Investigation Into Effects Of PC load On DAC Analogue Output

[opus101]

1 hour ago, manisandher said:

I'm certain it was because he didn't want to hear. A bias working in reverse perhaps?

 

It has a name - nocebo effect.

[manisandher]

BTW, where's @March Audio gone?

[jabbr]

The premise that bit perfect playback sounds identical is fundamentally flawed. 
 

Imagine these software settings: 

A: do not inject common mode noise into USB power/ground 

B: inject maximal common mode noise into USB power/ground

 

It should be obvious to anyone that ground loops exist and are audible. 
 

If this were known to be the setting no one would question the results. The problem are undocumented settings whose action have not been explained. The really scientific was to explain a black box is to open it up and look inside. ie reverse engineer — if anyone cared. 
 

Short of that this argument is pointless.

 

I am not saying that common mode noise is the mechanism just that it would be a possible mechanism — there are others : you could send signals down the USB power-ground, you could vary the USB d+|- within allowed values etc etc etc 

[pkane2001]

6 hours ago, PeterSt said:

 

What !?

 

If we change the story somewhat, then I ever back created playback software which today sounds better to anyone trying it out of the box, which means : in Demo mode, while 95% of (bit perfect) SQ features unleash when activated.

And now you are going to ask me how I do that, right ?

Don't expect any perceived clear answers, because from A to Z you won't believe a thing of anything. And if you are going to measure the (by others) perceived differences, then you won't believe one thing of it all definitely. 😉

Still I myself showed the differences easily, and you know it ...

... But there we go again ... those minute differences can't be audible.

 

You know, I don't care about this all. However, I do care a little if someone tells me that I don't know what I'm doing. Not that you said exactly that, but Paul, this is how it reads. I suppose I am biased. 😙

 

 

What? indeed!  What does that have to do with anything I said? All I said is that your software player setting (which you were unable to clearly articulate in the original thread) can alter timing sufficiently to make its effects audible. Believe me, I've done this with DeltaWave. I've done this with Distort, often intentionally. 

 

I didn't say anything about your software, its design, or whether it sounds better or worse. Could it be that you really are a little biased if you are responding to things that you read into what I posted that I clearly didn't say?

[pkane2001]

6 hours ago, manisandher said:

 

Actually, it was 8000 samples @ 176.4k, so around 45ms. But the discrepency was minute - from Mans's analysis:

 

 

Indeed, Mans was happy to accept that the signals all remained bit-identical throughout.

 

And in any event, this had no effect on the ABX. The bits entering the DAC were not affected in any way, shape or form. Only the first ~8000 samples of some digital captures (not all) were mangled by the auto-start function of the recorder.

 

Mani.

I disagree. A 45msec error can’t be safely ignored. To me, this is an error in measurement that needs to be eliminated if we are to trust the test results. Where’s the recording demonstrating that this error was not in the original digital stream? 
 

Mani, explaining away bad results and poor measurements after the fact is not the same as performing the test properly. 

[jabbr]

7 hours ago, PeterSt said:

 

What !?

 

If we change the story somewhat, then I ever back created playback software which today sounds better to anyone trying it out of the box, which means : in Demo mode, while 95% of (bit perfect) SQ features unleash when activated.

And now you are going to ask me how I do that, right ?

Don't expect any perceived clear answers, because from A to Z you won't believe a thing of anything. And if you are going to measure the (by others) perceived differences, then you won't believe one thing of it all definitely. 😉

Still I myself showed the differences easily, and you know it ...

... But there we go again ... those minute differences can't be audible.

 

 

How about this, forget what you think people will or won't believe state:

1) clear answers

2) describe how the software changes the timing

3) or how the software is changing the electrical signal

 

You are treating this too much like a trade secret and I'm afraid your software is not getting the attention it might otherwise deserve. Contrast this with HQPlayer which does a fair amount more documenting filters and modulators and has seen a  much higher usage growth.

 

Tell us what the settings do. 

 

You are being too coy. Shed some light.

7 hours ago, PeterSt said:

 

 

[The Computer Audiophile]

3 hours ago, manisandher said:

BTW, where's @March Audio gone?

He was banned for being himself, a perpetual line stepper who couldn’t help but break the rules. 

[pkane2001]

5 minutes ago, The Computer Audiophile said:

He was banned for being himself, a perpetual line stepper who couldn’t help but break the rules. 

Sad. He was banned from ASR for similar reasons.

[manisandher]

16 minutes ago, pkane2001 said:

I disagree. A 45msec error can’t be safely ignored. To me, this is an error in measurement that needs to be eliminated if we are to trust the test results. Where’s the recording demonstrating that this error was not in the original digital stream?

 

IIRC, the auto-start did something weird like swapping the channels (for the first ~45ms)... of just a few of the digital captures. None of the analogue captures displayed this behaviour, and so the 'glitch' in the digital capturing equipment had no bearing whatsoever on what I heard during the test. I can look into it further when I have some time, if you'd like.

 

But quite honestly, I'd be more inclined to use my time in repeating some tests using my RME.

[pkane2001]

3 minutes ago, manisandher said:

 

IIRC, the auto-start did something weird like swapping the channels (for the first ~45ms)... of just a few of the digital captures. None of the analogue captures displayed this behaviour, and so the 'glitch' in the digital capturing equipment had no bearing whatsoever on what I heard during the test. I can look into it further when I have some time, if you'd like.

 

But quite honestly, I'd be more inclined to use my time in repeating some tests using my RME.

 

If you do, use RME ADC at 384k rate with slow filter. This compensates for the effects of the digital filter up to 100kHz or so. At least that's what RME recommends.

[manisandher]

29 minutes ago, pkane2001 said:

 

If you do, use RME ADC at 384k rate with slow filter. This compensates for the effects of the digital filter up to 100kHz or so. At least that's what RME recommends.

 

OK, will do.

 

Meanwhile, feel free to take a quick look at the digital captures yourself (you'll no doubt be able to do things a lot faster than I can). Here they are for the ABX (30 samples in total), along with the original file used for playback:

 

https://drive.google.com/drive/folders/1z-oXn3Dvouop0pKdjTXliYseGqUnVaOt?usp=sharing

[jabbr]

33 minutes ago, manisandher said:

 

IIRC, the auto-start did something weird like swapping the channels (for the first ~45ms)... of just a few of the digital captures. None of the analogue captures displayed this behaviour, and so the 'glitch' in the digital capturing equipment had no bearing whatsoever on what I heard during the test. I can look into it further when I have some time, if you'd like.

 

But quite honestly, I'd be more inclined to use my time in repeating some tests using my RME.

 

Yes for those of us who understand that as an analogue device, it is the analogue input to the DAC which determines sound as opposed to the interpretation  as bits.

 

As such I am interested in changes in the analogue output of the DAC. It would be great to demonstrate that clearly.

 

My very firm opinion is that if there is a real audible difference in sound (not just SQ) then there will be a real difference in analogue DAC output. A failure to demonstrate that will necessarily be a failure to measure correctly. I am not placing any requirements on what the analogue changs are, nor whether they are bandwidth limited.

[pkane2001]

54 minutes ago, manisandher said:

 

OK, will do.

 

Meanwhile, feel free to take a quick look at the digital captures yourself (you'll no doubt be able to do things a lot faster than I can). Here they are for the ABX (30 samples in total), along with the original file used for playback:

 

https://drive.google.com/drive/folders/1z-oXn3Dvouop0pKdjTXliYseGqUnVaOt?usp=sharing

 

Do you have a key as to what file is what? I certainly don't remember this from three years ago ;)

 

[manisandher]

All the files ending in 0.1 are A. Those ending in 200 are B. (0.1 and 200 are essentially buffer settings in XXHighEnd.)

 

So each ABX starts with 0.1, then 200, and then X (0.1 or 200).

[pkane2001]

28 minutes ago, manisandher said:

All the files ending in 0.1 are A. Those ending in 200 are B. (0.1 and 200 are essentially buffer settings in XXHighEnd.)

 

So each ABX starts with 0.1, then 200, and then X (0.1 or 200).

 

We should probably switch this to another thread, as this is going completely OT here.

 

Curious about the recording chain. Was there something that was configured in XXHighEnd or was there something else in the digital path between the original file and the digital recorder that caused the recording to be so different in an all-digital chain?

 

 

[manisandher]

3 minutes ago, pkane2001 said:

We should probably switch this to another thread, as this is going completely OT here.

 

The original thread is probably as good a place as any?