The Digital Promised Land-DSD

o0OBillO0o said:
Paul, cool. I thought DSD is DAC benign, be it single, double or quad rate DSD- meaning the example DAC you mention is your opinion.

Further, plug in the sample rate for each DSD sample rate, ~2.8,5.6 &11.2, and give us a Y value so we can complete your calculations. It's not 100%, but very close. Analog is defined as analogous to the real thing, not Analog Tape/records.

For single rate:
In the graphic below there is a 1/42822400 s or 0.00000002335226 s, if that is sufficient for Y, then:
(please excuse my significant digits)

X = 1- (0.00000002335226 * (2,822,400/1)
X = 1- (0.065909418624)
X = 0.934091 or 93%

PCM_v_DSD_last_sharp-e1403099562269.jpg


(for double or quad multiply Y and S by 4 and 2, respectively.)

I'm pro DSD and pro PCM, But I really want better mixing and mastering. I also want honesty or transparency in provenance.

Pop/rock in Quad DSD should be excellent, as Rob Gordon, a Character in the book&movie, "High Fidelity " said, " Did I listen to pop music because I was miserable? Or was I miserable because I listen to pop music? "

I'm watching for releases in Quad DSD,more reviews of the material, and more Quad rate DSD DACs.
Click to expand...

I wouldn't even call it opinion; calling it my conjecture is probably better suited.

Your graphics make part of my point. The second part is that DSD that goes through a converter, be it a one bit (Meitner, PBD etc.) or a multi-bit never gets put back together in the time domain like it does by staying in native DSD like Ted and Lukasz have started doing. I believe this is the magic. To that end, I wouldn't be surprised if the PS Audio DS would sound closer to a Lampi if it used dht's and didn't do so much up/down sampling of the signal.

As I have said elsewhere, IMO the future is native dsd playback (with filters only) using on the fly conversion of all formats conveyed to the dac through i2s (avoiding time domain imperfections from converters and all the usb/packet noise, regen, blah, blah problems). It's a commin.

(Caution: All complete conjecture on my part.)
 
The good news is that many of us agree that DSD sounds better than PCM. But still, arguments will rage on by those that love PCM and like all things in audio, we will never reach consensus on anything unless its agreeing that we will never reach consensus on anything. :)

I felt compelled to use the smiley face by the way.
 
FlexibleAudio said:
I wouldn't even call it opinion; calling it my conjecture is probably better suited.

Your graphics make part of my point. The second part is that DSD that goes through a converter, be it a one bit (Meitner, PBD etc.) or a multi-bit never gets put back together in the time domain like it does by staying in native DSD like Ted and Lukasz have started doing. I believe this is the magic. To that end, I wouldn't be surprised if the PS Audio DS would sound closer to a Lampi if it used dht's and didn't do so much up/down sampling of the signal.

As I have said elsewhere, IMO the future is native dsd playback (with filters only) using on the fly conversion of all formats conveyed to the dac through i2s (avoiding time domain imperfections from converters and all the usb/packet noise, regen, blah, blah problems). It's a commin.

(Caution: All complete conjecture on my part.)
Click to expand...

Paul- I made some edits to my original response to you:

o0OBillO0o said:
For single rate (2,822,400 Hz):
In the graphic below there is a 1/42822400 s or 0.00000002335226 s, if that is sufficient for Y, then:
(please excuse my significant digits)

X = 1- (0.00000002335226 s * (2,822,400 samples /1 s))
X = 1- (0.065909)
X = 0.934091

I'm confused about the units? What is X's units if seconds is canceled out due to the seconds being in the numerator and denominator? You list X's units as seconds?
Click to expand...

Also check on Antelope Audio's page (never heard their products, found them on the Google). Direct Stream Digital (DSD) ? the technology explained | Antelope Audio Blog

Antelope addresses some of the benefits then quickly goes into "why our product is better.." blah, blah, but aligns with the concepts that HQ player and many other design incorporate. Specifically, the up-conversion, only in the case of DSD- not PCM, has potential benefits by pushing all the utra sonic energy way, way up there and using analog filters that continue to reduce that energy, yet none of the drawbacks of what many claim, today, of DSD high frequency filters.
 
FlexibleAudio said:
I wouldn't even call it opinion; calling it my conjecture is probably better suited.

Your graphics make part of my point. The second part is that DSD that goes through a converter, be it a one bit (Meitner, PBD etc.) or a multi-bit never gets put back together in the time domain like it does by staying in native DSD like Ted and Lukasz have started doing. I believe this is the magic. To that end, I wouldn't be surprised if the PS Audio DS would sound closer to a Lampi if it used dht's and didn't do so much up/down sampling of the signal.

As I have said elsewhere, IMO the future is native dsd playback (with filters only) using on the fly conversion of all formats conveyed to the dac through i2s (avoiding time domain imperfections from converters and all the usb/packet noise, regen, blah, blah problems). It's a commin.

(Caution: All complete conjecture on my part.)
Click to expand...
Reasonable conjecture, but then why are you not all over this as a complement to your main argument of TIME DOMAIN?
https://www.youtube.com/watch?v=tjF2O9lbddg
 
o0OBillO0o said:
Paul, cool. I thought DSD is DAC benign, be it single, double or quad rate DSD- meaning the example DAC you mention is your opinion.

Further, plug in the sample rate for each DSD sample rate, ~2.8,5.6 &11.2, and give us a Y value so we can complete your calculations.
My opinion the result is not 100%, but very close. Analog is defined as analogous to the real thing, not Analog Tape/records.

For single rate (2,822,400 Hz):
In the graphic below there is a 1/42822400 s or 0.00000002335226 s, if that is sufficient for Y, then:
(please excuse my significant digits)

X = 1- (0.00000002335226 s * (2,822,400 samples /1 s))
X = 1- (0.065909)
X = 0.934091

I'm confused about the units? What is X's units if seconds is canceled out due to the seconds being in the numerator and denominator? You list X's units as seconds?

PCM_v_DSD_last_sharp-e1403099562269.jpg


(for double or quad multiply Y and S by 2 and 4, respectively.)

I'm pro DSD and pro PCM, But I really want better mixing and mastering. I also want honesty or transparency in provenance.

Pop/rock in Quad DSD should be excellent, as Rob Gordon, a Character in the book&movie, "High Fidelity " said, " Did I listen to pop music because I was miserable? Or was I miserable because I listen to pop music? "

I'm watching for releases in Quad DSD,more reviews of the material, and more Quad rate DSD DACs.
Click to expand...


I did not make the units clear in my example because I used a discrete time period which in this case (for example) was a 1 second.

X(secs) = 1 (sec) - [Y (sec) x S (1/sec) x 1 (sec)]

X secs = 1 sec - portion of that second used in sample collection
 
o0OBillO0o said:
Paul- I made some edits to my original response to you:



Also check on Antelope Audio's page (never heard their products, found them on the Google). Direct Stream Digital (DSD) ? the technology explained | Antelope Audio Blog

Antelope addresses some of the benefits then quickly goes into "why our product is better.." blah, blah, but aligns with the concepts that HQ player and many other design incorporate. Specifically, the up-conversion, only in the case of DSD- not PCM, has potential benefits by pushing all the utra sonic energy way, way up there and using analog filters that continue to reduce that energy, yet none of the drawbacks of what many claim, today, of DSD high frequency filters.
Click to expand...


IMO, this is the future of digital as long as it includes filters only (i.e., no converters to screw with time domain).
 
FlexibleAudio said:
I did not make the units clear in my example because I used a discrete time period which in this case (for example) was a 1 second.

X(secs) = 1 (sec) - [Y (sec) x S (1/sec) x 1 (sec)]

X secs = 1 sec - portion of that second used in sample collection
Click to expand...

X(secs) = 1 sec - [0.00000002335226 sec x (2,822,400 samples/sec) x 1 (sec)]

X (secs) = 1 sec - [0.06590941862 samples x 1 sec] (the Y's units second cancel out when divided by S's denominator, then we can multiply units)


Okay I made it to here, what do I do now?

If I continue I would be mixing up units..

X (secs) = 1 sec - [0.06590941862 samples * 1 sec]
 
o0OBillO0o said:
X(secs) = 1 sec - [0.00000002335226 sec x (2,822,400 samples/sec) x 1 (sec)]

X (secs) = 1 sec - [0.06590941862 samples x 1 sec] (the Y's units second cancel out when divided by S's denominator, then we can multiply units)


Okay I made it to here, what do I do now?

If I continue I would be mixing up units..

X (secs) = 1 sec - [0.06590941862 samples * 1 sec]
Click to expand...


S units are 1/s not samples/seconds. Think about it more abstractly and it is clearer to see the units fall into place => in a given period of time, a portion of that period is used providing musical information and a portion is used in doing nothing. The existence of the latter is what separates digital from analog. In any given second, the time used providing musical informations is the time it takes to play a bit times the number of bits in the second. The time spent doing nothing is the rest of that second.

BTW, as a totally separate matter, as I look closer at your numbers I see you are using the sample rate as the time it takes to take a sample. This is not right as they are two completely different concepts.
 
FlexibleAudio said:
S units are 1/s not samples/seconds. Think about it more abstractly and it is clearer to see the units fall into place => in a given period of time, a portion of that period is used providing musical information and a portion is used in doing nothing. The existence of the latter is what separates digital from analog. In any given second, the time used providing musical informations is the time it takes to play a bit times the number of bits in the second. The time spent doing nothing is the rest of that second.

BTW, as a totally separate matter, as I look closer at your numbers I see you are using the sample rate as the time it takes to take a sample. This is not right as they are two completely different concepts.
Click to expand...

Paul- I'd like to dive as technical as it gets, because I think you can help all of us. (and this type of learning is fun for me.) :D

What S's units?

Abstractly, I understand the concept that digital is discrete. The "trick" or the science is to speed up that discrete nature fast enough that the brain cannot tell the difference between discrete or continuous. That's a notable feature of DSD, when it's at 2.8 or all the way up to 11.2mHz.

Speaking of Hertz (Hz), what I found to be the definition is cycles per second. So be it 44,000 Hertz, or 11,200,000 Hz, is the amount of cycles per unit of time. In one second a bit of DSD data is sampled 2,8 million times!

If you have some more information about how DSD quantizes analog data during the analog digital process, I would be interested in knowing the math behind that.

Could you re-frame the math for us and show us how you come up with your original statement of "Analog Equivalence of a Digital Signal = "AE" =1/XLimit of AE as X approaches 0 = Infinity = 100% Analog"

FlexibleAudio said:
Digital creates a virtually perfect square wave and no one can really argue it's superiority from a quantitative measurable standpoint but it just doesn't do for me what analog does. I am convinced it has something to do with the time domain not measured traditionally. No matter how accurate the clocks get, the time domain is not as natural as analog and the brain evolved around hearing all things analog in nature. I am convinced this is also why vinyl derived from digital masters sounds superior to digital derived from digital masters to my ears.

Vinyl sourced from digital will have more distortion (having gone into and out of the digital domain) but still sounds better because it returns to its natural time domain (analog) via a mechanical musical instrument (the stylus) as opposed to trying to match it up in the digital word.

I believe this is also why the Lampi native DSD is the best sounding digital on the planet---it uses no clock (or converter) and effectively is very close to playing a continuos (read analog) signal (using only filters).

X = Time between Bit Samples (secs)
Y = Time per Bit Sample (secs)
S = Sample Rate (1/secs)

X = 1 - (Y*S)

Analog Equivalence of a Digital Signal = "AE" =1/X

Limit of AE as X approaches 0 = Infinity = 100% Analog

When you playback 256DSD on a Lampi X gets pretty darn close to zero. I am not sure how this concept can really be discussed regarding digital formats that use converters and is why the Lampi sound gets so close to analog. The time domain don't get as mucked up. That said, it still ain't analog.

Ok, now everyone stop laughing at me.
Click to expand...

Thank you,

Bill
 
Bill, I don't have much more to add beyond my initial concepts. It really just comes down to my speculation that when super high sample rate dsd is played in its native form the time between samples gets really small which starts to approximate analog within the context of time domain.
 
FlexibleAudio said:
Bill, I don't have much more to add beyond my initial concepts. It really just comes down to my speculation that when super high sample rate dsd is played in its native form the time between samples gets really small which starts to approximate analog within the context of time domain.
Click to expand...

Paul, Okay. Thanks for sharing!
 
o0OBillO0o said:
That was actually a really good summary, for someone who was not there I enjoyed this coverage. Thanks for posting Myles.

-=-=-

I wonder what Mark Waldrep* offered, it is always good to see two sides to the story.

(For those who don't know, Mark rants about DSD, see his many post about DSD here Dr. AIX?s POSTS | Real HD-Audio )
Click to expand...

Not to mention his moronic trashing of me and Greg Beron and others after Axpona this year.
 
This?

Wrapping Up AXPONA: ?We?ll Be Back? | Real HD-Audio

And then there was the seminar that followed the panel on high-resolution audio called “Discover the Reel Truth” moderated by Myles B. Astor, PhD. I met Computer Audiophile’s Chris Connaker in the hall on Sunday and he told me that my head would have exploded at what was being pulled over on the attendees of this seminar. The description from the webpage says it all, “Audiophiles seeking the Holy Grail, the ne plus ultra of sound, need look no further than 15-ips/2-track reel-to-reel tape. These 21st century, real time duplicated, second or third generation [tapes], simply are as close as one can get to the original recordings.” It wasn’t a joke…these people either can’t read specification sheets or have never bothered to listen to real high-resolution audio. Second or third generation tapes of original recordings? Give me great quality transfers to high-resolution PCM digital files and forget losing fidelity through another generation. Save costs, complexity, and arrive at better sound with high-resolution PCM. Wow.
Click to expand...
 
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