well as your). You helped me to understand that I have NO straightforward confirmation that a cross-line between bits can drift substantially (in a poorly clocked bitstreams) and lead to messing the bit values -- while you state the opposite. I'm happy that we finally came to the key point of our discussion. I will check this up with my engineers. Thanks again!
Posted on: 07 May 2010 by alidubai
quote:
Originally posted by AMA:
Andy, reading your objections I start understanding the weak points in my story (as well as your). You helped me to understand that I have NO straightforward confirmation that a cross-line between bits can drift substantially (in a poorly clocked bitstreams) and lead to messing the bit values -- while you state the opposite. I'm happy that we finally came to the key point of our discussion. I will check this up with my engineers. Thanks again!


I'd like to look at it this way.

As all of you know I have some "mass market" sources that I use.

I brought the NDAC home for a test, not knowing what to expect. After all this is a GBP 2000 piece of kit. I really did not know what to expect.

After playing it within a few minutes - it proved it was worth the price tag.

Not only with the bass and treble and all the other hifi checklist, but it had an energy and groove to the way it played music.

It's a fine musical "instrument"....
Posted on: 07 May 2010 by AMA
quote:
As soon as the OP said that he couldn't hear a difference between transports I lost interest.

The fact that you're posting on Page 10 means the opposite Winker
Posted on: 07 May 2010 by nap-ster
quote:
Originally posted by AMA:
quote:
As soon as the OP said that he couldn't hear a difference between transports I lost interest.

The fact that you're posting on Page 10 means the opposite Winker


Good point.

If you get bogged down in the theory though it might It not be a case of the Emperors New Clothes though? The theory says it isn't possible so I WILL NOT hear a difference.
Posted on: 07 May 2010 by Andy S
quote:
Originally posted by AMA:
I'm happy that we finally came to the key point of our discussion. I will check this up with my engineers. Thanks again!
Glad it helped Smile I'm still confused as to what could be causing people to hear differently and would love to get to the bottom of it. I'll update the forum when I've borrowed a "decent" transport.

BTW, even the Naim guys front their DACs with PCs. See: http://forums.naim-audio.com/e...8019385/m/8132904137 Winker
Posted on: 08 May 2010 by AMA
quote:
I'm still confused as to what could be causing people to hear differently and would love to get to the bottom of it.

Andy, so far I only managed to capture one digital engineer from my colleagues. He designs digital electronics for subsurface data acquisition which includes data transmission along very long coaxial cables (several kilometers). They use Manchester coding. He never got into the details of SPDIF but he knows bi-phase coding. He said that he never seen a stock DVD clock onscreen but he believes that clock drifting is hardly that big to mess the bits. He also thinks that different transports differ because they read data from disc with mistakes -- some better some worse.

This also means that quality of digital cable is not important for re-clocking DACs -- which is a good idea to check. I will grab a couple of cables one day and check if they sound the same as my Naim DC1 (BNC-BNC).
Posted on: 08 May 2010 by Andy S
quote:
Originally posted by AMA:
He said that he never seen a stock DVD clock onscreen but he believes that clock drifting is hardly that big to mess the bits.
Exactly Big Grin

quote:
He also thinks that different transports differ because they read data from disc with mistakes -- some better some worse. [/|QUOTE]Which could explain how come a htpc with bit perfect rips sound as good as USB sticks with wav files Smile


[QUOTE]This also means that quality of digital cable is not important for re-clocking DACs -- which is a good idea to check. I will grab a couple of cables one day and check if they sound the same as my Naim DC1 (BNC-BNC).
Again, perhaps a reason why I can't tell the difference between cables/sources on my setup Smile

Anyway, glad you've got it straight in your head now - that's what matters Smile

I have the family here now and they're browsing my music collection via a remote keyboard controlling my htpc - it's really cool that people are listening to music again here instead of just switching the TV on when they come through the door!
Posted on: 10 May 2010 by DarrellK
quote:
Originally posted by Andy S:
quote:
Originally posted by DarrellK:
Hi Andy,

A question: what would happen if a source was so completely broken that it sent random noise down the S/PDIF intermingled with the digital signal? (Maybe this is outside the scope of this discussion, as it would not be a "bit perfect" source?)
Try injecting AC-3 into the DAC. It plays it but it is all buzzes (I've tried it for a few seconds!).


Sorry for the belated reply, but having read through the last few days of this thread, I have an idea or two - I'm one of those people who is by no means an expert on this stuff, but I'll stick my head above the parapet anyway, although I'm not even sure that I am posing the right questions, never mind coming up with any answers...

It seems clear that if an optical disk transport makes mistakes in recovering the data from the disk, all bets are off - we are not dealing with a bit-perfect source any more.

But what about streamed data? Obviously, there is the minefield of unintentional sample rate conversion in the process between data retrieval and its presentation to the S/PDIF interface. But assuming one has the correct hardware, and has ones software selected/configured so as not to do this, might there be mechanisms through which the streaming device gets it wrong, and presents an incorrect bitstream to the S/PDIF? I think that the answer to this must be yes, in theory, although I have no idea how common it might be, or how it might happen.

Perhaps some kind of noise might be added into the data stream by the source device, such that it is subtly altered, rather than destroyed, before it is presented to the S/PDIF interface? Is it possible for this sort of thing to happen in the digital, rather that analogue, domain? Or could some samples be lost and replaced with interpolated data somewhere in the data chain within the source device?
Posted on: 10 May 2010 by pcstockton
How is the DAC immune to any effect of a transport when I can now hear when I minimize my the media player. A little click happens.

With ASIO and Foobar's bit perfect output, and Naim's reclocking, I should hear that right?

Well it is clearly audible and clearly NOT how a CDP or USB input sound. If fact I can now definitively state that different sources will sound different period.

Even Andy couldn't disagree with this one.

-p
Posted on: 11 May 2010 by Andy S
quote:
Originally posted by pcstockton:
How is the DAC immune to any effect of a transport when I can now hear when I minimize my the media player. A little click happens.

With ASIO and Foobar's bit perfect output, and Naim's reclocking, I should hear that right?
No, you shouldn't if the output is bit-perfect. I know I hear a click when I start tracks on my player, but this is because the default output frequency of the ASLA system (I'm using linux) is 48kHz and when no track is playing, the system swaps between 44.1 and 48 then back again when the next track is cueued. This causes the DAC to momentarily lose sync and a small click to be heard.
I'm not saying that this is what's happening in your system, but there are other system level issues at work here too. I've also found bitstreams encoded in APE don't work too well in my system - sometimes losing sync mid track causing a whiteout. I know it's the APE format doing this as I have perhaps 300 CDs encoded in FLAC, and 2 in APE. The APE files drop out (randomly) whereas no FLAC ones do and the same APE files recoded to FLAC don't either. There are system interactions to worry about, and once you have an OS, you have potential for momentary screwups somewhere.

quote:
Well it is clearly audible and clearly NOT how a CDP or USB input sound. If fact I can now definitively state that different sources will sound different period.
Well, I'd prefer to say that there are sometimes your system doesn't output bit perfect audio (when you minimize foobar by the sound of it) rather than different sources sounding different period. What appears to be happening is you get bit perfect playback 99.9% of the time, and not bit perfect 0.1% of the time - perhaps due to a small latency issue in filling buffers in the low-latency ASIO output as the system context switches.

quote:
Even Andy couldn't disagree with this one.

-p
Couldn't he Confused Winker Sounds Roll Eyes like a system level issue you are experiencing....
Posted on: 11 May 2010 by Andy S
quote:
Originally posted by DarrellK:
It seems clear that if an optical disk transport makes mistakes in recovering the data from the disk, all bets are off - we are not dealing with a bit-perfect source any more.
Correct!

quote:
But what about streamed data? Obviously, there is the minefield of unintentional sample rate conversion in the process between data retrieval and its presentation to the S/PDIF interface. But assuming one has the correct hardware, and has ones software selected/configured so as not to do this, might there be mechanisms through which the streaming device gets it wrong, and presents an incorrect bitstream to the S/PDIF? I think that the answer to this must be yes, in theory, although I have no idea how common it might be, or how it might happen.


OK - let's examine this further. If this does happen, lets assume the framing data for SPDIF (which will be generated by the interface) is always correct. This means the DAC will sync to the incoming bitstream and always decode it. If the data is mangled in some way by the interface/software, this must be a random process meaning that you will get data loss at any point in the 16 bit word that makes up the individual sample value. For us to have a different sound from the DAC (i.e. less sense of space, less clearly defined sound, less tingle factor which can normally be attributed to jitter effects) the changes must be occurring pretty consistently (if they happened every "now and again" they'd be more likely to show up as pops or clicks), and we must be losing bits at the least significant end of the 16 bit word. That isn't a random process - it's more one where the timing isn't correctly processed. I've heard this effect on copied CDs against originals on a CD player - it's a "not quite right" feeling where things aren't as well defined or steady and I've always put this down to more jitter on the copied CD than the original.

quote:
Perhaps some kind of noise might be added into the data stream by the source device, such that it is subtly altered, rather than destroyed, before it is presented to the S/PDIF interface? Is it possible for this sort of thing to happen in the digital, rather that analogue, domain? Or could some samples be lost and replaced with interpolated data somewhere in the data chain within the source device?


The only way I know this subtle alteration can happen (as opposed to some systematic algorithm designed to do it in digital signal processing) in the digital domain is by adding jitter as you keep the samples the same but just move them around in time. Now, given the DAC claims to remove jitter, that shouldn't be the case. Just another quote if I may from the white paper (in fact, the last FAQ element on the last page):

quote:
Naim DAC white paper:
What is jitter exactly?
Jitter is variations in the time separation of digital
audio samples. All S/PDIF induced jitter coming in to
the Naim DAC is removed
Posted on: 11 May 2010 by rich46
quote:
Originally posted by AllenB:
Too much theory on this thread Roll Eyes


i wonder how many on this thread have the dac , ive had it more time than most 12/12/09.

simple it is agreat hub.
Posted on: 11 May 2010 by gav111n
Hi Andy,

I have found this an interesting thread. It’s made me think more deeply about the process of getting my digitally stored music converted into sound in my room, which for me has been a good thing.

I am not an electronics engineer but I like a good puzzle, so let me throw my thoughts into the discussion.

The nDAC white paper says:

'Moreover, S/PDIF circuitry represents a radio frequency (RF) noise source and its presence in a CD player is audible.'

It also says:

'How to overcome the problem of S/PDIF noise entering a DAC
The Naim DAC’s high-speed DSP (digital signal processor) front-end is electrically isolated from its high-resolution DAC and analogue circuits. Also, the two sections are run from separate power supplies. Together these measures significantly reduce the digital RF noise which could affect the analogue stage.'

This seems to suggest that simply having a S/PDIF signal in the nDAC has the potential to affect sound quality. The nDAC arrangement significantly reduces the effect but by implication does not eliminate it.

Even though bit perfect data may be transmitted by two (say) CDPs, the S/PDIF signal will surely be different from both i.e. if you plot the S/PDIF voltage signal versus time from the two CDPs having played the same chunk of music and overlay them. They will have different amounts of jitter but also presumably the actual shape of the waveform will be different even though they are both bit perfect. This would mean that the RF noise characteristic within the DAC is different with these two CDPs? Both are bit perfect, the jitter is removed from both by the nDAC’s clever system but the RF noise experienced by the nDAC is different.

Gavin.
Posted on: 11 May 2010 by Andy S
quote:
Originally posted by gav111n:
Hi Andy,

I have found this an interesting thread. It’s made me think more deeply about the process of getting my digitally stored music converted into sound in my room, which for me has been a good thing.
Hi Gavin,

Yes, it's made me question how to do all this too which is a good thing - I like to understand what is going on in the system too.

quote:
...
'How to overcome the problem of S/PDIF noise entering a DAC
The Naim DAC’s high-speed DSP (digital signal processor) front-end is electrically isolated from its high-resolution DAC and analogue circuits. Also, the two sections are run from separate power supplies. Together these measures significantly reduce the digital RF noise which could affect the analogue stage.'

This seems to suggest that simply having a S/PDIF signal in the nDAC has the potential to affect sound quality. The nDAC arrangement significantly reduces the effect but by implication does not eliminate it.

Even though bit perfect data may be transmitted by two (say) CDPs, the S/PDIF signal will surely be different from both i.e. if you plot the S/PDIF voltage signal versus time from the two CDPs having played the same chunk of music and overlay them. They will have different amounts of jitter but also presumably the actual shape of the waveform will be different even though they are both bit perfect. This would mean that the RF noise characteristic within the DAC is different with these two CDPs? Both are bit perfect, the jitter is removed from both by the nDAC’s clever system but the RF noise experienced by the nDAC is different.

Gavin.
Hmm... That's an interesting question. RFI normally gets transmitted over the power supply lines - the switching of all bits in the digital system at the same time loads the power lines more heavily at the switch point and thus causes spikes (which translate into RF interference) at the analogue section. Perhaps different jitter will load things differently, but this has to be a second order effect and therefore much lower impact compared to the jitter making it into the DAC clock circuitry.

If you take a look at pictures of the innards of the nDAC, you can see the optoisolators bridging into the DAC and you can quite clearly see the ground plane break between the two sections of circuits. You can also see though that the DSP and ADC sections are fed by different windings from the transformer (3 sets of windings - presumably 2 for the DAC, one for the digital circuits). I.e. Naim have done what they can (apart from encompassing the analogue sections in a metal can which is done for sensitive RF tuners such as satellite tuners) at reducing the RF interactions.

Having said that, if RF is being passed back up through the power supply chain, it might induce something else through the transformer (although you have the smoothing caps to reject any noise back again. Additionally, if this were a method of reducing the performance, you'd expect that the effect would be reduced even further once you put an external power supply on the DAC.

It is a possible reason for a difference in sound but it doesn't (IMHO) account for the large differences people are claiming - particularly when you are dealing with the effects of mains electricity and the associated noise on that (from fridges etc.. when I lived in the countryside, every morning my 135 transformers would buzz - presumably due to someone switching on a hairdryer in a nearby house - it's never happened in any other house other than that one!).
Posted on: 11 May 2010 by Andy S
quote:
Originally posted by gav111n:
The nDAC white paper says:

'Moreover, S/PDIF circuitry represents a radio frequency (RF) noise source and its presence in a CD player is audible.'
You know... every time I see this quote I wonder how much of it is folklore.

I know Naim did a lot of work with the original CDS1 - I remember seeing some posts by JV on the old forum whereby they'd modded one to get digital out and as I remember (although my brain could be fading) that this made a difference to sound out. Is this sound change still true or is this a hangover from the early 90's when technology and Naims understanding of these things were different.

Not saying that it doesn't make a difference in any circumstance, just wondering if it is still valid in todays designs...
Posted on: 11 May 2010 by pcstockton
quote:
Originally posted by Andy S:
lets assume the framing data for SPDIF (which will be generated by the interface) is always correct.


Andy,

This is a HUUUUUGE assumption. Why would it always be correct? How could it EVER be correct if ASIO/Foobar isn't bit perfect (according to you.... I have NEVER once seen the sync light go off in 100 hours of listening).

Every time someone brings up a completely valid reason you spin it around and say "well they should be using toslink", or "it must not be bit perfect."

Have you taken one second to think that maybe things arent always bit perfect (especially through CDPs), and people are connecting mostly through BNC and coax?

Have you considered for a moment that the "Chassis/Floating" switch does not cure EVERY single kind of ground issue ESPECIALLY when using coax or BNC?


You are starting to develop a nice long list of exceptions to your dogma. A list that is quickly and easily punching holes in your theory.

Yes you have clearly shown that in a vacuum all sources will sound the same due to an elimination of timing issues. But microphony, RF, ground loops, lack of bit perfection etc, all can play a part in degrading sound.

Please dont explain this away as avoidable with switches and cables. It is really happening in this real world.

My Foobar sounds great and I believe it to be bit perfect, not only with ASIO/Transit, but also ASIO or Wasapi on the Juli@. Yet I can hear a difference in sound when doing things like minimizing the Foobar window. I was told the graphics card is injecting a little noise into the mix.

While it does not change the overall SQ of the source, it does definitively point to ways in which sources could sound different given a handful of other things than jitter.

The Naim DAC is great, and for me is basically source independent. But there is no way it is a magical piece of kit that CANNOT possibly sound different given other systems and methods.

-patrick
Posted on: 11 May 2010 by pcstockton
quote:
Originally posted by Andy S:


It is a possible reason for a difference in sound but it doesn't (IMHO) account for the large differences people are claiming -


It is good to hear you are revising your position. And NO ONE ever said the differences were "large". Everyone who likes it, like it through all inputs, and with all sources. Just one or two of them sound a little better than others.