| garbage |
hi all
i have a monica 1545a dac and want to do the reclocker circuit as suggested at the same site for it.
question is, will an AES/EBU 110ohms transformer (assuming i am using balanced XLR cable from the transport) at the input of the CS8412 make any difference or should i just stick to RCA 75ohms and use normal coaxial cables?
thanks. |
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| hwb |
Hi!
Basically Coax is the better solution for digital audio interconnection, as it is designed for HF signal transmission, opposite to ordinary twisted pair cable. Even special 110Ohms digital microphone cable is always worse in transmission quality compared to Coax.
Regarding transformers, have a look at Cirrus' AN134, there you will find some references to background knowledge. Especielly Scientific Conversion has some interesting Papers about the influence of transformers on Jitter.
HTH,
Holger |
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| Jocko Homo |
Mostly true, but not for exactly those reasons.
Main problem with AES/EBU is the connector. While it is a 110R connector, it is unusable due to excessive reflections.
There are twisted pair cables designed for data transmission, that would work well if the proper connector was used. But sourcing them will be a problem, so coax does win out. As long as its impedance is matched to the impedance of the TX and RX circuits, as well as the connector.
Don't believe everything that Scientific Conversions says. In order for the jitter to be low, the reflections must be low. They provide no information on that, just a lot of their ideas about noise performance.
Jocko |
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| garbage |
looks like i'll go with plain jane.
initially wanted to order some sc939-06 trans from scientific conversions.
cheers
garbage |
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| hwb |
Yocko, you wrote: "Main problem with AES/EBU is the connector. While it is a 110R connector, it is unusable due to excessive reflections."
I think you've forgotten a "not" in your sentence. XLR is no 110Ohms connector and never will be. Even Neutrik's XCC Series, especially designed for the use with AES/EBU, has a lower impedance. It is kind of stupid, that the AES has specified XLR as connector for this signal.
The TP digital audio cables have 110Ohms, but usually +/- 10%. And if you bend the cable you get some inhomogeneities within the cable causing a decrease in signal quality, whereas a coax cable has by nature a more narrow tollerance and better homogeneity.
At least in professional applications with longer cable runs I will always prefer coax together with BNC connectors and transformers at the end.
Holger |
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| Jocko Homo |
I hate to tell you, but it really does measure 110R. If you can eliminate all the extraneous garbage on the TDR trace.
Forget the impedance.......it will never work as the pins are so far apart that it will have too many reflections.
You are correct that the cables that you refer to will have problems. Back in my telecom days, we had twisted pair cables for balanced transmisson that worked quite well.
The reason that they worked is that you could not bend them. Much too rigid. Basically, the only place that we used them was in elevator shafts, to transmit data between locations many floors apart.
And, yes...........transfomers on the end!
Jocko |
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| hwb |
Hmm, I've had a discussion with a representative of Neutrik on this topic, he told me, that the impedance of a XLR is lower than the required 110Ohms. But I don't want to argue about that, we both agree that a XLR is unsuitable for AES/EBU, no matter if it has the correct impedance or not.
BTW, I have heard some rumour that the AES is thinking about a new connector for AES/EBU or using a coded XLR variation. If that was true, they would badly need a reality check, coming up with this idea ten years after they have issued their standard... :bigeyes: |
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| Jocko Homo |
They needed a reality check years ago. The first version allowed for.......what was it..............3 TX and 7 RX on each cable??? Something like that. I tried to tell the AES, but would they listen...............
Eventually, they figured out part of it, and went to one T/R only. Still, the whole concept is flawed.
I guess if you include all the stray capacitance in the impedance, it would be lower than 110R. But it is no longer resistive, so like you said, it still doesn't matter.
Jocko |
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| Jocko Homo |
Anyway........we have strayed a lot here.
What you are refering to is commonly called an asynchronoius reclocker. Elso is fond of them, as are a few others. I have no use for them. Seems like another flawed concept.
Secondary PLLs, or ASRCs, are the only real ways to get rid of the jitter on the DAC.
Jocko |
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| stolbovoy |
| quote: | Originally posted by Jocko Homo
Secondary PLLs, or ASRCs, are the only real ways to get rid of the jitter on the DAC.
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What about source clocking + reclocking from master XO placed close to DAC chips? |
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| Vadim |
| quote: | Originally posted by stolbovoy
What about source clocking + reclocking from master XO placed close to DAC chips? |
Why would you want to re-clock, when the ASRC with about 20-bits resolution is available? Another thing to keep in mind is that your anti-imaging filter and its analog power supply will degrade the overall S/N down even further, therfore ASRC is not the weakest link in your processing chain.
When the signal finally gets to the output RCA jack you will have 16-bits of resolution left providing you are lucky and everything worked perfectly.
THis is the reason why there are very few outboard DAC's out there with better then 16-bit output, and none of them ever showed even an 18-bits output.
All in all the re-clock will not work any better then the ASRC like AD1895(6), - in fact my take on it is that it will be next to impossible to match the AD1896 if you set the input and the output frequencies to be the same (no rate converssion) and do a good job generating a clean output clock. |
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| rfbrw |
The Monica2 is a NOS dac i.e No oversampling dac. No digital filter. Adding an ASRC would be a hanging offence.Why, if I were a nosser, I'd sooner replace the chocolate flake in my icecream with a dog turd. :D
Vadim, I fear in thinking like a sane rational engineer, you completely miss the point of the asynchronous reclocking referred to in the opening post. It totally defies reason and it seems like adding correlated jitter to me but its proponents, and there are quite a few in this place, all swear by it. |
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| stolbovoy |
| quote: | Originally posted by Vadim
in fact my take on it is that it will be next to impossible to match the AD1896 if you set the input and the output frequencies to be the same (no rate converssion) and do a good job generating a clean output clock. |
How AD1896 can be better then simple reclocking when input and output frequencies are equal (same XO)?
| quote: | | Why would you want to re-clock, when the ASRC with about 20-bits resolution is available? |
Because I don't know which algorithms they are using inside, and I want to experiment with my own. Also, I could test my tract up to DF for bit-accuracy :) |
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| rfbrw |
| quote: | Originally posted by stolbovoy
Because I don't know which algorithms they are using inside, and I want to experiment with my own. |
Do you wish to design your own digital filter? |
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| stolbovoy |
| quote: | Originally posted by rfbrw
Do you wish to design your own digital filter? [/B] |
I'm thinking about it :) May be I'd just made a switch to turn hardware DF on/off. |
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| rfbrw |
| quote: | Originally posted by stolbovoy
I'm thinking about it :) May be I'd just made a switch to turn hardware DF on/off. |
Why not go one step further with glitch free switching. The HDCD demoboard for the PMD100 had two digital filters on it and it had glitch free switching so you could switch between the two without interrupting the music. That way you could switch back and forth comparing the two filters. I've often thought such a facility would be useful in other areas. For example, you could switch between different filters or even switch out the filter on the fly depending on the music playing. I've always thought it would be nice to have a dac that allowed me to switch between the PMD100, YSF210, SM5842 and the SM5847. |
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| Vadim |
| quote: | Originally posted by rfbrw
The Monica2 is a NOS dac i.e No oversampling dac. No digital filter. Adding an ASRC would be a hanging offence.Why, if I were a nosser, I'd sooner replace the chocolate flake in my icecream with a dog turd. :D
Vadim, I fear in thinking like a sane rational engineer, you completely miss the point of the asynchronous reclocking referred to in the opening post. It totally defies reason and it seems like adding correlated jitter to me but its proponents, and there are quite a few in this place, all swear by it. |
So we are up against a non-over sampling D/A, - WOW! I guess I should take my medicine now and or go back to school.
I truly fail to see a reason behind such design. It seems to me that a non-oversampling D/A would violate several principles of DSP, resulting in a rather noisy output. My guess, - 13 bits (on a good day) is all that is possible to get out of Monica2, - may as well stick with a turntable.
Certainly no amount of relocking will ever help matters. In fact, the contribution of correlated jitter to an overall THD+N numbers is completely negligible in light of the aliasing artifacts that will very much be present due to the 44.1 kHz sampling rate and 20 kHz signal bandwidth.
So, - in the case of a non-over sampling DAC approach, the relocking circuit is rather pointless. Come to think of it, - the ASRC used as jitter attenuator is useless just as well.
Vadim |
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| stolbovoy |
| quote: | Originally posted by Vadim
It seems to me that a non-oversampling D/A would violate several principles of DSP, resulting in a rather noisy output. [/B] |
But isn't it ultrasonic noise, which would not necessary manifest itself in audible range? |
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| Vadim |
| quote: | Originally posted by stolbovoy
But isn't it ultrasonic noise, which would not necessary manifest itself in audible range? |
Unfortunately the aliasing spectra are not ultrasonic at all. Here is the picture; - with no over-sampling the first image of the 20 kHz –wide data will appear at 44.1 kHz, contaminating the data substantially to put it mildly.
Without a proper anti-imaging filter the noise rejection is done by the D/A’s inherent zero-order hold function, that looks like [sin(x)]/x, with x=wT/2, w=2piF and T=period of the 44.1 kHz sinusoid. At 20 kHz this amounts to less then 0.2 dB. That means that the zero-order-hold phenomena only reject 0.2 db of noise at 20 kHz. This is all the noise rejection you will get if no anti-imaging filter is used. Noise city!
Even if you design in an anti-imagine filter, its order would be prohibitively high for the filter to have low or even linear group delay. To appreciate this you would need to ponder the requirement to reject about 96 dB or 16-bit equivalence in a space of about 4 kHz, which is a space between the edge of the first image and the edge of the data. Anyway, with such a fantastic requirement for an analog filter, I would forget about designing this filter if I have no over-sampling to begin with.
So, in a nutshell, here you have it, - with no over-sampling you cannot reconstruct the original 16-bit data at all. You may recover about 12-13 bits at the most, although I doubt that even that is possible.
Well, on the other hand, who is to say that listening to a dozen bits is a bad experience? I cannot speak for sound quality, but I can say this, - that a $100 antiquated turn-table will most likely do better in terms of signal integrity then a non-over sampling DAC. |
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| garbage |
| quote: | Originally posted by Vadim
Well, on the other hand, who is to say that listening to a dozen bits is a bad experience? I cannot speak for sound quality, but I can say this, - that a $100 antiquated turn-table will most likely do better in terms of signal integrity then a non-over sampling DAC. |
i think most people into vinyl will agree that a basic tt will trounce most basic cdplayers.
that aside, i would like to say that the non-oversampling dac really is nice to listen to, regardless of technical specs. it sounds more relaxed, open and non-fatiguing. |
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| Vadim |
It never seizes to amaze me that there are designs out there that make so little sense. The DAC’s on the site you linked to definitely fit this category. I looked at the most expensive DAC they have; standing at near $50,000 it is a testament to a business highwayman style.
Curiously the D/A chips used are the old AD1865, which were produced in 1988-89 time frame. The AD1865 cannot output 16-bit signal, - oh, well…Here is the quote that particularly grabbed my attention, -
“…the Audio Note™ 1xoversampling™ circuit is a genuine sonic revelation and demonstrates the potential of the 16Bit format to a far greater extent…”
I really like that statement, as it makes me feel all warm and fuzzy inside. However, what this statement demonstrates is a complete luck of understanding of the engineering principles that govern reality.
It seems that the immortal words of late Scotty the engineer, who said
“… I canna change the laws of physics…” - do not apply here. |
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| Vadim |
| quote: | Originally posted by garbage
i think most people into vinyl will agree that a basic tt will trounce most basic cdplayers.
that aside, i would like to say that the non-oversampling dac really is nice to listen to, regardless of technical specs. it sounds more relaxed, open and non-fatiguing. |
I have no problems with turn-tables. I would however argue that our hobby here is all about sound reproduction and not about sound generation. With this in mind, it should be clear that the goal is to faithfully reproduce what has been recorded on a CD or the vinyl record. The common logic would then dictate that any perceived sound which can be attributed to an electronic component, be it a turn-table or a DAC, - any sound at all, is a problem. Wouldn’t you agree?
The goal of a sound reproduction is to have an ideally zero contribution from the play-back device, whatever that device may be. Hence, - the best and arguably perfect electronic component must have no sound of its own by definition.
I, therefore submit that in this regard a CD beats the turntable any day. I further submit that a DAC with no over sampling facility will introduce an unwanted and very high distortion into the signal that your expensive speakers are trying to deliver to you. Let along the fact that a great deal of information will be simply lost to noise in this case just as well. |
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| Jocko Homo |
Agreed. And it seems that you can not change opinions that are without foundation.
Wish that I had a $ for every non-o/s maven that has told me "How do you know how it sound? Just try it, you'll be surprised."
No, I don't have to try it to know. Sorry. And I really don't want to.
OK.........I have heard it.......obviously not on my system. I can't believe it doesn't sound horrible as I feared. It just doesn't sound that good, either.
Jocko |
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| stolbovoy |
| quote: | Originally posted by Vadim
[B]
Unfortunately the aliasing spectra are not ultrasonic at all. Here is the picture; - with no over-sampling the first image of the 20 kHz –wide data will appear at 44.1 kHz, contaminating the data substantially to put it mildly.
Without a proper anti-imaging filter the noise rejection is done by the D/A’s inherent zero-order hold function, that looks like [sin(x)]/x, with x=wT/2, w=2piF and T=period of the 44.1 kHz sinusoid. At 20 kHz this amounts to less then 0.2 dB. That means that the zero-order-hold phenomena only reject 0.2 db of noise at 20 kHz. This is all the noise rejection you will get if no anti-imaging filter is used. Noise city! |
Correct me if I’m wrong please: for input signal with 0-22.05 spectra aliasing spectra will be from 22.05 to 44.1 which is ultrasonic to me. There is no "noise" in 0-22.05 spectra
X = 2*pi*F/Fs =~ 2*3.14 * 20/44.1 =~2.85
Sin(X)/X =~0.1. != -0.2 dB
| quote: |
So, in a nutshell, here you have it, - with no over-sampling you cannot reconstruct the original 16-bit data at all. You may recover about 12-13 bits at the most, although I doubt that even that is possible. |
The ultimate goal is not reconstructing samples, it is reconstructing analog signal on LPF/ADC input, which is not limited by 22.05 kHz. There is always error after Fs/2, no matter will aliasing spectra be cut or not. Because input signal has spectra components above 22.05 kHz, I don’t think that garbage from aliasing is worse then nothing. |
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| Jocko Homo |
Their words.....not mine.
| quote: | | Audio Note™ “magic” I/V transformer interfaces |
Trademark, I'll buy that........anyone can have one........
"Magic" I/V transformer interface...............!?.........give me a break.
Jocko |
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| rfbrw |
| quote: | Originally posted by Vadim
I have no problems with turn-tables. I would however argue that our hobby here is all about sound reproduction and not about sound generation. With this in mind, it should be clear that the goal is to faithfully reproduce what has been recorded on a CD or the vinyl record. The common logic would then dictate that any perceived sound which can be attributed to an electronic component, be it a turn-table or a DAC, - any sound at all, is a problem. Wouldn’t you agree?
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We are getting dangerously close to landing on planet Self where the abiding motto appears to be man in the service of machine. The thing either does what you want it to do or it doesn't and if by violating the rules you get it to do what you want, its only audio and I can live with that. The plain fact is for many that like this nos stuff, the numbers mean nothing. They don't care whether the dac contains a trained gerbil programming a minature cray on the fly, they just like what hear and that will do nicely. So what if I've tried it, didn't like it and certainly wouldn't buy it, I can think far dafter ways to spend one's money.
And having spent time with the kind of people who think the 5532 is too good for audio and 13 bit/32Khz is more than enough i.e. video people, I find the lunacy sort of balances itself out.
Back to reclocking. Using an ASRC when there is no need to change the sample rate strikes me as unnecessary. I can see the ease of use aspect and the seemingly minimal effect on audio quality but I would rather forego another round of processing when I've got an oversampling filter for in or preceding the dac. A PLL or a FIFO is the way I'd go. |
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| Vadim |
| quote: | Originally posted by stolbovoy
Correct me if I’m wrong please: for input signal with 0-22.05 spectra aliasing spectra will be from 22.05 to 44.1 which is ultrasonic to me. There is no "noise" in 0-22.05 spectra |
Well, the input signal should be band limited to 20 kHz. It is done during recording at A/D stage with analog low-pass filter. This filter is not perfect. While Shannon Theorem assumes perfect band-limited signals, the reality is not that. Also, it seems to me that there is very little point talking about 22.05 kHz which is half the sampling rate.
Anyway, the aliasing noise does not just stop at 22.05 kHz border. It continues to propagate into the 20 kHz band while diminishing in amplitude as it gets closer to lower frequencies.
Another point to keep in mind is that the image frequencies, if not properly rejected, will most definitely cause intermodulation products with desired baseband frequency components. This along will result in unacceptable degradation if the output signal.
| quote: | Originally posted by stolbovoy
The ultimate goal is not reconstructing samples, it is reconstructing analog signal on LPF/ADC input, which is not limited by 22.05 kHz. There is always error after Fs/2, no matter will aliasing spectra be cut or not. Because input signal has spectra components above 22.05 kHz, I don’t think that garbage from aliasing is worse then nothing. |
Naturally, we are trying to reconstruct the analog signal, not samples. However, the original data is band limited. Sure there is an error, but if the A/D is done right, and there is no reason to think that it is not, the error you are talking about is negligible. We do get the 16-bits in the end.
The anti-aliasing filter preceding the A/D is extremely important, so I disagree, - it is important if the aliasing spectra is cut or not. If it is not cut, - the resulting digital signal will never represent the analog data to the required 16 bits.
Vadim |
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| Vadim |
| quote: | Originally posted by rfbrw
We are getting dangerously close to landing on planet Self where the abiding motto appears to be man in the service of machine. The thing either does what you want it to do or it doesn't and if by violating the rules you get it to do what you want, its only audio and I can live with that. The plain fact is for many that like this nos stuff, the numbers mean nothing. They don't care whether the dac contains a trained gerbil programming a minature cray on the fly, they just like what hear and that will do nicely. So what if I've tried it, didn't like it and certainly wouldn't buy it, I can think far dafter ways to spend one's money.
And having spent time with the kind of people who think the 5532 is too good for audio and 13 bit/32Khz is more than enough i.e. video people, I find the lunacy sort of balances itself out.
Back to reclocking. Using an ASRC when there is no need to change the sample rate strikes me as unnecessary. I can see the ease of use aspect and the seemingly minimal effect on audio quality but I would rather forego another round of processing when I've got an oversampling filter for in or preceding the dac. A PLL or a FIFO is the way I'd go. |
Yes, planet Self is a place to behold. However, I agree with you, that there is no point insisting on the truth, whatever it may be. We are talking about perception here, and so, as the saying goes, - whatever turns you on…If it is the NOS DAC that does it for you, - great!
I also agree with you that a FIFO or a PLL is another credible approach to jitter reduction. You see, with ASRC, particularly those made by AD, the THD+N is the lowest if the input and output rates are kept the same. It has to do with the samples landing directly on top of the interpolation coefficients.
The problem as I would see it with the PLL is that it is so bloody difficult to make one that actually works well. You will most likely need to have 2 of them and in the end the complexity is not worth it in my opinion when the ASRC is such an elegant and simple solution.
FIFO, is another story. But I think it will get to be just as cumbersome as the PLL deal. Although I admit , I never tried the FIFO approach. |
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| garbage |
Vadim
I would like to seek your opinion on a circuit or possible schematic where I can build a receiver using the AD1895 and also a corresponding dac that would interface easily with it. Or should I just follow the eval board for a start?
This would make an interesting project in future for me to compare how it sounds with the non-os 1545a.
Thanks.
garbage |
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| bocka |
| quote: | | A PLL or a FIFO is the way I'd go. |
I've just measured a CS8416. It has a lot, really a lot of jitter. Far beyond the 200ps which the datasheet refers. Measured somethin in the 5ns range, dependant what (coax) SPDIF source I'm using. Yeah a second PLL and synchronous reclocking is the way to jet rid to that jitter.
| quote: | | The problem as I would see it with the PLL is that it is so bloody difficult to make one that actually works well. |
No, a VCXO with a PLL is not difficult to design, not more than a typical low or high-pass filter. BTW a simulation helps a lot for such a purpose... |
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| Jocko Homo |
That is awful. The much maligned YM3623 only has around 1 nSec. I don't think any of the Crystal RX are far from that. How they managed to make the '8416 that bad is, well...........about what I would expect from those guys.
Jocko |
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| stolbovoy |
| quote: | Originally posted by Vadim
Well, the input signal should be band limited to 20 kHz. |
Is it on the format specs? If some of ADCs have 20kHz LPF - it is design feature, IMHO.
| quote: | | It is done during recording at A/D stage with analog low-pass filter. This filter is not perfect. While Shannon Theorem assumes perfect band-limited signals, the reality is not that. Also, it seems to me that there is very little point talking about 22.05 kHz which is half the sampling rate. |
Isn't it mirroring classic DA conversion: LPF with higher bandwidth + oversampling + DF?
| quote: | | Anyway, the aliasing noise does not just stop at 22.05 kHz border. It continues to propagate into the 20 kHz band while diminishing in amplitude as it gets closer to lower frequencies. |
There is no data in the digital domain about frequencies higher then Fs/2. What make first (or any) aliasing component to go down below 22.05 on DA side?
| quote: | | Another point to keep in mind is that the image frequencies, if not properly rejected, will most definitely cause intermodulation products with desired baseband frequency components. This along will result in unacceptable degradation if the output signal. |
This is another story. Yes, it will cause some additional level of IMD, which depends on analog circuitry.
| quote: | | Naturally, we are trying to reconstruct the analog signal, not samples. However, the original data is band limited. |
Original "data" is a sound of cymbal, flute , etc. It is not band limited. And if you cut part of the original spectra - you just added an error equal to negative inversed A/D LPF frequency response applyed to this original signal which is not band limited.
| quote: | The anti-aliasing filter preceding the A/D is extremely important, so I disagree, - it is important if the aliasing spectra is cut or not. If it is not cut, - the resulting digital signal will never represent the analog data to the required 16 bits.
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Sorry for writing confusing way. I meant DA, not AD. For AD antialiasing filter is a must. For DA I personally concerned about frequency response, not aliasing. |
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| rfbrw |
| quote: | Originally posted by garbage
Vadim
I would like to seek your opinion on a circuit or possible schematic where I can build a receiver using the AD1895 and also a corresponding dac that would interface easily with it. Or should I just follow the eval board for a start?
This would make an interesting project in future for me to compare how it sounds with the non-os 1545a.
Thanks.
garbage |
http://www.diyaudio.de/html/body_dac5.html |
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| garbage |
wow... that's a super neat project!
thanks for the url! |
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| bocka |
| quote: | | That is awful. The much maligned YM3623 only has around 1 nSec. I don't think any of the Crystal RX are far from that. How they managed to make the '8416 that bad is, well...........about what I would expect from those guys. |
I think it's not ony a problem of the chip but also from the incoming SPDIF signal. But as the SPDIF receiver is only able to attenuate jitter above its cutoff frequency of about 30 kHz (I think so, but I've got no datasheet here at home). jitter below 30 kHz is directly fet through the receiver.
On the other side the jitter also changes when switching between the two different phase detectors in the CS8416. This shows that the CS8416 has an influence on the jitter performance. If I'd knew this before I'd better use the AKM SPDIF receiver... |
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| Vadim |
Stolbovoy,
I am sorry for a rather late reply.
| quote: | | Is it on the format specs? If some of ADCs have 20kHz LPF - it is design feature, IMHO. |
No, the filter must be there as a part of an overall circuit.
| quote: | | There is no data in the digital domain about frequencies higher then Fs/2. What make first (or any) aliasing component to go down below 22.05 on DA side? |
Naturally, the digital data as just before it gets to D/A chip only contains information with frequency content up to Fs/2. The problem is not there.
The problem of high transients has to do with the way the D/A works. Look at the simple ladder type D/A. Internally it has current switches that steer the current in to the collection point, which is ultimately the current output from the D/A chip. Those switches, usually made out of FETs, are controlled by the input digital data.
The transient behavior of the switch produces a multitude of various harmonics. When you look at the spectral content of the image, you will see those harmonics extending pretty much everywhere. The anti-imaging LP filter works to reduce that unwanted spectra. Naturally that filter needs space to work.
My point was that a few kHz available, as in the case of non-oversampling DAC, is simply not enough to attenuate the unwanted spectra to the tune of 16-bit precision or about 96 dB. Well, actually a little less if you account for the zero-order hold.
| quote: | | Original "data" is a sound of cymbal, flute , etc. It is not band limited. And if you cut part of the original spectra - you just added an error equal to negative inversed A/D LPF frequency response applyed to this original signal which is not band limited. |
Again, this is not so. The data is band-limited by the A/D converter circuitry. In fact the anti-aliasing filter is the very device the limits the frequency content of the data. I agree that the cymbal, or whatever the tone may be, might extend past the 20 kHz. Naturally we don’t care, - we can’t hear it anyway, unless you are 16 and sitting in the lotus position…Damn, I can’t hear anything past 17 kHz!
| quote: | | Sorry for writing confusing way. I meant DA, not AD. For AD antialiasing filter is a must. For DA I personally concerned about frequency response, not aliasing. |
Its ok, I get confused by all this terminology just as well. Anyway, as I pointed out above, you cannot get away from anti-imaging requirement on the D/A side. It is just not possible to recover the signal that the sound engineer heard during the recording process without it.
Vadim |
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| Vadim |
| quote: | Originally posted by bocka
I've just measured a CS8416. It has a lot, really a lot of jitter. Far beyond the 200ps which the datasheet refers. Measured somethin in the 5ns range, dependant what (coax) SPDIF source I'm using. Yeah a second PLL and synchronous reclocking is the way to jet rid to that jitter.
No, a VCXO with a PLL is not difficult to design, not more than a typical low or high-pass filter. BTW a simulation helps a lot for such a purpose... |
Well, I am extremely skeptical of jitter measurements. I know from experience and many experimental failures this is not a trivial task. Measuring anything down to 100’s of picoseconds range is tough to do in a reliable fashion. Alternatively you might consider the fact that correlated jitter is still a subject to a Fourier analysis and it will show as a THD measurement.
Also my experience shows that even the double PLL design will not do as well as ASRC. So, in the end I would still prefer to go with AD1896 |
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| bocka |
| quote: | | Well, I am extremely skeptical of jitter measurements. I know from experience and many experimental failures this is not a trivial task. Measuring anything down to 100’s of picoseconds range is tough to do in a reliable fashion. Alternatively you might consider the fact that correlated jitter is still a subject to a Fourier analysis and it will show as a THD measurement. |
Agreed that jitter measurement in the 100 (or less) ps range is not trivial. On the other side jitter of some ns is quite easy measurable and its dependant of the CS8416 settings. To obtain a higher resolution than 16 bits for higher frequencies lowest jitter (in the ps range) is a must. This can be done either with a ASRC or a PLL. A PLL is advantageous over a ASRC because it lets the incoming data untouched which a ASRC cannnot by principle. Cutoff frequencies are about the same, some Hz when well implemented. But - and this is very important - the VCXO control voltage must be as clean as possible, less than let's say less than 1mV. This is the real hard job because the incoming data from the PLL is a square / pulse wave. And a simple 74HC4046 as phase detector will definitely not do the job. So it's easier to implement an ASRC than a PLL. |
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| Guido Tent |
| quote: | Originally posted by bocka
the VCXO control voltage must be as clean as possible, less than let's say less than 1mV. This is the real hard job because the incoming data from the PLL is a square / pulse wave. And a simple 74HC4046 as phase detector will definitely not do the job. So it's easier to implement an ASRC than a PLL. |
when stating 1 mV, what BW are you refering to ?
and yes, designing decent PLL's is not easy...... |
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| Guido Tent |
| quote: | Originally posted by bocka
I think it's not ony a problem of the chip but also from the incoming SPDIF signal. But as the SPDIF receiver is only able to attenuate jitter above its cutoff frequency of about 30 kHz (I think so, but I've got no datasheet here at home). jitter below 30 kHz is directly fet through the receiver.
On the other side the jitter also changes when switching between the two different phase detectors in the CS8416. This shows that the CS8416 has an influence on the jitter performance. If I'd knew this before I'd better use the AKM SPDIF receiver... |
It is about 20kHz. So everything you put in, you get out, included data related jitter, that is why you measure 5ns (what BW by the way ?).
therefor:
- The SPDIF jitter should be as low as possible from the beginning
- A cascaded PLL should take care of the rest. It should be slow.
best |
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| stolbovoy |
| quote: | Originally posted by Vadim
>>Is it on the format specs? If some of ADCs have 20kHz LPF - it is design feature, IMHO.
No, the filter must be there as a part of an overall circuit.
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I did not question need of LPF, I just stated that cut off frequency can be anything in 20.0-22.05 range. Anyway it is offtopic.
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The transient behavior of the switch produces a multitude of various harmonics. When you look at the spectral content of the image, you will see those harmonics extending pretty much everywhere. The anti-imaging LP filter works to reduce that unwanted spectra. Naturally that filter needs space to work.
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Instead of zero-order-hold we have a different function. Ideal zero-order-hold has pretty wide spectra of harmonics anyway.
In case this behavior adds some additional nonlinearity, it will manifest itself in 0-22.05 band anyway. I don’t expect significant performance decrease because of it.
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My point was that a few kHz available, as in the case of non-oversampling DAC, is simply not enough to attenuate the unwanted spectra to the tune of 16-bit precision or about 96 dB. Well, actually a little less if you account for the zero-order hold. |
Typical NOS DAC doesn’t have dedicated LPF in analog path.
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>Original "data" is a sound of cymbal, flute, etc. It is not band limited.
>And if you cut part of the original spectra - you just added an error equal to
>negative inversed A/D LPF frequency response applied to this original signal
>which is not band limited.
Again, this is not so. The data is band-limited by the A/D converter circuitry. In fact the anti-aliasing filter is the very device the limits the frequency content of the data. I agree that the cymbal, or whatever the tone may be, might extend past the 20 kHz. Naturally we don’t care, - we can’t hear it anyway, unless you are 16 and sitting in the lotus position…Damn, I can’t hear anything past 17 kHz!
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I thought about sound prior to mike as original "data". Again, I don’t argue that ADC should have LPF with cutoff frequency <= Fs/2. I argue only with taking data stream after ADC as the reference.
As one friend of mine said “I can’t near tones past 18ęÍz, but I can hear absence of them”. Personally I clearly prefer 24/96 to 16/44.1 and probably mostly not because 16 vs 24 part.
| quote: | Its ok, I get confused by all this terminology just as well. Anyway, as I pointed out above, you cannot get away from anti-imaging requirement on the D/A side. It is just not possible to recover the signal that the sound engineer heard during the recording process without it.
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Don’t recover signal, recover sound.
Personally, I’m not NOS admirer, although I listened to Audio Note DACs and found their sound pretty attractive. |
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| bocka |
| quote: | | when stating 1 mV, what BW are you refering to ? |
About 10Hz for the second PLL. The BW for the VCXO is much higher indeed (some 10kHz typical), so noise figures above the cutoff frequency (where ever they induced) results in higher jitter.
| quote: | | So everything you put in, you get out, included data related jitter, that is why you measure 5ns |
Yes, it's my transport (a not so cheap Perreaux CDP1 and a low cost Mustek DVD Player both SPDIF coax), another transport (Denon MD-Player TOSLINK) showed much lower jitter. |
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| jwb |
| quote: | Originally posted by Jocko Homo
There are twisted pair cables designed for data transmission, that would work well if the proper connector was used. But sourcing them will be a problem |
That's an odd claim. The whole planet is overrun with a perfectly suitable cable: category 5 (5e, 6) unshielded twisted pair wire with modular plugs. Operates perfectly well at 350MHz with suitable cross-talk and reflection properties even at 100 meters. Pennies per foot. |
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| Jocko Homo |
Yes, of course it works. But you should know that jitter affects sound more than it affects data.
The cables that I am refering to were designed for harsher environments. All we had back then, but they were rugged, and worked. And I would bet better than the CAT5 cable used everywhere now.
Also............we used lots of pre-equalisation on twisted pairs back then. Not sure what the FCC would have to say about that without a lot of screening. The cables we used did have such measures.
Jocko |
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