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Reference DAC Module - Discrete R-2R Sign Magnitude 24 bit 384 KHz

"CONCLUSION

In order to determine the maximum acceptable size of jitter on music signals, detection thresholds for artificial random jitter were measured in a 2 alternative forced choice procedure. Audio professionals and semi-professio-nals participated in the experiments. They were allowed touse their own listening environments and their favoritesound materials. The results indicate that the threshold for random jitter on program materials is several hundreds ns for well-trained listeners under their preferable listening conditions. The threshold values seem to be sufficiently larger than the jitter actually observed in various consumer products
"

Whats wrong here?

(PDF) Detection threshold for distortions due to jitter on digital audio

//

Nothing wrong. But the text you quote is about random jitter noise. The type of noise discussed currently in this thread is of periodic nature.
 
Andrea, I think by now everybody know your your opinion, and that your pushes it at any opportunity you get, apparently not for commercial gain, more like a religious belief that you view as the only right one and that you want to have everybody to believe in....

Even the best SPDIF receiver chips have something like 50 pS of random jitter, most consider that to be fine enough for the chips to be used in older very good DACs without further jitter reduction. I remember reading somewhere that R-2R DACS are not as sensitive to jitter as DS DACS, where jitter can be amplified in their DS modulators.... And you do know how much wow and flutter a record player have ?

So I'm happy using the Si514 in my R-2R DACs, it's a fine clock generator with much less jitter than the SPDIF receiver chips. If you really want to compare, the dam1121-02 use the Si514, while the dam1121-01 use the Si570, otherwise they're the same clock design. Get one of each and you can easily do an ABX listening test, or do your measements. Real ABX listening test are the way to go, as there are a tendency for people to believe that their modifications are better....

Remember long time ago, when absolute phase was discussed here ? One person were absolute sure he could hear the effects of the dam1021's "mode = normal | invert" command, when in fact the command only take effect after power cycling the dam1021....
 
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Just for a little distraction, and of course to console my own curiosity (which is of a far, far lower level [emoji18]):

Feel the same here as sometimes the level here is way beyond my understanding but still I find it fascinating and nice to read

The DACs from the 19xx I think are the easiest from Soekris DAC to assemble, the dam1921 just need 3 power supplies and you are ready to go. You could assemble the power supply psu1951 that Søren sell as accessory ;-)
 
@Andrea,

Of course you're entitled to your opinion, however that doesn't include posting claims about products without providing factual data to back up those claims, as then it quickly becomes slander, and slanderous posts are just something we don't want or need.

If I were you I'd provide irrefutable measurement data for your claims as then the discussion becomes technical, and unlike opinions, technical discussions are based on facts and thus can be checked and verified.

The data are on the Silabs datasheet: jitter < 1 ps with an integration bandwidth from 12 kHz to 20 MHz.
No needs of further measurements since the manufacturer has already published them.
Although I will measure the phase noise (or the jitter if you prefere) of the master clock and the latch clock of the DAC and then I will publish the results.
I only have to get the DAM1021 I bought running, so I need at least a transformer and so on.

Then, does the close in phase noise of the master clock affect the SQ?
Here we are in the field of the listening impressions since there is no way to measure the influence.

I think so, you think no.
Different opinions.
Where is the problem?
 
Just for a little distraction, and of course to console my own curiosity (which is of a far, far lower level [emoji18]):
I‘ve been lurking around quite some time now, and have visited soekris site, too, and still am a little overwhelmed...
Are the DACs from the oem-line available to the diy-community, and would it be reasonable to build a single one 1921? Or are there too many constraints—for a unextperienced builder that I am?

I sell everything to everybody, the dam19x1 is just geared towards OEMs, maybe I should put all the boards together in one "boards" section....

The only problems is that the dam1921 is out of stock, and I'm about to run out of the dam1941 too.... I will make more, but don't know when exactly....
 
"CONCLUSION

In order to determine the maximum acceptable size of jitter on music signals, detection thresholds for artificial random jitter were measured in a 2 alternative forced choice procedure. Audio professionals and semi-professio-nals participated in the experiments. They were allowed touse their own listening environments and their favoritesound materials. The results indicate that the threshold for random jitter on program materials is several hundreds ns for well-trained listeners under their preferable listening conditions. The threshold values seem to be sufficiently larger than the jitter actually observed in various consumer products
"

Whats wrong here?

(PDF) Detection threshold for distortions due to jitter on digital audio

Pretty much everything is wrong. What type of DAC chip did they use? It matters because different chips have different jitter sensitivities. How did they recover a master clock signal from AES? What is the jitter performance of the clock recovery system? How did they determine that ground noise from the PC didn't affect the minimum jitter possible from the system? And so on...

The stated conclusions are also completely statistically invalid. There were far too few test subjects for the law of large numbers to apply.

Overall, its a mess that should never have been published.
 
Andrea, I think by now everybody know your your opinion, and that your pushes it at any opportunity you get, apparently not for commercial gain, more like a religious belief that you view as the only right one and that you want to have everybody to believe in....

Even the best SPDIF receiver chips have something like 50 pS of random jitter, most consider that to be fine enough for the chips to be used in older very good DACs without further jitter reduction. I remember reading somewhere that R-2R DACS are not as sensitive to jitter as DS DACS, where jitter can be amplified in their DS modulators.... And you do know how much wow and flutter a record player have ?

So I'm happy using the Si514 in my R-2R DACs, it's a fine clock generator with much less jitter than the SPDIF receiver chips. If you really want to compare, the dam1121-02 use the Si514, while the dam1121-01 use the Si570, otherwise they're the same clock design. Get one of each and you can easily do an ABX listening test, or do your measements. Real ABX listening test are the way to go, as there are a tendency for people to believe that their modifications are better....

Remember long time ago, when absolute phase was discussed here ? One person were absolute sure he could hear the effects of the dam1021's "mode = normal | invert" command, when in fact the command only take effect after power cycling the dam1021....

Soeren,

mine is not a religious belief, the fundamentalists are others than me.

My statements are based on empirical experience (although many members have experienced the same).

I can assure you that a R2R DAC is heavily affected by the quality of the latch clock, I have directly experienced with the old TDA1541A (old but still to beat about the sonic results) with listening comparison replacing the master clock only.
I have made the comparison simply replacing the oscillators of Ian's FIFO buffer, the remaining audio chain was always the same during the comparison.

Every time I replaced the master clock the differences were huge and clearly audible, even by non audio professional people.

I have tested the original oscillators supplied with the board, the Crystek and finally I have installed true low phase noise oscillators.
Every time the quality of the oscillator was improved even the sound quality has been improved.
More details, wider stage, better location of the instruments on the stage, but what was really impressive was the human voices. They became much more natural, no more harshness, no more annoyance after a few minutes of listening.

Following these empirical tests, my conclusion could only be one: the quality of the master clock in digital to analog conversion is decisive and essential.

So, what is the difference between the clocks I have tested?
The answer is clear from the datasheet and the measurements: the close in phase noise of the oscillator.

Any oscillator on the market, even the worst, has a noise floor from -150 to -170 dBc (the Crystek has a noise floor around -170 dBc), hence the only difference between the tested oscillators is the close in phase noise, at 20-30 Hz from the carrier and below.
In this region the difference is huge, up to 40-50 dBc.

In conclusion, I have no way to mathematically demonstrate the results, although I believe it's related to our brain sensitivity to the timing errors, but it was clearly audible.

That's the reason why I invite everyone to make the empirical comparison by themselves.
 
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Pretty much everything is wrong. What type of DAC chip did they use? It matters because different chips have different jitter sensitivities. How did they recover a master clock signal from AES? What is the jitter performance of the clock recovery system? How did they determine that ground noise from the PC didn't affect the minimum jitter possible from the system? And so on...

The stated conclusions are also completely statistically invalid. There were far too few test subjects for the law of large numbers to apply.

Overall, its a mess that should never have been published.

Okey - but then we can surely say that sitting alone at home is an even worse statistically invalid situation ;-D

Joke aside, is it even possible to evaluate this rigorously given all the variables?


//
 
Remember long time ago, when absolute phase was discussed here ? One person were absolute sure he could hear the effects of the dam1021's "mode = normal | invert" command, when in fact the command only take effect after power cycling the dam1021....

Actually, absolute phase (= polarity) is absolutely audible.

It was named "The Wood Effect" and the eponymously named book from 1988 explains it as follows:

Masked by random combination with other distortions in the music reproduction chain, an unsuspected major contributor has lain hidden: Aural sensitivity to "phase inversion", the Wood effect.

Musical instruments normally create compression waves in their attack transients. Electronics, however, often invert that natural, positive polarity to negative, unnatural rarefaction as emitted from loudspeakers, thus diminishing physical and aesthetic impact. The term Absolute Polarity uniquely describes the correct arrival to the ear of acoustic wavefronts from loudspeakers, with respect to actual musical instruments.

Wrong polarity, when isolated, is obvious to almost everyone. Its present neglect results primarily from habitual disregard for linear phase response in loudspeakers, due largely to the erroneous auditory theory of Helmholtz.​


In the mixing studio I first go through a newly received multitrack and find the correct (or at least best sounding) polarity for each track. Don't expect the recording engineer to get it right. It really is clearly audible, especially on kickdrums, snares, percussive bass lines, male vocals and asymmetric brass/synth waveforms. And I might still opt to flip the phase on certain tracks later on, because the different sound from the polarity inversion works better in context with the rest of the mix.
 
Actually, absolute phase (= polarity) is absolutely audible.

It was named "The Wood Effect" and the eponymously named book from 1988 explains it as follows:

Masked by random combination with other distortions in the music reproduction chain, an unsuspected major contributor has lain hidden: Aural sensitivity to "phase inversion", the Wood effect.

Musical instruments normally create compression waves in their attack transients. Electronics, however, often invert that natural, positive polarity to negative, unnatural rarefaction as emitted from loudspeakers, thus diminishing physical and aesthetic impact. The term Absolute Polarity uniquely describes the correct arrival to the ear of acoustic wavefronts from loudspeakers, with respect to actual musical instruments.

Wrong polarity, when isolated, is obvious to almost everyone. Its present neglect results primarily from habitual disregard for linear phase response in loudspeakers, due largely to the erroneous auditory theory of Helmholtz.​


In the mixing studio I first go through a newly received multitrack and find the correct (or at least best sounding) polarity for each track. Don't expect the recording engineer to get it right. It really is clearly audible, especially on kickdrums, snares, percussive bass lines, male vocals and asymmetric brass/synth waveforms. And I might still opt to flip the phase on certain tracks later on, because the different sound from the polarity inversion works better in context with the rest of the mix.

I don't doubt that some can hear absolute phase. My point was that most people want to hear something when they: either spend a lot of money or spend a lot of time building or modifying something. Not that there necessarily is something to hear....