[jkeny]

    Why do you think that this is not a valid engineering point? The DAC takes bits and converts them. If it gets the *right* bits then altering the shape of the bits can make no difference. It's either the right bits or the wrong bits, period, assuming the rest of the DAC is designed properly. It's not a matter of opinion, it's inescapable fact.

     Brett

OK, let me spell it out for you - the bits need to arrive at exactly the right time to recreate the original waveform. This is the piece that you "bits are bits" guys are missing.

A waveform is recorded with a clock that allows the wave to be split into 44100 slices per second with the exact same interval between each slice. SO what do you think happens if the interval changes between a number of slices when you are trying to recreate the original waveform?

[srb]

OK, let me spell it out for you - the bits need to arrive at exactly the right time to recreate the original waveform. This is the piece that you "bits are bits" guys are missing.

I don't understand why this concept would not be universally acknowledged by all electrical engineers.
 
The concept of reducing jitter (timing) is of high importance to major DAC chip manufacturers like Analog Devices, Crystal, ESS, Texas Instruments and Wolfson.  The effects of signal reflections in digital transmission lines (timing) is of equally high importance to companies like AT&T, Motorola and Hewlett Packard.
 
If these physical phenomena don't exist, then there would be absolutely zero differences between the AES/EBU, S/PDIF coaxial, S/PDIF optical or I2S interfaces as well as zero differences between cable types and lengths.
 
Are all of these companies part of a conspiracy to hoodwink audio consumers and equipment manufacturers?
 
Steve

[festuss]

Well, you might have read but you didn't understand. What does "changing the shape of bits" mean? You are still clinging onto the bits errors as the only possible problem. Here's a hint - I gave it already - digital is analogue with only two voltage levels. Ask yourself how digital works - how a bit is sensed?

Now the same question as I asked Dr. festuss - what is the other element besides bits that you need for digital audio?

What other element do I need for digital audio? A power supply? Since there is no bits, if there is no supply to have change.  But does that supply need better sounding capacitors?    Your question is rhetorical? I ain't got no PhD, I am hardly at that level, just a grunt, but not a gullible  grunt. Ya need an AD converter in order to be able to use a DA converter.  Ya need Ying in order to have Yang. There is no Costello without Abbot. There is no Dumber without Dumb.

[festuss]

OK, let me spell it out for you - the bits need to arrive at exactly the right time to recreate the original waveform. This is the piece that you "bits are bits" guys are missing.

A waveform is recorded with a clock that allows the wave to be split into 44100 slices per second with the exact same interval between each slice. SO what do you think happens if the interval changes between a number of slices when you are trying to recreate the original waveform?

Error correction ckts are why your claims are nebulous and irrelevant.  Timing is clocked quite well, in the ckts.  Rubidium and Quartz solve that, next boogie man in your imaginary ckt issues? How does HDTV or planes stay up?  How do automobile controls function so well, using digital ckts in all that engine management and brake controllers, and even electric steering assys?  The magic of digital and pretty inexpensive digital components.  Just think how pricey and impractical if the stuff was analog? Timing control, is pretty well perfected, if you want to keep hunting for an imaginary boogieman in your audio ckts, keep on hunting. It's like religion, everybody has their own superstitions of non reality based issues.

[festuss]

I don't understand why this concept would not be universally acknowledged by all electrical engineers.
 
The concept of reducing jitter (timing) is of high importance to major DAC chip manufacturers like Analog Devices, Crystal, ESS, Texas Instruments and Wolfson.  The effects of signal reflections in digital transmission lines (timing) is of equally high importance to companies like AT&T, Motorola and Hewlett Packard.
 
If these physical phenomena don't exist, then there would be absolutely zero differences between the AES/EBU, S/PDIF coaxial, S/PDIF optical or I2S interfaces as well as zero differences between cable types and lengths.
 
Are all of these companies part of a conspiracy to hoodwink audio consumers and equipment manufacturers?
 
Steve

I'm sure the nice full page fancy ads in many audiophile magazines have learned you all about how a magic wire will transform the sound of your digital ckts.  If your coax doesn't have a magic name, or mineral, or planetary nomenclature it doesn't sound right. Only in audio, at the audiophile level is the BS so rampant, but the purveyors of mystical BS cures, for non existant issues have moved into video, with magic cures like special USB or HDMI that will look better than mere stock HDMI cables.  The reality based mfg. have components made to the standards of the product lines, so the stuff works quite well if not perfect, using stock non defective parts. Audiophiles, don't exist on a plane of reality.  How does your CD enhancing fluid, green pens, demagnetized records,demagnetized cables sound?  How does my GPS work so well,yet a CD playback system is filled with issues? Cured only with mystical cures?

[festuss]

  Looks like THIS is the cure all cable that kills the boogieman, according to this magic supplier.  www.   blackcatcable.net/veloce.php  Heck the boogieman keeps the U.S. military always employed.

[jkeny]

I don't understand why this concept would not be universally acknowledged by all electrical engineers.
 
The concept of reducing jitter (timing) is of high importance to major DAC chip manufacturers like Analog Devices, Crystal, ESS, Texas Instruments and Wolfson.  The effects of signal reflections in digital transmission lines (timing) is of equally high importance to companies like AT&T, Motorola and Hewlett Packard.
 
If these physical phenomena don't exist, then there would be absolutely zero differences between the AES/EBU, S/PDIF coaxial, S/PDIF optical or I2S interfaces as well as zero differences between cable types and lengths.
 
Are all of these companies part of a conspiracy to hoodwink audio consumers and equipment manufacturers?
 
Steve

I haven't a clue what are you saying? You seem to be all over the place but not as confused as Dr. festuss who is resorting to spouting mantras!

I've explained why timing is the critical factor in digital audio that you "bits is bits" guys always overlook. You seem to think that because digital works in computers & computers are so ubiquitous, that it's all just bits. In the main computers work asynchronously & there is no requirement for real-time operation as we have in audio i.e. the correct bits have to arrive at the correct time in order to recreate the original waveform exactly. Hell, there are special real-time operating system written to try to deal with some of these issues!

Dr. Festuss knows so little that he talks about using Rhubidium clocks because he picked this up somewhere - complete waste of time using a $4,000 Rhubidium clock to time digital audio. I won't even bother answering his inanities.       

[festuss]

so now a pricey audiophile Rubidium clock is a waste of time in order to get all the bits to line up at the correct time, but so is your $20 usless "tweak" I use neither, but I am sure the ones with above mortal ears have to have Rubidium in their clocks.  Audiophiles want the finest.  And they plate their 50 cent connectors with magic plating, and then have the honor of paying a scam company hundreds of dollars, and it always improves things, always.  You obviously have ears that are beyond mortals, as you can here timing errors that don't exist, let alone audible.  Of course you car digital controllers are not operating in real time, it's all just an audio issue where timing is critical.  Clueless is, as clueless wants to be.  What time are you operating at?  Certainly not any time zone on this planet.  Timing is not an issue in digital controllers, for medical equipment, digital displays, industrial controllers, robotics, only AUDIO, where the trained noodnick can always know when it's not right, cus they can hear things that the boogieman brings, and the marketing companies have educated you real well.  No real time controllers?  Only in audio functions, how dumb are you? Tests have also proved that the human hearing is full of issues and the human brain is easily fooled.  Tweak on, keep hunting the boogieman

[jkeny]

Sorry Festuss but you have demonstrated what a fool you are! You have ignorantly lumped all "tweaks" into the same pot which demonstrates your inability to deal with any level of research or comprehension. Your claim of knowledge in these areas is a laughable, pathetic attempt to pump up your image but all you've done is make a fool of yourself. Enjoy your ignorance - I for one am not going to waste any more time on you.

Who said this thread was polite  

[avahifi]

So, jkeny, please tell me, pray tell, which attenuator to use to improve all my DACs, not just make some better and some worse.

Or, in other words, which handrail on the Titanic needs the most polish to keep the dumb thing from sinking?

Sugar pills abound, wrapped in all kinds of technical double-speak, and lots of lots of hundred dollar bills.

Regards,

Frank Van Alstine

[jkeny]

So, jkeny, please tell me, pray tell, which attenuator to use to improve all my DACs, not just make some better and some worse.

Or, in other words, which handrail on the Titanic needs the most polish to keep the dumb thing from sinking?

Sugar pills abound, wrapped in all kinds of technical double-speak, and lots of lots of hundred dollar bills.

Regards,

Frank Van Alstine

Frank, there is no answer to your question - which value to use - it depends on a lot of factors. But if you have an open mind you can test their effectiveness yourself - they cost ~$20 each (not a $4,000 Rhubidium clock!) from minicircuits (I don't sell them, btw ) - try a 6 & 10dB from here http://www.minicircuits.com/products/attenuators_coax_fixed.html - the 75ohm versions bottom of the page. And report your findings with a post back here!

You asked me for the engineering behind them - I gave it to you. So now I'm asking you to try them & report your findings here. You may find the exercise interesting?

[srb]

I haven't a clue what are you saying? You seem to be all over the place but not as confused as Dr. festuss who is resorting to spouting mantras!

I don't know why you are quoting me then responding with this?  I am not "all over the place" nor am I a "bits is bits" guy.  I merely agreed with you and acknowledged the existence of the importance of timing when transferring digital audio signals, supported by every company that actually produces the technology.
 
I think you need to read the responses more carefully before you inappropriately place me in the festuss camp, which does not acknowledge or believe in it.
 
Steve

[jkeny]

Sorry, when being attacked for suggesting something as simple & as cheap as an RF attenuator, I can read things the wrong way unless they are very clear!

I see now that your post was very clear but at the time I read it I couldn't make out if you were saying it was a good idea or not & therefore thought you were all over the place! Apologies again!

[Brett Buck]

OK, let me spell it out for you - the bits need to arrive at exactly the right time to recreate the original waveform. This is the piece that you "bits are bits" guys are missing.

A waveform is recorded with a clock that allows the wave to be split into 44100 slices per second with the exact same interval between each slice. SO what do you think happens if the interval changes between a number of slices when you are trying to recreate the original waveform?

   I grasp that, but why does it matter, since you need to resync the inputs locally anyway? You wouldn't want to count on the time of arrival of the input data, you need to only transfer the data word to the converter when the time for the next word comes up.

And, I might add, you would want to square up the waveform locally, too.  So all the "analog signal" you tout needs to do is fall in the threshold of a "1" or a "0" to the data receiver. So it can be incredibly ratty, ringing, etc, and still have no effect - zero- on the integrity of the signal seen by the D/A. That's *the whole point* of having digital data transfers and why it is better and less prone to interference and errors. And there is error correction, so if it does happen, it gets straightened out before it ever gets shipped downstream.

     You can't tell the effects of any of this by looking at the raw stream from the transport or digital source. The raw signal can look like hell and still be fine once it gets to the D/A itself.

     And, I might add, gently - I don't think any of us appreciates the "you guys are all idiots, I have secret information, here are some hints" tone on this. I almost certainly have more relevant experience with similar systems than you do, and so do many others here. Don't assume we are all morons.

      Brett

[jkeny]

   I grasp that, but why does it matter, since you need to resync the inputs locally anyway? You wouldn't want to count on the time of arrival of the input data, you need to only transfer the data word to the converter when the time for the next word comes up.

Ok, so you know about timing errors, jitter and you know that the timing is only important at 2 places A/D & D/A. The waveforms can be s**t as you say, all along the signal chain, as long as the error correction is working. But now you ar etalking about resynching the signal locally - what I'm talking about & where the RF attenuators seem to matter most is in use where a clock is recovered from the SPDIF signal at the PLL before the DAC. If you are not doing this it will theoretically not have any effect. Maybe this is why the Vision benefits & the Vision+ doesn't? But asynchronous resampling seems to introduce other problems. 

And, I might add, you would want to square up the waveform locally, too.  So all the "analog signal" you tout needs to do is fall in the threshold of a "1" or a "0" to the data receiver. So it can be incredibly ratty, ringing, etc, and still have no effect - zero- on the integrity of the signal seen by the D/A. That's *the whole point* of having digital data transfers and why it is better and less prone to interference and errors. And there is error correction, so if it does happen, it gets straightened out before it ever gets shipped downstream.

     You can't tell the effects of any of this by looking at the raw stream from the transport or digital source. The raw signal can look like hell and still be fine once it gets to the D/A itself.

Probably so, if you are re-clocking the SPDIF stream

And, I might add, gently - I don't think any of us appreciates the "you guys are all idiots, I have secret information, here are some hints" tone on this. I almost certainly have more relevant experience with similar systems than you do, and so do many others here. Don't assume we are all morons.

      Brett

Brett, by your previous answer you seemed to be saying that timing was not important & it was just bits so forgive if I came across as you say but I'm retiring from this thread now.

The RF attenuators are cheap & worth experimenting with as lots of people have already testified to on various forums - they can't all be wrong! As to the exact principles of their operation - I gave some guesses but I'll throw out another one:
- it may also have to do with cutting down RF noise incursion?

Regards, John   

[WGH]

try a 6 & 10dB from here http://www.minicircuits.com/products/attenuators_coax_fixed.html - the 75ohm versions bottom of the page.

Note that the Minicircuits 75 ohm attenuators have a BNC connector. One place I found that sells the 75 ohm BNC to RCA adapter is B&H Photo for $2.95 each, they sell single items if you only need one of each.

http://www.bhphotovideo.com/c/product/534347-REG/Bolide_Technology_Group_BP0006_BP0006_RCA_Male_to.html


http://www.bhphotovideo.com/c/product/534348-REG/Bolide_Technology_Group_BP0023_BP0023_RCA_Female_to.html




Or get the ARM Electronics adapters with copper contacts for $1.00 more
http://www.bhphotovideo.com/c/search?Ntt=rca+bnc+adapters&N=0&InitialSearch=yes

Best Tweek of the Year: Rooting my Nook Color, now I can read Tone Audio, surf the net, and a whole lot of other stuff while listening to music, it's a small book that does everything.

Wayne

[Brett Buck]

Ok, so you know about timing errors, jitter and you know that the timing is only important at 2 places A/D & D/A. The waveforms can be s**t as you say, all along the signal chain, as long as the error correction is working. But now you ar etalking about resynching the signal locally - what I'm talking about & where the RF attenuators seem to matter most is in use where a clock is recovered from the SPDIF signal at the PLL before the DAC. If you are not doing this it will theoretically not have any effect. Maybe this is why the Vision benefits & the Vision+ doesn't? But asynchronous resampling seems to introduce other problems. 
Probably so, if you are re-clocking the SPDIF stream

     This is a feature of essentially all workable streaming data schemes. Usually, you detect the incoming clock rate, filter it heavily so that you generate a second local clock with essentially no jitter but at the same average data rate - and shifted from the input far enough to cover the worse-case jitter. The bit rate can vary slowly, or be off the standard, but the jitter is essentially non-existent.  The data words come in, they are received, and held until the next local clock rising or falling edge, then transferred to the D/A converter. So, essentially no effects of jitter (only the jitter in the local clock, which can be made almost arbitrarily small) regardless of the incoming bit or word rate. If for some reason a word comes in too late, then you just miss it, but that's effectively a failure.

   Alternatively, you can generate your own local oscillator at desired rate, independent of the incoming stream. That gives perfect control over the bit rate but can be prone to the working its way in and out of sync if either the source or sink is not exactly right.

    So, in a sense, yes, I am saying jitter is not an issue, because the data receiver and transfer to the DA eliminates it in all but the very crudest systems. But even a Discman has a buffering scheme to eliminate skips that, coincidentally or not, also would eliminate the data jitter arising from the reading of the bits from the disc. Correctly designed, the local oscillator will not respond to the super-high-frequency reflections, ringing, etc, because they will only rarely cause a bit transition, and the rate detection filter will be ~5 orders of magnitudes slower.

     BTW I have absolutely no Insight (heh!) into the guts of how Frank's DACs work but this is so fundamental that I am sure that this problem is covered.

    Brett

[avahifi]

The most hopelessly unreliable pieces of my lab testing setup are those BNC to RCA adaptors.  Nothing I do makes them work reliably.  So I finally gave up and ordered a set of cables from Blue Jeans, with RCAs on one end and BNCs on the other to interface my scope directly to the equipmeht under test.  The scope needs the BNCs, no other hardware available. Problems solved.

So --- I hardly am going to try attenuators that require also stuffing a BNC to RCA adaptor back into the digital signal path too.  If you want a bad signal, this would be the sure way to get it.

Regards,

Frank Van Alstine