| lucpes |
| I have a DAC that uses two 10nf caps for SPDIF coupling. What should I use for those - I tried so far WIMA FKP2 - too dark sounding and WIMA MKS02, which seems to be too harsh and 10nf ceramics which are 'dirty sounding'. Guess MKP or MKTs (low dielectric loss caps/good HF & pulse behaviour) would fit the bill better? |
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| Jocko Homo |
Well, part of it depends where they are in the circuit.
If they connect directly to the input, a lot of the sonic characteristics will be because of reflections.
I would expect ceramics to not sound good, although their lower inductance would be a benefit.
Jocko |
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| roibm |
I had about the same and I just removed them. Since the source is a good device, I don't need them anymore.
Do you really need them? As far as I know the perfect coupling caps are no caps ;) |
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| lucpes |
Here's the schematic right out of the datasheet.
@roibm: are you sure they are safe to remove? The transport device is transformer coupled. |
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| lucpes |
| Forgot to attach... |
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| MalichiConstant |
| Use a good metalized polypropelyne cap (MKP) with the lowest voltage you can get to keep the cap physical small. You can go to higher values and 0.047uF would be fine. I use some of the ERO 100V MKP caps to AC couple 75 load like the input of pulse transformers. Always put a cap in front of a pulse transform. I have heard a few 100 uA screw the sound of the pulse tranformer up big time when driving it with a video amp with a DC offset in the low millivolt range. NO DC CURRENT THROUGH THE PULSE TRANSFORMER! One of the Black Gate 0.1 uF nonpolar electrolytics also sounded extremly good in the digital interface. |
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| Jocko Homo |
If you are only going to use a transformer on one end, make it the RX end.
Kepping the outer conductor on th TX will reduce radiation, although there are ways around that. Besides, the RX side will perform better if you use a differential input, and the best way to do that is with a transformer.
Jocko |
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| jewilson |
There nothing wrong with using a CKO5 style NPO, COG or even X7R ceramic material capicator in this application. There are many more areas of TX and RX interface to cause problems that those caps. However, make you use low inductance caps.
Jocko, he can elaborate on all the pit falls in the SPDIF interface. You might even get him to sell you a an impedance matched design the has been spectral tested. Then you won't have to get concerned about the caps and poor designs that plague 99.9% of all SPDIF interfaces.
My sermon is over for at least week, till some one asks the same questions again.
;) |
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| roibm |
| quote: | Originally posted by lucpes
@roibm: are you sure they are safe to remove? The transport device is transformer coupled. |
Well, in my case, I got no transport, it comes from the computer and I trust it. If I'll get problems, then I'll learn a lesson too ;) |
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| lucpes |
| quote: | Originally posted by Jocko Homo
If you are only going to use a transformer on one end, make it the RX end.
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Thanks, but I'm sure to mess up things if I do that, as I need to do more reading on this subject :rolleyes:
| quote: | Originally posted by jewilson
There nothing wrong with using a CKO5 style NPO, COG or even X7R ceramic material capicator in this application. There are many more areas of TX and RX interface to cause problems that those caps. However, make you use low inductance caps.
Jocko, he can elaborate on all the pit falls in the SPDIF interface. You might even get him to sell you a an impedance matched design the has been spectral tested. Then you won't have to get concerned about the caps and poor designs that plague 99.9% of all SPDIF interfaces. ;) |
Ceramic caps caused wreck havoc on the SPDIF signal (sonicly) at least in my case so I won't try it again. I know there are very knowledgeable people on this forum, that's why I posted here, and sorry for opening the same can of worms again :angel:
| quote: | Originally posted by roibm
Well, in my case, I got no transport, it comes from the computer and I trust it. If I'll get problems, then I'll learn a lesson too ;) |
What soundcard are you using? I've seen some cards (M-Audio) which are transformer coupled & have a 10nf ceramic SMD in series with the SPDIF out which has to go. |
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| roibm |
| quote: | Originally posted by lucpes
What soundcard are you using? I've seen some cards (M-Audio) which are transformer coupled & have a 10nf ceramic SMD in series with the SPDIF out which has to go. |
M-Audio Audiophile 2496.
Didn't check the spdif out path. I'll do it next time I open the machine. |
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| jewilson |
| quote: | | Ceramic caps caused wreck havoc on the SPDIF signal (sonicly) at least in my case so I won't try it again. I know there are very knowledgeable people on this forum, that's why I posted here, and sorry for opening the same can of worms again |
The fact are that the ceramic caps that I mentioned or made for high frequency signals, it obvious by you statement that you don’t a have clue. The majority of high frequency oscillator phase lock loops use either Ceramic or Mica capacitor. There is a big difference in running a low frequency analog signal through a ceramic vrs a good PP PS film cap, but Bud we are not talking about analog signal are we.
Sorry I waisted your time, I did not realize by your statements that your were an ex-pert! :rolleyes: |
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| BrianL |
_IF_ 10nF is indeed "the" value, I'd use 10nF C0G SMT caps.
You can get 10nF/50V/C0G in a 1210 SMT package, which gives
you about as ideal a cap (good DF, DA, minimal parasitics) as
you can get for this sort of application. |
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| lucpes |
| quote: | Originally posted by jewilson
The fact are that the ceramic caps that I mentioned or made for high frequency signals, it obvious by you statement that you don’t a have clue. The majority of high frequency oscillator phase lock loops use either Ceramic or Mica capacitor. There is a big difference in running a low frequency analog signal through a ceramic vrs a good PP PS film cap, but Bud we are not talking about analog signal are we.
Sorry I waisted your time, I did not realize by your statements that your were an ex-pert! :rolleyes: |
Correct, I have almost no scientific knowledge of what I'm talking about and having to rely solely on ears to discern if a certain component is good for a specific application can be sometimes deceiving :) But I have no chance to get CKO5 ceramic caps in my neck of the woods and silvered Mica would be a tad bit larger than the 5mm pitch I need for 10nf so it seems that I'm stuck to low inductance/low dielectric loss/good hf behaviour film caps instead or no cap at all.
Maybe this would be good for my application if I bend the leads a bit?
http://uk.farnell.com/jsp/endecaSea...1&Ntk=gensearch |
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| jewilson |
lucpes
Sure, two of these in parallel would work fine. I don't believe that being 600pf off is a big deal however you could get a 500pf to go with it.
This Farnell in One this company has the same look to their web site as US Newark Electronics http://www.newark.com they must be related. I see that Farnell sales NPO ceramic's. but the Mica is a little better. |
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| emuman100 |
roibm,
You must have hearing more sensitive than a dog or cat. I highly doubt simple tiny capacitors will make any difference in audio quality of an S/PDIF signal. I have an i2s to TTL to optical to RS485 to coax and into jwb's great DAC. It has a SC982-04 pulse transformer and a 0.1uF SMT cap on the input to block DC and couple the signal into the pulse transformer. The sound is excellent. It will sound the same if the cap was or wasn't there, if the pulse transformer was or wasn't there, it will make no difference. I advise you keep the capacitors and use a pulse transformer. That will keep everything safe. Use 10Mbps RS422/485 tranceiver IC's like the MAX3443 from Maxim, SP1485 from Sipex, SN75176 from TI, or the DS3695 from National Semi for your S/PDIF transmission and input the TTL signal right into the S/PDIF receiver. You will be satisfied. My RS485 to coax circuit uses a CMOS inverter, some resistors and a 1:1 pulse transformer. The output of the circuit gets connected to my DAC. I have transformer coupling on both sides, just for the hell of it. Even though the signal is going through all these conversions, it still sounds perfect. |
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| roibm |
| quote: | Originally posted by emuman100
roibm,
You must have hearing more sensitive than a dog or cat. I highly doubt simple tiny capacitors will make any difference in audio quality of an S/PDIF signal. I have an i2s to TTL to optical to RS485 to coax and into jwb's great DAC. It has a SC982-04 pulse transformer and a 0.1uF SMT cap on the input to block DC and couple the signal into the pulse transformer. The sound is excellent. It will sound the same if the cap was or wasn't there, if the pulse transformer was or wasn't there, it will make no difference. I advise you keep the capacitors and use a pulse transformer. That will keep everything safe. Use 10Mbps RS422/485 tranceiver IC's like the MAX3443 from Maxim, SP1485 from Sipex, SN75176 from TI, or the DS3695 from National Semi for your S/PDIF transmission and input the TTL signal right into the S/PDIF receiver. You will be satisfied. My RS485 to coax circuit uses a CMOS inverter, some resistors and a 1:1 pulse transformer. The output of the circuit gets connected to my DAC. I have transformer coupling on both sides, just for the hell of it. Even though the signal is going through all these conversions, it still sounds perfect. |
I never said I heard a difference when I removed the cap. But then again, there was a pause of several days between removing the cap and serious listening again.
The cap was a really bad 0.1uF electrolytic, so I guess the change couldn't have been for worst.
As about your setup over there... Why? How long is the cable? Any numbers to show the difference between what you got and the standard spdif? |
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| MalichiConstant |
" The fact are that the ceramic caps that I mentioned or made for high frequency signals, it obvious by you statement that you don't a have clue."
I guess I don't have a clue either.......... BTW, find me a decent 0.1uF
(or even 0.01uF) ceramic or Mica cap. :whazzat: All I can do is recommend people listen. It is not that hard to find a decent MKP cap from DigiKey of Mouser. The signal is digital statement is not going to fly very far since there is a whole engineering discipline devoted to optimizing the digital interface between ICs. Its called Signal Integrity. Tell them Howard Johnson said so.
http://www.sigcon.com/ |
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| emuman100 |
roibm,
Only reason why is because I got the idea from a web site and it sounded pretty cool. With my tests, I've gotten 100ft with no noise or audio degridation. I don't have any numbers though. With these RS485 transceivers, you can use cable lengths of thousands of feet. |
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| roibm |
| quote: | Originally posted by emuman100
Only reason why is because I got the idea from a web site and it sounded pretty cool. With my tests, I've gotten 100ft with no noise or audio degridation. I don't have any numbers though. With these RS485 transceivers, you can use cable lengths of thousands of feet. |
That does sound impressive :)
My cable is ~5m long(~15ft). I think it is ok, but for longer distances your implementation could be better. Never tried it tho, but if I'll need to run longer distances, I'll give it a try.
Could you post the link of that website? |
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| emuman100 |
roibm,
I forget where I found the site, but all it suggested was to use RS422 tranceivers for long distance transmission. I ordered some samples of the MAX3443 from Maxim and am quite satisfied with the circuit. Read the datasheet so you know the pinout and know how to hook it up. It's a simple 8 pin device. Just connect the TTL level signal to it's receive output or transmit input and you will be good to go. |
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| jean-paul |
| Maybe ADM1485 is a good choice for that purpose. Those were the ones I had in mind once for this purpose but I never used them as the project was cancelled. |
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| emuman100 |
| They will all perform just as well. You can wait two weeks to get samples of MAX3443's, or order SN75176's from digikey, or get samples of SP1485's or SP1481's or ADM1485's. Just make sure it can operate at a minimum of 10Mbps, and is compatible with TTL levels of your circuit. I have everything powered with 5V and I have it operating for hours on end with no problems at all. I really suggest you all try it. In my opinion, it should be a 3rd option of transmission. |
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| Cameron |
What is the mechanism by which a coupling capacitor in this kind of application would have any impact on the final sound?
I could accept that it might somehow increases jitter. By temporal smearing of the transients due to various capacitor related imperfections, like dissipation factor. Or perhaps creating an impedance gradient and thus reflections, also smearing the transients.
But other than that, what could it possibly do? It's a digital signal. Change the bits? As far as I am aware, there are only two issues. Is the data stream coming out the spdif receiver correct or not. And the quality of the clock. What else is there?
If the capacitor is so crappy as to actually corrupt the data stream, I imagine you are well past the PLL being able to lock onto the clock anyways. |
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| tubenut |
| quote: | Originally posted by Jocko Homo
If you are only going to use a transformer on one end, make it the RX end.
Kepping the outer conductor on th TX will reduce radiation, although there are ways around that. Besides, the RX side will perform better if you use a differential input, and the best way to do that is with a transformer.
Jocko |
What can one do to reduce radiation. I have a Theta Data 2 transport connected to a Theta DS Pro Basic 2 DAC. The transport has a transformer on the SPDIF and the AES EBU outputs, the DAC is straight in (no transformer). The transport creates terrible interference on TVs in the house, even when not connected to the DAC. As soon as the coax out cable is plugged in to the transport it seems to work as a transmitter aerial and causes diagonal lines on my TV.
Any suggestions on how to kill this. Cable used is either Monster MV1000 silver or Belden 1695.
Would like to keep my CD gear warm when the soon to be wife wishes to watch soaps and other drivel.
Thanks |
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| ekaerin |
Hi there,
I recall when I made my coax to optical converter the amplitude of the
SPDIF was something like 1Vpp. Dunno if the TTL level may be a problem though.
/ Mattias |
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| Jocko Homo |
Tubenut:
In your case, I would try a 0.1 uF cap from the shield on the TX end to ground, unless it is already grounded by the connector. The chassis will need to have a good earth ground.
A ferrite bead, right at the TX end, may help. Putting inside the chassis where it exits is where I would try.
The SPDIF level is 0.5 V_pp, terminated into 75 ohms. It will be 1.0 V_pp if it is not properly terminated.
Jocko |
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| jewilson |
| quote: | | What can one do to reduce radiation. I have a Theta Data 2 transport connected to a Theta DS Pro Basic 2 DAC. The transport has a transformer on the SPDIF and the AES EBU outputs, the DAC is straight in (no transformer). The transport creates terrible interference on TVs in the house, even when not connected to the DAC. As soon as the coax out cable is plugged in to the transport it seems to work as a transmitter aerial and causes diagonal lines on my TV. |
1) make sure that you TX out is not connected to circuit ground of the player. If it's AC couple to ground, cut one lead of that cap. These caps or suppose to help however many time they make thing worst.
2) With the DAC disconnected check the noise of the shield. If that is high can be a problems.
3) Check to see what logic family is used in the Transport, some families can cause severe noise problems.
4) Purchase a Transformer receiver interface assembly from Jocko Homo, this will isolated at both ends and sound better.
5) Purchase a TX card from Jocko-Homo and a timing card.
6) Check the ground of both systems.
7) or 1) Call Theta and complain about the problems. They should not be selling equipment that interferes with communications and TV. This equipment would never pass the FCC tests.
:) |
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| ekaerin |
Yap,
Jocko you are right about that. I was measuring on an opened output. I also remember it was centered around ground.
/ Mattias |
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| Cameron |
| Is no one going to attempt to answer my questions? |
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| jewilson |
| quote: | | Is no one going to attempt to answer my questions? |
Cameron,
Regards the cap; you need a good cap that designed for hf applications. Much has been written about the different mechanism and non linear effects of capacitor so I don’t fell like repeating it, however it can be found by searching the web. In addition, caps like electrolytic suck at high frequency and be come inductors so they should not be used. Caps like Mica and NPO ceramic work well in high frequency applications.
Dielectric absorption, inductance leakage, or just a few problems there are caps. Dielectric Absorption is one of the biggest offending and a enemy of the audio signals, however, since we in the digital domain and DA it not the critical since we only care about the using the change of the signal going zero to a positive state and from a positive to a negative state. So, Mica very good application, ceramic can be also if the correct material is use like an NPO or even a X7R not a Z5U. ALso, you can use some film cap here too.
:) |
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| Cameron |
I am quite familiar with how various capacitors and their properties affect analog signals. But were not talking about analog. At least in the traditional sense. It is analog if you consider the infinite fourier expansion of sin waves which create the square wave. And transmission line theory.
So my questions stand unanswered. An electrolytic would clearly trash the datastream and the receiver pll would not even lock. I agree a silver mica or ceramic should work well. The datastream should come through nicely. Though they were described as sounding bright and harsh respectively. How is that possible? What are they doing to the data which could change it's sound?
This same question applies to the sound quality of digital transports. Why would one sound any different than another? I assume they all output exactly the same bits. Otherwise you have either a crappy transport or damaged source media. The only thing left of which I am aware is clock quality/jitter. Consider a DAC with an asynchronous resampler. The jitter attenuation is massive. So that possible source of "sounding different" should be virtually gone. What else is there? |
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| MalichiConstant |
| Come on people ........there must be someone who can help this guy out...... |
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| jewilson |
| quote: | | I could accept that it might somehow increases jitter. By temporal smearing of the transients due to various capacitor related imperfections, like dissipation factor. Or perhaps creating an impedance gradient and thus reflections, also smearing the transients. |
:idea:
Anytime a digital signal goes though a capacitor it enters the world of non linear effects, that’s the analog world. So it seems that you have answered your own question. |
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| Jocko Homo |
To reduce the EMI, the shield should be grounded. Maybe it will muck up the sound, as it did for you. But leaving the shield "hot", as it will be in a transformer coupled output, will radiate.
Thanks for the kind words, but none of my boards will reduce EMI. That is not their purpose. Yes, going to a slower logic family may cut it down, but then too many harmonics are chopped off.
The Philips based CDPs that I have worked on did that. While EMI was low, they sounded awful.
As for why a cap can muck up the sound..........
If it is too inductive, it will cause reflections. And reflections make stuff sound bright. Sometimes. Always awful, just the type and degree vary from case to case.
Jocko |
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| Cameron |
Jewilson: You are thinking superficially. And I definitely did not answer my own question. How do these non linearities impact the sound? They are JUST BITS at that point. Way down the digital chain at the last stage of the DAC chips are some switching transistors creating the analog audio current output. There is no analog audio before that. How is the way those transistors are switching influenced by slight distortion in the analog characteristics of the data stream way back there? You say there are numberous articles on capacitors and their effects. Refer me to just one that discusses how coupling capacitors effect DIGITAL signals. I have not been able to find any.
Jocko: How do reflections make it sound bright? I can easily see how reflections can smear the transients and thus increase jitter. So does this all boil down to recovered clock jitter? If so, I am very unclear as how increased jitter will lead to sounding bright. It can definitely trash signal to noise ratio at high frequencies. Hum. That could end up sounding very harsh on the high end. Bright sorta implies too much high end but still relatively clean.
My moto: Suspect any statement which can not be explained. |
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| jewilson |
| quote: | | My moto: Suspect any statement which can not be explained. |
You need to change you motto to suspect any statements that are not explained to your liking. To the rest of us, that sound like your demanding we debate you till your happy with the answer or you succeed in starting an argument :hypno1: |
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| jewilson |
| quote: | | To reduce the EMI, the shield should be grounded. Maybe it will muck up the sound, as it did for you. But leaving the shield "hot", as it will be in a transformer coupled output, will radiate. |
Jocko, your right you can't remove the ground if you not couple at both ends. However, you can determine if the CD ground is adding to the noise of you converter, by scoping out power on or output the dac while disconnecting the SPDIF cable.
As for my application the CDP, AC ground cap to the SPDIF was a major source noise for the DAC. Removing that cap caused the noise in the dac dropped, to bad a did run a S/N measurement. :) |
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| Cameron |
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. |
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| Cameron |
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. |
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| Jocko Homo |
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 |
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| fastcat95 |
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 |
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| Cameron |
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. |
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| tubenut |
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 |
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| lucpes |
| quote: | Originally posted by Jocko Homo
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.
|
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.
| quote: | Originally posted by fastcat95
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.
|
Can't you post it here? Please :)
| quote: | Originally posted by Cameron
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?
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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. |
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| ergo |
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 |
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| Cameron |
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. |
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| lucpes |
| quote: | Originally posted by Cameron
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? |
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 :angel: |
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| jbokelman |
| quote: | Originally posted by Cameron
How could a coupling capacitor in the digtal domain effect the final audio sound quality.
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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:| quote: | | One problem with the Schmitt trigger input of the 74HCU04 is that it reflects **** back on the S/PDIF line. To solve that problem, you can use a fast comparator at the input. |
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| Elso Kwak |
Hi, To quote a quote:| quote: |
One problem with the Schmitt trigger input of the 74HCU04 is that it reflects **** back on the S/PDIF line. To solve that problem, you can use a fast comparator at the input. |
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........
:rolleyes: |
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| 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 |
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| Jocko Homo |
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 $$$ and stuff that they don't know or have access to.
And just who is pushing clocks???? No one has mentioned clocks in this thread.
If you have a problem with that, then tough.
Jocko |
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| BrianL |
Pedja,
I'm not so sure that this is a correct conclusion. Jitter in
the clock, more correctly - "phase noise", can be viewed
as a signal that intermodulates with the desired signal.
Thus it doesn't necessarily follow that a greater amount
of LF phase noise is less audible. |
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| jbokelman |
| quote: | Originally posted by Jocko Homo
Well, I'm glad that we have a "real" self-proclaimed expert here.
None of us has ever said that you don't need to have clean clock at the DAC. You solution is above the capabilities of the average Joe here. We are trying to get them to implent things that are easy to grasp, and make it sound better, without $$$ 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 |
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? |
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| Guido Tent |
| quote: | Originally posted by jbokelman
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. |
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 |
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| Guido Tent |
| quote: | Originally posted by jbokelman
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? |
analog signals do not contain jitter |
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| OliverD |
| quote: |
How in the world can someone with even a modicum of knowledge and experience confuse a CS8412 with a 74HCU04?
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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 |
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| Jocko Homo |
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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| jbokelman |
| quote: | Originally posted by Guido Tent
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.
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The fact that you and your customers don’t like the AN DAC3 in no way invalidates my statement that Audio Note DACs are highly regarded. (I don’t like the DAC3, either.) AN has many happy customers and many rave reviews in the mainstream and not-so-mainstream audio press. I haven’t seen any rave reviews of your DAC in any audio press.
| quote: | Originally posted by Guido Tent
analog signals do not contain jitter
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Have you ever looked at the output of a DAC? It’s a stair-stepped ANALOG of the digital input. If there is jitter in the timing of those stair-steps then there is jitter in the analog output of the DAC. You claim to be able to hear 1ps of jitter. If the jitter didn’t cross the boundary that separates the digital and analog domains and appear in the analog output of the DAC, how can you hear it?
Look at the data sheet of most any multi-bit, audio DAC and you will see a spec for settling time. That’s the time it takes for the analog output to reach the level that represents the ANALOG of the digital input. The TDA1541, Thorsten’s favorite, settles to within +/- 1 LSB in 1 MICROsecond. Although the Philips spec doesn’t give the range settled, the norm for settling time is a one-half of a full-range current swing. The settling time varies depending on the numeric distance between consecutive samples and, in some cases, the number of bits changing. A smaller current swing will settle in less time. If two successive samples are identical, the settling time is zero.
Fact: The TDA1541 typically takes up to one MICROsecond, or longer, for the correct current to appear at the output of the DAC.
Fact: The actual settling time depends on the data.
Conclusion: It looks like data induced jitter to me.
| quote: | Originally posted by Jocko Homo
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?
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Good question, but I didn't say DACs had microseconds of jitter; I said the jitter was MEASURED in MICROseconds, as the Philips spec proves. |
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| guido |
| quote: | Originally posted by jbokelman
Fact: The TDA1541 typically takes up to one MICROsecond, or longer, for the correct current to appear at the output of the DAC.
Fact: The actual settling time depends on the data.
Conclusion: It looks like data induced jitter to me.
Good question, but I didn't say DACs had microseconds of jitter; I said the jitter was MEASURED in MICROseconds, as the Philips spec proves. |
Err,
Isnt this just a microsec of DELAY (BCK change to analog output change) which is not always exactly 1.0000... microsec but a bit more/less. That would then be the jitter which comes from:
- DELAY not being equal for all data/steps between samples
internally in the DAC = jitter
- BCK jitter.
Can't imagine one can hear changes to the clock jitter (induced with BCK) if the delay would range from 0 to microseconds depending on the data. But if there is an average delay of one microsecond, jitter would not be in microseconds.
The BCK jitter is what can be minimized by working on the clocks.
Writing this without looking at the spec or having ever done any jitter measurements (with what ?? :D ).
The other |
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| Guido Tent |
| quote: | Originally posted by jbokelman
The fact that you and your customers don’t like the AN DAC3 in no way invalidates my statement that Audio Note DACs are highly regarded. (I don’t like the DAC3, either.) AN has many happy customers and many rave reviews in the mainstream and not-so-mainstream audio press. I haven’t seen any rave reviews of your DAC in any audio press.
Have you ever looked at the output of a DAC? It’s a stair-stepped ANALOG of the digital input. If there is jitter in the timing of those stair-steps then there is jitter in the analog output of the DAC. You claim to be able to hear 1ps of jitter. If the jitter didn’t cross the boundary that separates the digital and analog domains and appear in the analog output of the DAC, how can you hear it?
Look at the data sheet of most any multi-bit, audio DAC and you will see a spec for settling time. That’s the time it takes for the analog output to reach the level that represents the ANALOG of the digital input. The TDA1541, Thorsten’s favorite, settles to within +/- 1 LSB in 1 MICROsecond. Although the Philips spec doesn’t give the range settled, the norm for settling time is a one-half of a full-range current swing. The settling time varies depending on the numeric distance between consecutive samples and, in some cases, the number of bits changing. A smaller current swing will settle in less time. If two successive samples are identical, the settling time is zero.
Fact: The TDA1541 typically takes up to one MICROsecond, or longer, for the correct current to appear at the output of the DAC.
Fact: The actual settling time depends on the data.
Conclusion: It looks like data induced jitter to me.
Good question, but I didn't say DACs had microseconds of jitter; I said the jitter was MEASURED in MICROseconds, as the Philips spec proves. |
The DAC output is the result of both data and clock.
The settling time should not be confused with jitter.
Assumed that the data is correct, the analog signal distortion due to the DAC depends on conversion timing errors as the timing error is converted in an amplitude error.
It can easilly be meaured, I did that feeding known jitter to a PCM63 (using a VCXO). Sidebands appear, depending on amplitude and frequency of jitter added (the PCM63 is more jitter sensitive for higher audio frequencies).
I never said my customers didn't like their AN-DAC3, but they all seem to prefer either our own DAC or the DAC clock upgrade I offer.
Our DAC is never reviewed as it is a DIY DAC hence a direct thread to the advertisement income of any audio magazine.
cheers |
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| jbokelman |
Of course you don’t hear the effects of settling time because it is omnipresent. You would probably recognize it once it’s gone.
Settle time illustration
Pictured is a highly stylized representation of the effects of settling time. The dotted lines show the output of an ideal DAC. Samples 1 and 6 are both zero. Samples 2 and 5 are same and are a little higher than zero. Samples 3 and 4 are the same and are a lot higher than zero.
The solid lines show the output of a real DAC. The pairs of samples that should be identical are not. They differ in the time to reach the correct height, the average height during the sample period, and the area under the curve for each sample period.
I called the aberrations caused by settling time “intrinsic jitter” because jitter seems to be the only digital aberration discussed in this forum. Also, the effects of settling time are similar to jitter. Jitter, as you all know it, changes the time a sample starts and stops and that changes the area under the curve and the amount of work/energy that sample imparts to the resulting output signal. As you can clearly see, the effects of settling time also affects the area under the curve. And like some forms of jitter, settling time aberrations are related to the data.
In the old days of parallel DACs, S/H registers were commonplace. Because the DACs were expensive and required numerous latches and support logic, the left/right channel data was decoded by a single DAC and the left/right channel output was separated by S/H registers. The use of S/H also eliminated the effects of settling time and all upstream clock jitter. In fact, recommended S/H delay times were part of the DAC chip specs.
As DACs got faster, settling time became less of an issue and the pressure to reduce costs led to the serial interface, stereo DACs, upsampling, etc. and the S/H was eliminated. With the renaissance of non-oversampling and the rediscovery of ancient DAC chips, like the TDA1541, I am surprised no one has resurrected the venerable S/H. |
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| Elso Kwak |
Jbokelman,
I guess your picture is wrong. Attached a scope picture of the analog output of a 6k5 squarewave of a 8x oversampling DAC before low-pass filtering. (CXD1244 + AD1865). Your picture looks like simple lowpass filtered of this.:bigeyes: |
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| rfbrw |
| Methinks ye have missed the point of the illustration. |
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| mikewu99 |
The settling time of a DAC can be split into two types: linear settling and non-linear settling. Linear settling is primarily determined by RC time constants inside the DAC chip and/or the time constant of an external I/V converter. It is exactly equivalent to lowpass filtering the DAC output, typically at a fairly high cutoff frequency (time constant << DAC sample rate). As long as the time constant is short enough linear settling does not degrade the DAC output and is in no way equivalent to jitter.
Nonlinear settling is most commonly caused by slew rate limitations in the I/V converter. Proper design of the I/v converter WRT large-signal bandwidth and output swing can minimize the effects of slew rate. A well designed DAC will have negligible nonlinear distortion.
There is a possible signal-dependent jitter source inside the DAC chip and that is internal clock skew on the sample clock. Typically the sample clock needs to be gated and routed to a number of analog switches (transmission gates) which have some physical separation on the chip; if the clock does not arrive at precisely the same time this can cause a jitter-like effect. The effect is highly dependent on the DAC topology. In any case, careful physical design (layout) of the DAC chip can bring the clock skew down to the picosecond level (or better). |
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| guido |
Just info:
TDA1541: 1 microsec typical
TDA1541A: 0.5 microsec typical
TDA1545: 0.2 |
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| jewilson |
jbokelman
Your statements here regarding Sample & Holds are completely humorous. First, the current generations of DAC’s have deglitching circuitry! That means a S&H is not required or even remotely needed. So, If we add a sample hold it would have to be a discrete one cause the monolithic ones or horrible. That means adding bunch of parts. So now, for no reason at all we have had to add new timing hardware, match fets, resistors opamps, low DA and Leakage caps to re-quanatize something that already been done, not to funny. So would you make these recommendations in design reviews?
As far as Bi Phase Mark goes would you even know how to decode or encode it from NRZ data stream, I bet not. My self, I did my first Bi Phase Mark decoder and encoded at TI back in 1981, whoopee.
So it seems you enjoy your habit here of arguing over issues that you just don’t completely grasp. Maybe, if you take the time to listen to Guido, Jocko or others you might be able to improve your technical skill in this engineering area.
Back to jitter, while we do have some intrinsic jitter in all digital systems the major source of jitter in the audio CD DAC chain is cause by a number poorly implemented design issues. System noise, grounding methods, power supplies, master clocks, logic types, impedance mismatches – reflections and more. |
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| jbokelman |
| quote: | Originally posted by jewilson
Your statements here regarding Sample & Holds are completely humorous. First, the current generations of DAC’s have deglitching circuitry! That means a S&H is not required or even remotely needed. |
The thing is, most of the DAC projects around here are not using current generation chips. The favored chips are close to 20 years old and they don't have deglitching circuits built in. If they did, the datasheets wouldn't specify 1us typical settling times. Maybe you missed that part of the discussion.
| quote: | Originally posted by jewilson
So would you make these recommendations in design reviews? |
You’re damn right, I would, and I’ve had many heated discussions with EE’s trying to get them to “think outside the box,” as they say. But this is not the time or place for me to list all my engineering accomplishments in both software and hardware.
| quote: | Originally posted by jewilson
My self, I did my first Bi Phase Mark decoder and encoded at TI back in 1981, whoopee. |
Whoopee, indeed. You must be one of them EE-types. I’ve worked with a number of EE’s through the years. They’re real smart but they have no imagination. Their world-view is limited to what they learned in school.
| quote: | Originally posted by jewilson
Also, as the clock speed increases for High Speed CMOS the it will use more power than ALS at high speeds. |
I thought the switch to CMOS years ago was, in large part, to reduce power consumption. But what do I know. I’m just a dumb programmer. Yeah, I’m real dumb. That’s how come I was able to retire at very young age while all you real smart, college educated, EE-types are still working. |
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| Jocko Homo |
BFD.
I guess that makes you smarter and richer than Bill Gates and/or Ross Perot.
Like any of us care, or are impressed.
No imagination?????? Speak for yourself, not us.
But I must be smart, since I never learned squat in school.
Except how to **** off all the professors.
Jocko |
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| pinkmouse |
:cop:
Guys, cool the personal comments please, let's keep this civilised.
:cop: |
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| jbokelman |
| quote: | Originally posted by jewilson
So it seems you enjoy your habit here of arguing over issues that you just don’t completely grasp. Maybe, if you take the time to listen to Guido, Jocko or others you might be able to improve your technical skill in this engineering area. |
Good suggestion. I spent some time pouring over the archives to see what I could learn from Jocko and Guido. Jocko’s expertise appears to be in RF, not digital. When Bernard wanted to know which 74 logic family to use for >100MHz and Jocko recommended ECL. Is ECL a 74 logic family?
| quote: | Originally posted by Jocko Homo
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. |
Is ECL within the capabilities of the average Joe?
Guido’s expertise seems to be in layout, grounding, and related items. He really likes serial resistors and he uses some unorthodox symbols in his schematics. For example, he uses the symbol for an AND gate to represent a NOR gate. I don’t know, maybe they use different symbols in Europe. The way he organizes the clocks in his DAC violates every rule of proper clock generation and distribution I learned when I worked for a mainframe company. |
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| Jocko Homo |
That is because it was a mainframe company, not the inside of a CD player or DAC.
But you are correct. RF is my background.
You can build your own ECL for our purposes with 2 transistors, 3 resistors, and a bias network. How hard can that be??
As for S/H circuits...........
They were commonplace at one time (YUK!), and the DAC chip makers learned how to make decent ones that did not need de-glicthers. In the meantime.........it gave rise to companies like UltraAnalog, who's main selling point was that they invented a special de-glitcher.
One that allowed them to use **** like 5532s in the I/V stage.
(You can't prove that by looking inside one.........they remove all the ID markings. But if you want one, I have 2 to sell. Cheap.)
Speaking of I/Vs......and no imagination.........I wonder who can name the goofball who was one of the first......if not the first......to use a true transimpedance amp for an I/V????
Anyone????
Hint: It was back in '89............
I don't need no steenking boxes............
Jocko
Only semi-retired |
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| jewilson |
| quote: | | I thought the switch to CMOS years ago was, in large part, to reduce power consumption. But what do I know. I’m just a dumb programmer. Yeah, I’m real dumb. That’s how come I was able to retire at very young age while all you real smart, college educated, EE-types are still working. |
The facts or Bud at high speeds HCMOS will uses more power that ALS, check out the clock speed vs. power. But that not the real issue, I was discussing noise cause by some logic families when they undershoot ground and how that can effect other things like jitter.
Well it good to know that some one has taken up the cause to be the DIY biographer and has proclaimed him self Grand OZ..
:eek: |
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| Guido Tent |
| quote: | Originally posted by jbokelman
Guido’s expertise seems to be in layout, grounding, and related items. He really likes serial resistors and he uses some unorthodox symbols in his schematics. For example, he uses the symbol for an AND gate to represent a NOR gate. I don’t know, maybe they use different symbols in Europe. The way he organizes the clocks in his DAC violates every rule of proper clock generation and distribution I learned when I worked for a mainframe company. |
I have an analog background.
Among that I design pick up units for high speed optical recording (so far 16X DVD, but blu is cooking in our kitchen)
The essentials of AD and DA are in the conversion. That's an area where analog, RF, EMC and layout are quite welcome.
I am not an expert in digital, maybe that is why I mix up symbols.
My clocking scheme may differ from what mainstream industry does, but that probably tells more about that industry than about myself.
best regards |
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| kittaylor |
Reading this prompted me to try a pair of 10nF 1206 NPO coupling caps on my nonOS dac.
With a 1206 resistor they make a very neat and compact package. They don't sound worse inferior than the Wima polyprops and Welwyn RC55 they replaced, perhaps a bit sharper but nothing that I imagine would cause trouble in a balanced sounding system. |
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| ABO |
Inspired by this thread I did some cap swapping too. I tried several ceramic and plastic caps. I even tried bypassing with silver mica caps.
All film caps sounded veiled, including vishay KP1830. The ceramic caps I tried were notably better. The best result gave some Kermet NPO cap from mouser. I don't remember the actual type.
Bypassing didn't make an audible difference in any case.
All in all a very worthwhile upgrade. |
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