You still not getting my base question:
How could a coupling capacitor in the digtal domain effect the final audio sound quality.
I realize the cap does not care if the signal is digital or analog. What I am saying is that given a particular set of bits, wether they are pristine with razor sharp transients, or a little bit more rounded on the edges, they are still the same bits. They are interpreted the same way by the digital switches. Thus they should result in the exact same final analog audio output.
A slightly distorted 1 from the analog perspective is still a 1 from the digital perspective. Unless somehow some bits sound better than others.
I am not trying to argue. I am trying to understand. And you keep answering a question that I am not asking. None of those links answer my question either.
How could a coupling capacitor in the digtal domain effect the final audio sound quality.
I realize the cap does not care if the signal is digital or analog. What I am saying is that given a particular set of bits, wether they are pristine with razor sharp transients, or a little bit more rounded on the edges, they are still the same bits. They are interpreted the same way by the digital switches. Thus they should result in the exact same final analog audio output.
A slightly distorted 1 from the analog perspective is still a 1 from the digital perspective. Unless somehow some bits sound better than others.
I am not trying to argue. I am trying to understand. And you keep answering a question that I am not asking. None of those links answer my question either.
Okay. This is going to a bad place.
I appologize if I am coming off as advisarial. I really do not intend to. I think we are just having a lot of difficulty getting on the same page.
Please allow me to try once more. Here is another way to look at it which I hope will make my question more clear:
Yes, digital is analog, but it is not audio. It is only a representation of the audio. Even with various distortions of the analog qualities of the bit stream, that representation can remain unchanged. So long as the digital values of the bits can be clearly determined by the schmitt trigger input buffer of the spdif receiver and they are what they should be.
What I am trying to understand is how a coupling capacitor, which should have relatively small impact to the analog qualities of the bit stream, could somehow change the actual digital representation of the audio to become subjectively bright or mellow.
I would think if the analog qualities of the bit stream were of sufficiently poor quality, the digital representation would be corrupted in a fairly random sort of way.
I appologize if I am coming off as advisarial. I really do not intend to. I think we are just having a lot of difficulty getting on the same page.
Please allow me to try once more. Here is another way to look at it which I hope will make my question more clear:
Yes, digital is analog, but it is not audio. It is only a representation of the audio. Even with various distortions of the analog qualities of the bit stream, that representation can remain unchanged. So long as the digital values of the bits can be clearly determined by the schmitt trigger input buffer of the spdif receiver and they are what they should be.
What I am trying to understand is how a coupling capacitor, which should have relatively small impact to the analog qualities of the bit stream, could somehow change the actual digital representation of the audio to become subjectively bright or mellow.
I would think if the analog qualities of the bit stream were of sufficiently poor quality, the digital representation would be corrupted in a fairly random sort of way.
Do you really understand jitter??
Jitter has a spectrum, and there is no predictor of what will affect the jitter spectrum, and how it will correlate to perceived sonic qualities. All that I can say is that if you have jitter, it won't sound right. Period.
I think that a lot of tweaks that you guys do.........futzing with op-amps and coupling caps and the like in your D/A boxes, is because there is too much jitter, and you are using those as band-aids to fix a problem that has its roots elsewhere.
Typical SPDIF D/A thingie has around 1 nSec of jitter. It needs to be down around 10 pSec for stuff to sound right.
And almost every D/A box that I have looked at has a rotten SPDIF interface, with excessiive reflections. Making a bad situation worse.
Jocko
Jitter has a spectrum, and there is no predictor of what will affect the jitter spectrum, and how it will correlate to perceived sonic qualities. All that I can say is that if you have jitter, it won't sound right. Period.
I think that a lot of tweaks that you guys do.........futzing with op-amps and coupling caps and the like in your D/A boxes, is because there is too much jitter, and you are using those as band-aids to fix a problem that has its roots elsewhere.
Typical SPDIF D/A thingie has around 1 nSec of jitter. It needs to be down around 10 pSec for stuff to sound right.
And almost every D/A box that I have looked at has a rotten SPDIF interface, with excessiive reflections. Making a bad situation worse.
Jocko
Camron:
Send me an E-mail, and I will tell you a
few things that I have learned relating
to your quest to understand the SPDIF/
component issue.
fastcat95@juno.com
Send me an E-mail, and I will tell you a
few things that I have learned relating
to your quest to understand the SPDIF/
component issue.
fastcat95@juno.com
Jocko: In a previous message I asked you if it all boiled down to jitter. So basically, your saying, yes. But that jitter is in fact a rather compilated phenomenon. Okay. I can accept that.
From what I gather, I think there are very few individuals who really understand jitter. There are probably only like 3 or 4 in this whole forum. Like you and Werewolf. That dude is scarry.
I can see how the various non-linear imperfections of a coupling capacitor could have a extremely difficult to predict impact on the jitter spectrum. But once you knew the jitter spectrum, would you not be able to at least model how that would impact the audio? My crude understanding is that jitter ultimately becomes noise. Though I have no idea what kind, random, correlated, whatever. And that basically it decreases signal to noise ratio with increasing frequency. I can see how an interesting jitter spectrum could cause that SNR vs F function to be clearly non-linear. I still think the impact resulting in the equivalent of a tone control as being a bit of a stretch.
Again, what about a DAC with an asynchronous resampler? Given the amount of jitter rejection those devices have, would it not make any and all sources sound the same? Even if the local oscillator was of poor quality, since everything is being resampled by it, it would make every source sound equally bad. What do you think?
fastcat95: I think I will. Thank you.
From what I gather, I think there are very few individuals who really understand jitter. There are probably only like 3 or 4 in this whole forum. Like you and Werewolf. That dude is scarry.
I can see how the various non-linear imperfections of a coupling capacitor could have a extremely difficult to predict impact on the jitter spectrum. But once you knew the jitter spectrum, would you not be able to at least model how that would impact the audio? My crude understanding is that jitter ultimately becomes noise. Though I have no idea what kind, random, correlated, whatever. And that basically it decreases signal to noise ratio with increasing frequency. I can see how an interesting jitter spectrum could cause that SNR vs F function to be clearly non-linear. I still think the impact resulting in the equivalent of a tone control as being a bit of a stretch.
Again, what about a DAC with an asynchronous resampler? Given the amount of jitter rejection those devices have, would it not make any and all sources sound the same? Even if the local oscillator was of poor quality, since everything is being resampled by it, it would make every source sound equally bad. What do you think?
fastcat95: I think I will. Thank you.
Noise from SPDIF
I do not believe the spdif ground is coupled to ground, if it is it is AC coupled, I will check this out. I will try a ferrite bead or 2 as well.
The transformer is terminated with a 75 ohm resistor.
I will have a look at all this over the weekend and let you all know.
Thanks
Guillaume
I do not believe the spdif ground is coupled to ground, if it is it is AC coupled, I will check this out. I will try a ferrite bead or 2 as well.
The transformer is terminated with a 75 ohm resistor.
I will have a look at all this over the weekend and let you all know.
Thanks
Guillaume
Jocko Homo said:
So most of the problems are related to jittery signal coming from the transport? Most of these tweaks end up in the 'it sounds great but it's too bright' area anyway.
fastcat95 said:
Can't you post it here? Please 🙂
Cameron said:
The DA box I'm using uses the AD1892 receiver set to 1:1 resampling, and the clock is verified to be very quiet with a 'quality' oscillator and a PLL1705 to derive the MCLK; the PSs use standard 7xxx regs and TL431 shunt regs, with separate clock, digital & analog supplies.
All of these would be a great way to reduce incoming noise/jitter, in theory. Not so very much in practice 🙁 It sounds royally great though, but different transports sound, err, different.
I have also looked deep into the circuit solutions and theory on spdif interfaces and come to the same conclusion as Jocko. It seems that even most of the regarded DIY DAC projects published don't have anything other than the datasheet solution in the spdif input.
I'm getting more and more wound up to really try to experiment and make something more ....
Only problem is that there are guite a bit of different soluton that might work. At the moment I'm trying to determine where to start 🙂
Ergo
I'm getting more and more wound up to really try to experiment and make something more ....
Only problem is that there are guite a bit of different soluton that might work. At the moment I'm trying to determine where to start 🙂
Ergo
Lucpes: Thanks for the reply. Hum. That is interesting. When you say 1:1 you mean roughly, not exactly, right? Your solution sounds like it should supress jitter quite nicely. So the question becomes, what is different between theory and reality? Perhaps a higher quality ASRC like the AD1896 would work better, though you would then need a seperate DIR. My only guess is that enough jitter is still getting through to effect final audio quality. If I interpreted Werewolf's big posts on ASRCs correctly, any jitter not rejected is converted to noise. But instead of it happening at the DAC as usual, it happens at the ASRC and that noise is encoded into the outgoing bits. I can not think of another explanation why any digital source should sound different than any other.
I think I will send some email to Werewolf. Being that he actually designs ASRCs for a living, I am sure he would have some insight into this topic.
I think I will send some email to Werewolf. Being that he actually designs ASRCs for a living, I am sure he would have some insight into this topic.
Cameron said:
I was told that it does 1:1 resampling by the guy who designed it; I'll pass answering 'too' technical questions to avoid saying any stupid things
Cameron, you have to remember that while the folks here don’t know diddlysquat about digital circuits, they know even less about bi-phase. That’s why they are obsessed with jitter and worry about the eye patterns and shape of the S/PDIF transitions. You will notice that the ones harping the most about jitter are the clock mongers because they have something to sell.
I think the contrasts are striking. On one hand, none of the highly regarded Audio Note DACs do anything special to mitigate jitter. They use the data and clocks directly from the CS8412 with no special oscillators, PLLs, or reclocking circuits. They don’t even remove the stagger between the left and right channels. I think the highly regarded Zanden uses a similar, no-frills approach. On the other hand, the SOP here is to use a fancy oscillator in the CDP to reduce jitter at the source. Then, add a discreet RS-422 receiver and fancy capacitors in front of the RS-422 receiver integrated in the CS8412 to remove jitter at the destination. Then, reclock the data and clocks to remove jitter before the DAC.
You will notice the self-proclaimed “experts” here obsess about jitter everywhere except where it really matters - the analog output of the DAC. Anyone truly concerned about jitter should just add a fast S/H on the analog output of the DAC and forget about everything else.
I browse this forum for amusement: There is no useful information here. If you explore the archives you will find some real gems. My favorite is:
To Quote a Quote
Hi, To quote a quote:
The funny thing is that the comparator works best without any coupling capacitors between the the AD8561 and the CS8412. I got got this hint from Fred Dieckmann & Jocko Homo. So now you know who the experts are........
🙄
Hi, To quote a quote:
The funny thing is that the comparator works best without any coupling capacitors between the the AD8561 and the CS8412. I got got this hint from Fred Dieckmann & Jocko Homo. So now you know who the experts are........
🙄
Is it really about the Jitter?
Julian Dunn has an interesting statement about the audibility of jitter. It applies to the sampling jitter but I guess the same can be applied to the interface jitter. Here is the quotation (AP Newsletter, Vol. 15, # 1, the same part can be found here, page 2):
”It is one thing to be able to identify and measure sampling jitter. But how can we tell if there is too much?
A recent paper by Eric Benjamin and Benjamin Gannon describes practical research that found the lowest jitter level at which the jitter made a noticeable difference was about 10 ns rms. This was with a high level test sine tone at 17 kHz. With music, none of the subjects found jitter below 20 ns rms to be audible.
This author has developed a model for jitter audibility based on worst case audio single tone signals including the effects of masking. This concluded:
“Masking theory suggests that the maximum amount of jitter that will not produce an audible effect is dependent on the jitter spectrum. At low frequencies this level is greater than 100 ns, with a sharp cut-off above 100 Hz to a lower limit of approximately 1 ns (peak) at 500 Hz falling above this frequency at 6 dB per octave to approximately 10 ps (peak) at 24 kHz for systems where the audio signal is 120 dB above the threshold of hearing.”
In the view of the more recent research, this may be considered to be overcautious. However, the consideration that sampling jitter below 100 Hz will probably be less audible by a factor of more than 40 dB when compared with jitter above 500 Hz is useful when determining the likely relative significance of low- and high-frequency sampling jitter.”
Now, following this source, a colossal amount of low frequency jitter is inaudible. High freq jitter is more audible but when it comes to the interface, receiver’s PLL filter can be designed (just different values) to attenuate it (raising intrinsic low freq jitter but it is, as per reference, no problem).
Comments?
Pedja
Julian Dunn has an interesting statement about the audibility of jitter. It applies to the sampling jitter but I guess the same can be applied to the interface jitter. Here is the quotation (AP Newsletter, Vol. 15, # 1, the same part can be found here, page 2):
”It is one thing to be able to identify and measure sampling jitter. But how can we tell if there is too much?
A recent paper by Eric Benjamin and Benjamin Gannon describes practical research that found the lowest jitter level at which the jitter made a noticeable difference was about 10 ns rms. This was with a high level test sine tone at 17 kHz. With music, none of the subjects found jitter below 20 ns rms to be audible.
This author has developed a model for jitter audibility based on worst case audio single tone signals including the effects of masking. This concluded:
“Masking theory suggests that the maximum amount of jitter that will not produce an audible effect is dependent on the jitter spectrum. At low frequencies this level is greater than 100 ns, with a sharp cut-off above 100 Hz to a lower limit of approximately 1 ns (peak) at 500 Hz falling above this frequency at 6 dB per octave to approximately 10 ps (peak) at 24 kHz for systems where the audio signal is 120 dB above the threshold of hearing.”
In the view of the more recent research, this may be considered to be overcautious. However, the consideration that sampling jitter below 100 Hz will probably be less audible by a factor of more than 40 dB when compared with jitter above 500 Hz is useful when determining the likely relative significance of low- and high-frequency sampling jitter.”
Now, following this source, a colossal amount of low frequency jitter is inaudible. High freq jitter is more audible but when it comes to the interface, receiver’s PLL filter can be designed (just different values) to attenuate it (raising intrinsic low freq jitter but it is, as per reference, no problem).
Comments?
Pedja
Well, I'm glad that we have a "real" self-proclaimed expert here.
None of us has ever said that you don't need to have clean clock at the DAC. You solution is above the capabilities of the average Joe here. We are trying to get them to implent things that are easy to grasp, and make it sound better, without $$$ and stuff that they don't know or have access to.
And just who is pushing clocks???? No one has mentioned clocks in this thread.
If you have a problem with that, then tough.
Jocko
None of us has ever said that you don't need to have clean clock at the DAC. You solution is above the capabilities of the average Joe here. We are trying to get them to implent things that are easy to grasp, and make it sound better, without $$$ and stuff that they don't know or have access to.
And just who is pushing clocks???? No one has mentioned clocks in this thread.
If you have a problem with that, then tough.
Jocko
Pedja,
I'm not so sure that this is a correct conclusion. Jitter in
the clock, more correctly - "phase noise", can be viewed
as a signal that intermodulates with the desired signal.
Thus it doesn't necessarily follow that a greater amount
of LF phase noise is less audible.
I'm not so sure that this is a correct conclusion. Jitter in
the clock, more correctly - "phase noise", can be viewed
as a signal that intermodulates with the desired signal.
Thus it doesn't necessarily follow that a greater amount
of LF phase noise is less audible.
Now that’s funny. I’m a software guy, not an EE – I don’t even have a college degree. I’ve been involved with music synthesis, digital audio, and high-speed computers since 1974 and during that time I’ve picked up a little knowledge about digital circuits. So, if I can see gross errors and misinformation presented here, it must be really bad. How in the world can someone with even a modicum of knowledge and experience confuse a CS8412 with a 74HCU04?
What’s so hard about making a S/H? It’s certainly no harder than some of the other cockamamie ideas floating around here ostensibly designed to reduce jitter. Do you really think a few less PICOseconds of clock jitter in the CDP amounts to a hill of beans when the intrinsic jitter of the DAC’s analog output is measured in MICROseconds?
jbokelman said:
which is why all AN DACs lack resolution and precision
I won't tell you how many people sold their AN DAC3 after they built our DAC and I won''t tell you how many XO DAC upgrades I sold to AN DAC owners either.
So, bad example.
Jitter is key and that has been acknowledged ever since the "invention" of digital audio - I am talking 70 years ago here.
cheers
Strange things happen when you post at 5'o clock in the morning after a long working day... You realized I corrected that mistake later on, did you?
/OliverD - a guy who sometimes confuses things - he is not ashamed to admit
Yes, Oliver.....
We remember. Some folks apparently never heard of a typo. We have all forgiven you long ago.
Now........will someone explain how DAC, operating at 8X oversampling.....which comes out to a period of around 2.8 uSec at 44.1 kHz, can have MICROCSECONDS of jitter?
Well, Signoro Software.......some of us are EEs. And have spent a great deal of our professional career measuring jitter on digital transmissions.
The effects of jitter are were documented. The detrimental affects of refections on a SPDIF signal can be demonstrated. I have done so many times. Whether you care to accept that or not, I really don't care. On many systems, some of them as bad as what you seem to regard as being good.
So we respond to your posts as amusement for us.
You may be a software guru, but your knowledge of bi-phase data, as it applies here, is lacking. Maybe you should invest in a spectrum analyser and a TDR. Then you can refute what I say.
Until then, I will maintain that you appear to be as confused as a barking bird.
Jocko
We remember. Some folks apparently never heard of a typo. We have all forgiven you long ago.
Now........will someone explain how DAC, operating at 8X oversampling.....which comes out to a period of around 2.8 uSec at 44.1 kHz, can have MICROCSECONDS of jitter?
Well, Signoro Software.......some of us are EEs. And have spent a great deal of our professional career measuring jitter on digital transmissions.
The effects of jitter are were documented. The detrimental affects of refections on a SPDIF signal can be demonstrated. I have done so many times. Whether you care to accept that or not, I really don't care. On many systems, some of them as bad as what you seem to regard as being good.
So we respond to your posts as amusement for us.
You may be a software guru, but your knowledge of bi-phase data, as it applies here, is lacking. Maybe you should invest in a spectrum analyser and a TDR. Then you can refute what I say.
Until then, I will maintain that you appear to be as confused as a barking bird.
Jocko
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