DeltaWave null-testing audio comparator (beta)

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7 hours ago, fas42 said:

My estimate is that the 3rd is the end of distorting cycle - the circuitry can't process the input waveform; and "smooths" it until it can... Will examine the 3rd next... 

 

Are you suggesting that after the third generation the files will no longer degrade?

[STC]

37 minutes ago, fas42 said:

the copied sample alters for the first 3 passes, and then stabilises.

 

I wonder how many passes the so called original copy went thru before reaching Esldude. So why an additional of three needed to stabilize?  Maybe something else you are overlooking. 

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No one can ask for a better support than this. Good job @pkane2001 , unfortunately for me, I do not have enough knowledge of this software for my need. You have mentioned several places where phase could be detected and it could be useful for me. 

[STC]

1 hour ago, fas42 said:

X-Correlation offset: 1 samples

 

Frank, I am trying to understand the software and hopefully you could explain what is the meaning of the quote? Does it tell the two files are delayed by one sample? Are you even comparing two files in the above post?

 

thanks. 

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21 minutes ago, fas42 said:

This is something that happens virtually every time you go through a DAC, ADC loop; the clocks are slightly different, unless they are deliberately set up to use a single clock.

 

Thanks for the reply. I am trying to measure actual drift of two identical signal but one is delayed by 60 microseconds. I am trying to see if the output is also at 60microsec. Not sure if DW could do that. 

 

Thanks again. 

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6 hours ago, esldude said:

It probably could handle that just fine.  You might give some more details of what the signal is. 

 

Thanks Dennis. I am taking a shot , no several shots, in the dark to understand the FR in my application. Not even sure I understand what to ask. 

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2 hours ago, pkane2001 said:

 

That makes me think I that should compute the error value and thresholds in seconds, rather than in samples. A 1 sample error for 44.1kHz sampling rate is 23μs, which might be audible. A 1 sample error in a 352kHz signal is about 3μs, which is below audibility. Thoughts?

 

My view from different POV where I use samples for delay. I can confirm It is audible. That’s the reason I upsample everything to 96kHz. Ideally, the higher the better. My understanding is the SPDIF coding itself contains a delay of one sample in one channel. Not sure how this thing work inside the software itself.