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Shanti Dual LPS 5V/3A , 5V/1.5A

@cdsgames


1.
When doing your measurements you shouldn't forget to also show numbers
with your isolator in place. Many people own and appreciate it.

The RPI + isolator is a very capable and affordable combo. And does make a difference.







2. THD-N

You meanwhile have the noise and the jitter very well under control on your RPIs AIFs . As I said before, and you basically agreed to it, it's the audio interface which is in charge to deliver highest quality results. No matter which source is feeding it. You've been doing extremely well on that account.

The new question if the SBC suddenly should now be in charge to improve THD-N on the audio interface is nonsense. It's again the audiointerface which is in charge.
If a new "audiophile" SBC improves the THD-N on the the audio interfaces it simply shows there's still a loophole/weak spot on the audio device that impacts THD-N when changing the input conditions.

It just shows there's still room for improvement on the audio interfaces.


Enjoy.


1. We will retest the Boss +isolator + RPI vs RPI + Bossand see the THD+N numbers , last I remember they did not change.



We will test isolator + USBridge



2. I think the question is "WHY" . The digital stream is same , jitter is same, but we observe much more noise on the i2s lines from RPI. Every book , every theory says that noise on the digital streams do not matter . just ones and zeroes do. We practically observe thats all incorrect (theory)


In any case , please have patience , we have mass production going on , FCC testing (Nirvana) , Revolution DAC improvements , fine tuning the USBridge (we made sure you can draw 1A with only 180mV drop on USB lines)


I want to reiterate that USBridge Sig / Audiophile SBC has enough bandwidth on USB for DSD 1024 (DSD 512 was tested with multiple DACs )
 
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We practically observe thats all incorrect (theory)

I wouldn't agree to that.

What you actually observe is that there are more variables in the game.

Those variables not known to you (yet) or not addressed by you (yet) I called loopholes. There are plenty of them out there.

The fact - you outlined it - that your main focus is on "noise" only shows the weakness of your approach.

If all this matters at these low levels - who knows !?!?

Anyhow. Nobody really needs a proper scientific paper about what's actually going on there. The results you guys deliver are simply highly competitive. And you keep pushing the limits in the affordable audio arena.

Great times.

:D
 
I am unable to find any other loophole or variable . Everything is the same , DAC board , jitter , power


The only change is the i2s source .


Now , I have to disagree with you once again . Reducing noise on audio is not a weakness . Its the basis of audio design . Yes , impedance on rails matters but its not like we have pursued noise reduction and sacrificed everything else.. in fact we have more capacitance on PCB than ever .We are using thousands of uF and even F caps :)


Of course , for sound quality (DAC for example) a lot of things matter , your analog stage design , choice of caps (film , cog) R tolerance , running in class A...and we implemented all that. Still it would be a lost cause if the noise and impedance on opamp rails is high.


Coming back to USbridge Signature/ Audiophile SBC, we made a choice. We treated all digital as analog signals . We cleaned them , we scrub them and we provided plenty of decoupling . Yes we also take care of jitter (when possible)



Weakness you say ? I say vision . However in this case the vision belongs to the DIY community , to the many testers out there , that reported time and time again "better SQ with better digital source"


We believed and implemented on board.
 
Weakness you say ? I say vision . However in this case the vision belongs to the DIY community , to the many testers out there , that reported time and time again "better SQ with better digital source"


We believed and implemented on board.


That's nonsene. These "testers" are simply clueless. And that includes me.
And you simply draw the wrong conclusions by listening to this clueless bunch.

I am in this computer audio tweaking audio longer then most people around.
I experienced a lot of headwind in the early days (2007/2008).


The key issue is NOT the computer. As long as the databits are properly delivered ( 1=1 0=0) to the audio interface the computer has done its job.

The audio interface will then be in charge to cope with these bits.
And not only that. The audio interface is also in charge to cope with the
computer induced problems surrounding these bits (noise, jitter, power variations, vibrations, heat, you name it.. )

As long as the golden ear fraction experiences differences by swapping out digital transports, we all know that the audio interface is not doing its job properly. It's as simple as that.

All this computer tweaking (HW and SW) has always been a measure to cope with the inadequate audio interfaces attached to it.

The ultimate goal is to get an audio interface that sounds the same no matter what transport it gets attached to.
 
Nope. The source just delivers 0 and 1s.


cdsgames statements pretty much reflect the situation.
They simple havn't understood the whole very complex situation yet.
Allo is pretty new in the audio business.
What you can see is a steep learning curve. With every new product they
release they go a step up.
And they made it quite far in shortest time and with full commitment.

Not long ago cdsgames was telling us their 11$ power supply would be more than sufficient.
Obviously that looks different today. ;)


If somebody would have told you 5 years go that

* a 35$ computer and
* a 250$ DAC
* a 55$ isolator
* a 100-200$ power supply


would be able to compete in the audiophile world out there, I think nobody would have believed that.

And no. I don't think the transport will always have an impact. There are already top audiophile devices out there which pretty much do not exhibit
any differences no matter which (digital) input is used.
 
What my testing is showing , is that for different sources with the same ones and zeros, THD changes. (jitter is same)


Furthermore is the harmonics amplitude that changes.


This is not subjective but test data. I think its because of the noise on the carrier but at this point..unclear.
 
What my testing is showing , is that for different sources with the same ones and zeros, THD changes. (jitter is same)

Nobody questions your measurements. I'm questioning your conclusions (and measures).

You're simply not able to nail the problem yet. And therefore you are not in the position to draw (final) conclusions.


As I said. There's a loophole on the audio interface ( the logic behind that I outlined earlier).

As a workaround you can of course improve the source. That's what many of us doing since more then 10 years. ;)

But that simply puts the focus away from getting the audio interface done properly.
As a consequence the loophole(s) is/are going to persist. :ill:


From a commercial strategy point of view this is a nice move of course.
You can sell another product. Fair enough. :D

And, as I said, for those who want to stick to their flawed DACs, such a transport might be a very valid tweaking option.
There's a huge market out there. I'm sure about that.
 
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Nobody questions your measurements. I'm questioning your conclusions (and measures).


Just to be clear , you are questioning my conclusions but you agree that test data is correct .


You say its a "loophole" in the DAC itself , but you are unable to explain it (noise , distortion , etc)...


This loophole sounds a lot like a black hole , mysterious and escaping all knows laws.

:)
 
Just to be clear , you are questioning my conclusions but you agree that test data is correct .

It's shouldn't be that difficult to understand.

1.
I do not "agree that your test data is correct". How could I?
I do believe your measurements, the ones you are presenting, are quite reliable, since ASR usually confirms them more or less.

Non of us around here has the tools to confirm your measurements at these
low levels.


2.
We all also know that you lack certain knowledge in certain areas - you basically told us - remember - you wouldn't know how to measure key performance indicators rise-time/settling-time and transient response of a power supply. This I'd call a blind spot potentially allowing for loopholes.

Then you pretty much stick to the textbook audio measurements and you're "mainly focused on noise" all the time - your words! I do think that's another potential weak spot.

And I do think because of that I'm not exaggerating when saying yes there might be certain blindspots on your side that allow for loopholes - weak spots you havn't identified yet.

As a blindspot example from the past we can use Katana V1 and the awful power rail modulations we've seen in the beginning.
The measurements looked OK on the output. There's been still something going on on the inside.
The buffers on the rail very were simply to small, causing the modulation. And this then led to a major retrofit from your side. There's nothing wrong with it. These things happen.


Since there'll always be certain blind spots, you can't come up with final conclusions. And tweaking the transport remains a workaround to me.