TOTALDAC DAC
Oct 20, 2015 at 2:08 PM Post #226 of 593
  What's the general opinion of the d1-integrated? Which is the d1-single and server/clock in a single box if I'm not mistaken.

Quite a lot cheaper than the mono or d1-dual + server/clock. I wonder what degree of gains you get for 1/3 or 1/2 the price.

This is a good question.  I understand what @bmichels is saying and I believe he is right but on the TotalDac website, it says it has an "embedded reclocker".  Should you decide to go with this option, you would have the option of using a USB Regen by Uptone Audio between the Server and DAC which could probably effectively take the place of the reclocker.  @preproman tried this and didn't find a difference between the two.  The only problem with the d1-integrated is that it is underpowered.  It will not drive the Abyss well.
 
Oct 20, 2015 at 2:19 PM Post #228 of 593
Don´t just think reclocker instead of Regen. Reclocker AND Regen - there comes the real magic. They are not doing the same thing.
 
Oct 20, 2015 at 2:20 PM Post #229 of 593
Using RCA takes no advantage of the dacs

Actually, it does.  It just doesn't take full advantage of it.  Effectively, it becomes like a d1-single which is 100 resistors per channel instead of 200 resistors.  You will have to decide if the compromise is noticeable.
 
Oct 20, 2015 at 2:25 PM Post #230 of 593
Don´t just think reclocker instead of Regen. Reclocker AND Regen - there comes the real magic. They are not doing the same thing.

They are doing similar things.  I have a Regen (with linear PSU) on my d1-monoblocs now which by default includes the reclocker and the Regen is adding nothing noticeable.  When I had the Regen between my Auralic Aries and Bricasti M1 DAC, the difference was quite apparent.
 
Oct 20, 2015 at 2:43 PM Post #231 of 593
  They are doing similar things.  I have a Regen (with linear PSU) on my d1-monoblocs now which by default includes the reclocker and the Regen is adding nothing noticeable.  When I had the Regen between my Auralic Aries and Bricasti M1 DAC, the difference was quite apparent.

 
I confirm that Vincent told me over the phone that the d1-integrated do not have the reclocker;
 
 
Romaz, please can you tell me what type of Regen (with linear PSU) did you use ?  (curently I use an ARIES)
 
and...
 
we all will be interested to see if, at the end of the day, you will still use an external Headphone amp or...if you will only use the HE1000 + TotalDAC Monobloc  (direct connection) and get rid of all your external amp, including your BHSE/SR009 combo  
wink.gif
   (I wait for your finding before I pull the trigger on a BHSE or... a KGSSHV Carbon )
 
Oct 20, 2015 at 3:09 PM Post #232 of 593
   
I confirm that Vincent told me over the phone that the d1-integrated do not have the reclocker;
 
 
Romaz, please can you tell me what type of Regen (with linear PSU) did you use ?  (curently I use an ARIES)
 
and...
 
we all will be interested to see if, at the end of the day, you will still use an external Headphone amp or...if you will only use the HE1000 + TotalDAC Monobloc  (direct connection) and get rid of all your external amp, including your BHSE/SR009 combo  
wink.gif
   (I wait for your finding before I pull the trigger on a BHSE or... a KGSSHV Carbon )

My USB Regen is connected to the 9v output on my HD Plex linear PSU.  This same linear PSU also powers my cable modem/router/switch (12v) and my small Bantam Gold class D amplifier (19v) so this linear PSU has come in very handy.  In comparison to the stock switching PSU that comes with the Regen, I noticed a subtle improvement.  
 
I also have an Entreq Poseidon arriving tomorrow and I will be grounding the signal on the monoblocs + music server + cable modem/router/switch.  This is supposed to make a very significant difference based on the experience of others and could elevate a d1-dual to monobloc status but we'll see.
 
The CAT arrives tomorrow and I should get the Aurender N10 back by the weekend.  Hopefully, I can answer your other questions by the end of next week.
 
Oct 20, 2015 at 3:53 PM Post #233 of 593
My USB Regen is connected to the 9v output on my HD Plex linear PSU.  This same linear PSU also powers my cable modem/router/switch (12v) and my small Bantam Gold class D amplifier (19v) so this linear PSU has come in very handy.  In comparison to the stock switching PSU that comes with the Regen, I noticed a subtle improvement.  

I also have an Entreq Poseidon arriving tomorrow and I will be grounding the signal on the monoblocs + music server + cable modem/router/switch.  This is supposed to make a very significant difference based on the experience of others and could elevate a d1-dual to monobloc status but we'll see.

The CAT arrives tomorrow and I should get the Aurender N10 back by the weekend.  Hopefully, I can answer your other questions by the end of next week.


OT
Im about to place a order on a MSB Analog and i have not decided if il also go for the upgraded power plant if i gain anything or just looks.
Then the Quad DSD USB module:
Do you think i would gain any improvment with my new Regen, because the Regen disconnect the 5volt signal to the USB module, and from what i think that will improve the signal from signal distortion or what is your opinion ?
 
Oct 20, 2015 at 4:21 PM Post #234 of 593
OT
Im about to place a order on a MSB Analog and i have not decided if il also go for the upgraded power plant if i gain anything or just looks.
Then the Quad DSD USB module:
Do you think i would gain any improvment with my new Regen, because the Regen disconnect the 5volt signal to the USB module, and from what i think that will improve the signal from signal distortion or what is your opinion ?

 
@negura would be the one who can comment best regarding the relative differences that the Analog Power Base and Quad DSD USB module make.  As for the USB Regen, Steve Plaskin tried it on his MSB Analog in his review on AudioStream and thought very favorably of it.  Here is the link:
 
http://www.audiostream.com/content/uptone-audio-usb-regen-and-audioquest-jitterbug#OuYyTKO1GWBfCLpD.97
 
Oct 20, 2015 at 5:25 PM Post #235 of 593
 
 

Quote:
I wish I could speak French half as good as you can speak English.  There are actually numerous studies and this threshold has been reported to be anywhere from 5-10µs.  Here is a link to one of those studies and it references others:  
http://boson.physics.sc.edu/~kunchur/temporal.pdf
 
Here is also a video by Hans Beekhuyzen that you might find helpful:
 
https://www.youtube.com/watch?v=r_wxRGiBoJg

thanks. the video wasn't relevant to me(ok vulgarisation though, I was expecting more BS from someone pushing for his product). but the paper was nice. at last I know how those crazy low numbers are obtained, and it has nothing to do with listening to music.
they use iterated ripple noise in one case, and square waves in the other, none of which you can really hope to find in natural sounds. still the results seem to be relevant in the context of the experiment. it's trying to use them as is, in the context of listening to music, that doesn't work with me at all.
 
 with the variety of sounds in music, all the possible masking, and the little musical content in high frequencies, I'm not sure there actually is a need for such sampling speed for most music content. but more than that, I highly doubt the average human could hope to still react to such low temporal thresholds while listening to music.
another example that come to my mind to explain why I believe this: maximum audible dynamic range.
when testing it, we can rapidly demonstrate in an anechoic chamber that the average human can hear very very low sounds at, let's say 1db to make it simple. it is then very easy to push the loudness until it hurts, somewhere in the 130 or 140db. and there we have it, most humans can hear up to almost 140db of dynamic!!!  that's true within that given test. not at all true when listening to music. 
with average musical content, sounds of different loudness are coming in at once, not one, then silence, then the other. in this case most humans will struggle to discriminate more than the first 60db of music in instantaneous dynamic. and that's a figure that actually relates to my own experience of music. if I add some sounds 80db below most tracks, I fail to notice the added sound almost all the time. so we can't just take the 140db figure and slap it onto what music needs to do to sound right. it's a misuse of data.
 
I'm confident it's the same with timing, first because again there are no convenient cues like IRN or square waves in a song, so it is obvious the threshold will crumble with real music. 
but also because the music will most likely be complex with many temporal cues of many different nature. when the brain is working on getting the pitch, the position, recognizing the song, the instruments... we fall in the drop of performance due to multitasking and the question of the brain focusing on what matters most to it and rejecting most of the information coming into the ear.
and that aligns with the typical failure of controlled tests to discriminate 16/44 vs highres. while your numbers and logic do not. if all the stuff between 7µs ans 20µs was really noticed with real music content, why would we fail so easily to notice them in a blind test?
I see this as just another out of context misuse of a valid science experiment. all this does not make a case against highres, but it also fails IMO to make a valid case against CD quality not being enough.
 
Oct 20, 2015 at 6:45 PM Post #237 of 593
 
thanks. the video wasn't relevant to me(ok vulgarisation though, I was expecting more BS from someone pushing for his product). but the paper was nice. at last I know how those crazy low numbers are obtained, and it has nothing to do with listening to music.
they use iterated ripple noise in one case, and square waves in the other, none of which you can really hope to find in natural sounds. still the results seem to be relevant in the context of the experiment. it's trying to use them as is, in the context of listening to music, that doesn't work with me at all.
 
 with the variety of sounds in music, all the possible masking, and the little musical content in high frequencies, I'm not sure there actually is a need for such sampling speed for most music content. but more than that, I highly doubt the average human could hope to still react to such low temporal thresholds while listening to music.
another example that come to my mind to explain why I believe this: maximum audible dynamic range.
when testing it, we can rapidly demonstrate in an anechoic chamber that the average human can hear very very low sounds at, let's say 1db to make it simple. it is then very easy to push the loudness until it hurts, somewhere in the 130 or 140db. and there we have it, most humans can hear up to almost 140db of dynamic!!!  that's true within that given test. not at all true when listening to music. 
with average musical content, sounds of different loudness are coming in at once, not one, then silence, then the other. in this case most humans will struggle to discriminate more than the first 60db of music in instantaneous dynamic. and that's a figure that actually relates to my own experience of music. if I add some sounds 80db below most tracks, I fail to notice the added sound almost all the time. so we can't just take the 140db figure and slap it onto what music needs to do to sound right. it's a misuse of data.
 
I'm confident it's the same with timing, first because again there are no convenient cues like IRN or square waves in a song, so it is obvious the threshold will crumble with real music. 
but also because the music will most likely be complex with many temporal cues of many different nature. when the brain is working on getting the pitch, the position, recognizing the song, the instruments... we fall in the drop of performance due to multitasking and the question of the brain focusing on what matters most to it and rejecting most of the information coming into the ear.
and that aligns with the typical failure of controlled tests to discriminate 16/44 vs highres. while your numbers and logic do not. if all the stuff between 7µs ans 20µs was really noticed with real music content, why would we fail so easily to notice them in a blind test?
I see this as just another out of context misuse of a valid science experiment. all this does not make a case against highres, but it also fails IMO to make a valid case against CD quality not being enough.

I'm glad you found the paper useful.  We assume that hearing is a non-discrete phenomenon when in reality, we hear things in 5-10µs intervals.  It is interesting to read your thoughts on time resolution from the standpoint of listening to music and I agree with your well thought out reasoning on the validity or lack thereof with using 192kHz oversampling as a standard.  Is it qualitatively superior to 16/44?  I have not found that to be the case unequivocally but I do have some DSD material that seems to portray that extra bit of refinement.  Whether this is due to other factors such as superior mastering, I don't know.  While no one is ready to proclaim the death of CD, increased bit depth and upsampling seems to be where music is heading which is as much financially driven, no doubt.
 
Oct 21, 2015 at 6:30 AM Post #238 of 593
 
The CAT arrives tomorrow and I should get the Aurender N10 back by the weekend.  Hopefully, I can answer your other questions by the end of next week.

 
Latest (good) news received from Scott from CAD : "  ROON implementation on CAT is done.  I have fully tested ROON on the CAT and it works fine.
We can offer ROON installed on the CAT but due to the high cost of ROON we offer it as an option rather than standard.
As an option we also have upgraded DC interconnects for the CAT Linear Power Supply using OCC copper wire made just for us - it's seriously good."
 
biggrin.gif

 
Oct 21, 2015 at 8:12 AM Post #239 of 593
 
 
thanks. the video wasn't relevant to me(ok vulgarisation though, I was expecting more BS from someone pushing for his product). but the paper was nice. at last I know how those crazy low numbers are obtained, and it has nothing to do with listening to music.
they use iterated ripple noise in one case, and square waves in the other, none of which you can really hope to find in natural sounds. still the results seem to be relevant in the context of the experiment. it's trying to use them as is, in the context of listening to music, that doesn't work with me at all.
 
 with the variety of sounds in music, all the possible masking, and the little musical content in high frequencies, I'm not sure there actually is a need for such sampling speed for most music content. but more than that, I highly doubt the average human could hope to still react to such low temporal thresholds while listening to music.
another example that come to my mind to explain why I believe this: maximum audible dynamic range.
when testing it, we can rapidly demonstrate in an anechoic chamber that the average human can hear very very low sounds at, let's say 1db to make it simple. it is then very easy to push the loudness until it hurts, somewhere in the 130 or 140db. and there we have it, most humans can hear up to almost 140db of dynamic!!!  that's true within that given test. not at all true when listening to music. 
with average musical content, sounds of different loudness are coming in at once, not one, then silence, then the other. in this case most humans will struggle to discriminate more than the first 60db of music in instantaneous dynamic. and that's a figure that actually relates to my own experience of music. if I add some sounds 80db below most tracks, I fail to notice the added sound almost all the time. so we can't just take the 140db figure and slap it onto what music needs to do to sound right. it's a misuse of data.
 
I'm confident it's the same with timing, first because again there are no convenient cues like IRN or square waves in a song, so it is obvious the threshold will crumble with real music. 
but also because the music will most likely be complex with many temporal cues of many different nature. when the brain is working on getting the pitch, the position, recognizing the song, the instruments... we fall in the drop of performance due to multitasking and the question of the brain focusing on what matters most to it and rejecting most of the information coming into the ear.
and that aligns with the typical failure of controlled tests to discriminate 16/44 vs highres. while your numbers and logic do not. if all the stuff between 7µs ans 20µs was really noticed with real music content, why would we fail so easily to notice them in a blind test?
I see this as just another out of context misuse of a valid science experiment. all this does not make a case against highres, but it also fails IMO to make a valid case against CD quality not being enough.

I'm glad you found the paper useful.  We assume that hearing is a non-discrete phenomenon when in reality, we hear things in 5-10µs intervals.  It is interesting to read your thoughts on time resolution from the standpoint of listening to music and I agree with your well thought out reasoning on the validity or lack thereof with using 192kHz oversampling as a standard.  Is it qualitatively superior to 16/44?  I have not found that to be the case unequivocally but I do have some DSD material that seems to portray that extra bit of refinement.  Whether this is due to other factors such as superior mastering, I don't know.  While no one is ready to proclaim the death of CD, increased bit depth and upsampling seems to be where music is heading which is as much financially driven, no doubt.


well there are some benefits to oversampling a song, most commonly suggested is to get an easier/cheaper time dealing with the low pass filtering to stay within nyquist/shannon theorem. but how much audible benefits we get from having the guy doing it before selling the track to us(as least for any slightly old albums), or us doing it with SOX, or simply let a DAC do it? in the end the low pass filter used in the DAC could be the same when playing those 3 files(OK last option might be a problem to test with NOS DACs ^_^).
the rest is really just a matter of money as you say(and my one and only real grudge against highres as we have it).
 
Oct 21, 2015 at 8:28 AM Post #240 of 593
 
What's the difference in the two?

 



It is best to go in to computeraudiophile.com to get those answers. I could hear an improvement with the Regen.
 

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