DeltaWave null-testing audio comparator (beta)

[esldude]

I know you were going to be away from working on Deltawave for a while.  When you get back to it, I've run across a small bug in 1.37B.  In the original spectra tab.  And only in that tab, if you select or de-select log axis, this tab doesn't get updated.  If shows whichever way the software opens.  If Log axis is checked it stays log regardless of the selection.  If log axis is unchecked when it opens then the scale stays linear.  All the other tabs update properly, but not the Orignal Spectra display. 

[esldude]

Maybe I just didn't notice it before, but the various spectrogram palletes are a nice touch. 

[esldude]

I think someone reported this in the 39 version.  In 42, if I play the result of the comparison, the difference file, I don't hear anything until I change the gain setting.  Even stopping and restarting by hitting the green play button, I hear about one second of the result and then it goes silent until I nudge the gain slider a bit up or down. 

[esldude]

6 hours ago, pkane2001 said:

 

Is this possibly because the delta waveform is very low level after the first second or so? I can play the whole delta file with no problems, and not touching the volume control.

 

Is there some specific set of steps that you can think of that lead to this, or just pressing the delta play button causes this on the first try?

I hear something for about a second and then silence even if I let the whole track play.  Nudge the gain slider at any point and I get sound.  I'll mess with it some more later and see if it matches any particular steps, but it doesn't seem to be the case so far. 

[esldude]

5 hours ago, pkane2001 said:

 

Strange, I'm unable to reproduce. Do you click Match before, or just Show? Does it do this every time with any file? What type of audio driver (ASIO or WASAPI)?

I was using WASAPI, but also tried it with an external interface over ASIO. 

 

Load two new tracks.  Hit the green play button, get the notice of raw processing only with all matching turned off.  Then it plays as you would expect.  If however I stop play and start it again it drops volume after a second. 

 

I do a match, and after the match hit play, and it plays at the right volume for a second, and then reduces greatly.  Listening to some files that null poorly it does not go silent, but the volume drops maybe 20 db.  I just barely move the gain slider and it puts volume in the proper position.  I can pause playback and start it again, and it behaves the same dropping volume after a second until you nudge the gain slider.  I can listen to Ref or Comp files and everything is fine.  Listen to the difference and volume drops after the first second until gain is nudged. 

 

I can load one or two new files, do a match and it acts just the same.   The only difference in doing a match first is even the first play of the difference file drops volume after one second.  

[esldude]

10 minutes ago, pkane2001 said:

 

Thanks, Dennis. Let me see if I can get to the bottom of this. Can’t imagine how or why this would happen

I am puzzled how it could happen too.  

[esldude]

All looks good to me now. 

[esldude]

What do you think a delta/null value is between the original source file and one captured from the output of a DAC? 

 

If you correct for amplitude and phase errors before nulling, as you can with DeltaWave, you get the more complex SINAD value for a musical recording. That is, RMS value of all the noise and harmonic distortions with all the fundamental frequencies ‘notched out’.

 

The above from Paul I wanted to continue discussing here. 

 

Also this:

 

Right. The software is not perfect, but a perfect ADC is not needed for a comparison, as long as the same one is used. There’s also a way to eliminate the influence of the ADC in measurements, but let’s continue this in the DeltaWave thread...

 

What do you have in mind in regards to eliminate the influence of the ADC in measurements?

 

[esldude]

Some observations about good results.  

 

Looking at the Diffmaker loop results at Gearslutz, my Zen Tour is currently 12th in a very long list of gear.  Yet it has a difference null of a bit more than -56 db.  

 

Unsynchronized  DAC and ADC, the March Audio Dac and my Zen Tour sit currently 3rd in a large list at about the same number.  This is a list nearly 9 years in the making. 

 

The best numbers of all are all good DACs feeding good ADCs where the DAC also provides the master clock. Top of the list is a Forsell MDAC feeding a Focusrite Blue Mastering 245 ADC.  This Focusrite is not your regular little interface it is a pro quality device.  The best of those get -81 db for the difference null.  Right behind it fractionally is an SPL Madison DAC feeding an evaluation board TI ADC.  

 

If I invoke Level EQ and Phase EQ those top two only improve to about -90 db.  If I do the same with my Zen Tour loopback and my March-Zen Tour they improve much more to the -86 db range.  So it would appear phase and FR are the main differences.  It looks like viewing the graphs in Deltawave the very low end is the biggest difference.  The top ones are flatter down to a 1hz or less, while others are starting to droop around 3 or 5 hz.  

 

Surprisingly the RME ADI is a couple db lower in results than my Zen Tour despite boasting better individual measurements of the ADC and DAC. It too improves to around -84 db difference if Level EQ and Phase EQ are used.  So is my Zen Tour really better than the RME all things considered?  Yeah, I know a couple db isn't much.

[esldude]

8 hours ago, pkane2001 said:

 

The differences shown in the Diffmaker list are inclusive of phase and amplitude filter errors. DeltaWave removes these with nonlinear-EQ engaged, so you get a cleaner looking result. It doesn't mean that the phase and filter errors are not important or even audible, it just means that DiffMaker clumps all the errors together in one number, while DeltaWave allows you to separate them out. If you subtract the RMS error with and without nonlinear EQ in DW, you'll get the degree of influence of the filter phase/amplitude errors on the result.

 

In effect, the -84dB difference compared to a -90dB difference with nonlinear EQ means that the SINAD of one device is better by 4dB when measured with a complex music stimulus, excluding non-linear phase and amplitude effects of the device. Again, that's just one measure that's better, not including the effects of digital/analog filters, which may make it measure and/or sound worse than the other device.

 

I think I understood as you intended.  Once you've shown a difference the next question is what is causing the difference.  Your other choices for EQ, Phase and non-linear timing let us see where some differences are.  It appears most of them are FR and phase. Which isn't surprising.  Though still nice to confirm with some rigor. 

 

One interesting thing your software made easy to see.  Comparing a recording done now and a few minutes from now with nothing changed the nulls are of course very good.  Most of the remaining difference is 60 hz hum.  Everything in the process is the same, but I can't synchonize the 60 hz hum (even if below 100 dbFS) from one run to the next.  So sometimes it is closer in phase on 60 hz than other runs.  Which is visible in the Difference spectrum plot, and alters the already low numbers a few db.  Then you can of course notch out 60 hz.  

[esldude]

2 hours ago, TomCapraro said:

Hi Paul, can you explain better what you mean by this process? maybe it's something I do as well but ... when you talk about inverting the ADC add and then split I can't translate it

Yes, I meant to ask about this too, but sidetracked myself. 

 

Actually wouldn't these signals council out?  I've got a few of those where I recorded everything twice.  Wouldn't this be the same as inverting the second one and mixing together or am I missing something.  I also am not sure this separates the effects of DAC and ADC if you do this. 

 

I did think about taking the difference signal, inverting it and adding to the original.  Run the original plus inverted difference back thru, and compare this one to the unaltered original.  But if that worked perfectly, I still just get the original file.  

 

 

[esldude]

3 hours ago, pkane2001 said:

 

I should know better than to come up with "brilliant" ideas before my first cup of coffee

 

Well don't feel bad.  I've had such "brilliant" ideas many times.  About once very 5 times it has some real merit.  If I didn't dabble in such ideas I would not have the really good ones from time to time.  

 

And sometimes you just get lucky.  After complaining about how awful Diffmaker was, but how good the idea of it is, some fellow said he thought he could do better.  I sneakily encouraged him at every turn.  I asked for new features and he quickly supplied them almost like having my own accomplished programmer and signal analyst work for me.  And here we are less than a year later with this useful accomplished bit of software called Deltawave. 

 

 

[esldude]

Here is what I get using TomCaproro's modulated white noise signal using a loopback vs a loopback a few minutes later.  This is about what I get using music the same way. 

 

 

I get good result comparing the loopback vs the original, but not as good as Tom is getting.  I also don't get the low picosecond jitter that way.  Though I get that low jitter comparing two loopbacks. 

 

A signal I was working on with similar thinking behind it is 997 hz at -6 db mixed with white noise at -6 db.  That actually worked a little better for me.  Here is that signal comparing original to loopback.  

 

[esldude]

Here is an interesting result using the 32 multi-tone test signal Amir uses from the AP test company.   This is the original vs loopbacked result. 

 

[esldude]

5 minutes ago, vortecjr said:

Can you post a link to the source file used in this test?

It is provided by AP for use with their analyzers.  

https://www.audiosciencereview.com/forum/index.php?threads/sinad-measurements.4071/post-245803

 

There is a link to it in this post. 

 

The following link is a post describing the file.  It has frequencies that fall directly in the middle of an FFT bin when used at 192 khz and when using a 64K FFT.  This allows you to use rectangular windowing on the FFT for the cleanest look at the noise between tones.  The tones are also altered in phase to allow a higher average level without clipping. 

 

https://www.audiosciencereview.com/forum/index.php?threads/review-battle-of-schiit-audio-dacs.5487/post-121956

[esldude]

3 hours ago, TomCapraro said:

How many seconds?
Tone at 997hz of the same length as white noise?
I also try
Is the rectangular FFT window on DW absent? I don't see any "rectangular" item

I used 1 minute 10 seconds. Yes same length on both signals. I didn't see rectangular as a choice either. I think I used hamming for the one I posted. It might have been better to use minus 7 db for the white noise to prevent clipping.

[esldude]

8 minutes ago, TomCapraro said:

I imagined that the pseudo-random signal pair (with 64k pseudoperiod) + the rectangular FFT ... that Paul put as Dirichlet would have given very good results.
The representation on the spectrum is perfectly flat, as it should be.
Indeed, for me they are excellent because the jitter rms has dropped to only 19ps, the spectrum of delta shows a very low comb that corresponds to the amplitude of the non-linear alterations of the DAC.
Paul, you're close to perfection ... congratulations ... !!!

What DAC and ADC are you using for this?

[esldude]

Here is the 997 hz tone mixed with white noise that worked pretty well for me in loopback using simple waveform.  

 

You can try it out and see if it is of any use. 

 

997 white noise minus 1 db.zip

[esldude]

2 hours ago, TomCapraro said:

I compared the result obtained with your signal. I think that every ADC should be coupled with a certain signal and then always use that.
I say this because I get worse results than the 64k white noise.

That is odd.  I get better results with the signal I posted than yours.  And yours is just the reverse.  With the modulated white noise I get 15 db poorer difference results and 12 db poorer correlated null results.  This is with measure simple waveforms checked. 

 

With Measure simple waveforms unchecked.  My results on the modulated white noise deteriorate another 7 db, while with the signal I posted it improves an additional 10 db.   Makes me think neither signal is a good signal for general checking of how an ADC/DAC loop is performing. 

 

 

 

[esldude]

23 minutes ago, TomCapraro said:

Can you please post the wav delta file that I overlay with mine?
Probably there will be some signal even more suitable, but 19ps of RMS jitter and the difference spectrum so low from me produces results in line with the non-linear alterations of the dac

I'll do that later when I get back home.

[esldude]

Is this what you want, anything else that would help.  Same overlay only reversed so you can see results for both files in the lower pane of the window.  Settings were the same for both. 

 

 

[esldude]

With the same exact settings.

 

 

[esldude]

1 hour ago, pkane2001 said:

 

Not sure if that's the problem here, but remember that Simple Waveforms setting disables drift correction. If there's a clock drift between reference and comparison files, the null will not be as good.

Yes that was my understanding of how it worked.  I was just copying Tom's settings.  I normally uncheck non-linear drift compensation as I understood no drift compensation happens if Simple is checked. 

 

I don't know there is a problem, but it is surprising to me.  I didn't realize changing the window for the spectrogram also changed the nulling results.  Am I correct about that being how it works. 

 

I wasn't surprised settings for phase limit and EQ threshold would change the results.  It looks like the best setting for that is going to be down to the individual characteristics of the file.  

[esldude]

3 hours ago, TomCapraro said:

Good evening, I posted an original file (tone 1000hz) and its registration.
Notice if the THD values come out correct ... it doesn't seem to me ...

orig.wav 7.32 MB · 1 download rec.wav 7.32 MB · 1 download

It looks correct to me in the spectrum of delta.  It might appear wrong in the delta of spectra, but actually is correct from what I am seeing.  

[esldude]

1 minute ago, TomCapraro said:

What values do you get in the results?

-27.47 db for difference (which just eyeballing looks right for the 3rd harmonic in the recorded file).