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DAC measurements using DeltaWave

Mani, as I mentioned before, non-linear EQ is designed to eliminate the differences in the null caused by filters or other, consistent frequency response differences between two waveforms.

Thanks Paul. Good to know that I'm correct in not applying any of DW's EQ.

But my post was really in response to this:
Having now seen the length of this file, as Paul already suggested, you'll get better results with a longer piece of music to work with. 30 seconds is near the lower end where Deltawave works well. After using it quite a bit I don't bother with less than a minute. 2-4 minutes is optimum I think.

Edit: I purposely used the full track of >3.5 minutes in my post: https://audiosciencereview.com/foru...asurements-using-deltawave.59822/post-2194642

DW works perfectly well with my reference track of 30 seconds, as I have no need of clock drift correction or EQ.
 
With the Christmas break, I finally managed to find some time to compare the analogue outputs of some DACs playing real music (as opposed to sine waves), using @pkane 's excellent DeltaWave nulling software.

In the next post I’ll outline the equipment and method I’ve used. But I’d like to reserve this OP for a ‘league table’ of DACs.

Here are the results to date (10 samples each):

View attachment 418836

I find the charts in DeltaWave to be more insightful.

Wadia 15 (4x PCM63-K)
View attachment 418837

SMSL SU-10 – no filter (2x ES9038Pro)
View attachment 418838

SMSL SU-10 - fast linear
View attachment 418839

Chord DAVE (custom FPGA)
View attachment 418840

The Chord DAVE actually performs the best in these tests, even though it falls far short of the SMSL SU-10 in THD+N (see https://audiosciencereview.com/forum/index.php?threads/smsl-su-10-dac-review.38415/ and https://audiosciencereview.com/forum/index.php?threads/chord-dave-review-dac-hp-amp.35974/).

Due to the requirement to synchronise clocks (to obtain accurate nulls), I’m restricted to DACs with spdif inputs. Other DACs that I have here that I’ll get around to testing eventually include:
  • Gustard X30 (4x ES9039Pro)
  • SMSL DL200 (ES90392QM)
  • Audio Synthesis DAX (2x PCM63-K)
  • Audio Synthesis DSM (UltraAnalog D20400A)
All thoughts welcome.
This seems rather intriguing, but I'll have to read up on what it actually is :)

Is it essentially taking two time-domain wave forms, scaling/aligning them and then computing the differential between the two? Then performing further analysis on the resultant differential waveform?
 
Thanks Paul. Good to know that I'm correct in not applying any of DW's EQ.

But my post was really in response to this:


DW works perfectly well with my reference track of 30 seconds, as I have no need of clock drift correction or EQ.

30 seconds is plenty if you’re not using clock drift or non-linear EQ corrections.
 
i found this in the RME user manual. Unfortunately Audacity does not go below 0.1 Hz for the simple high pass filtering. Maybe some other software could do it. Or maybe Mani would dare to take a soldering iron to his RME?

@pkane feature request :)

The combination of relatively high input impedance (47 kOhm) with 47 μF coupling capacitors
results in an extremely low cutoff frequency of (calculated) 0.07 Hz.

The ADI-2/4 Pro SE can actually use this low cutoff frequency effectively - many other converters
could not, because the analog input section has a low DC offset, which is detected by the ADC,
and shows up in the recorded audio signal as a fixed DC component. If this offset is too high, click
noises occur at the beginning, at the end, during editing etc.. The ADI-2/4 Pro SE is factory ad-
justed to minimum DC offset, typically -90 dBFS - problem solved.
The DC offset is usually eliminated in the digital domain via a high-pass with 1 Hz cut-off frequency
within the ADC. However, the ADC of the ADI-2/4 Pro SE does not provide such a filter. Therefore
a filter developed by RME is used, in the FPGA, which has a cut-off frequency below 0.5 Hz and
very low phase deviation. This filter is active by default. The DC component is thus reduced to
below -130 dBFS.

I'm not sure that I'm following the DC removal discussion/suggestion. Here's my recording of original2.wav with ADI-2/4 Pro SE with DC filter turned off, matched in DeltaWave. Where's the DC offset that remains to be removed? I can certainly add a separate HP DC filter, but is the DC removal option not sufficient?

1736646402029.png
 
I'm not sure that I'm following the DC removal discussion/suggestion. Here's my recording of original2.wav with ADI-2/4 Pro SE with DC filter turned off, matched in DeltaWave. Where's the DC offset that remains to be removed? I can certainly add a separate HP DC filter, but is the DC removal option not sufficient?

View attachment 420433
The feature request was for a very low frequency high pass filter to allow correction for DC blocking filters like the RME ADC uses.

There was also discussion in the thread that there seems to be DC and low frequency in Mani's test file but I don't think it turns out to much affect nulling.
 
The feature request was for a very low frequency high pass filter to allow correction for DC blocking filters like the RME ADC uses.

There was also discussion in the thread that there seems to be DC and low frequency in Mani's test file but I don't think it turns out to much affect nulling.
Are there files that demonstrate the need for such a filter? I’m still not sure why the DC removal setting that’s already there wouldn’t do exactly what you are asking.
 
Are there files that demonstrate the need for such a filter? I’m still not sure why the DC removal setting that’s already there wouldn’t do exactly what you are asking.
See the result I got comparing your RME loopback recording to a 0.7hz high passed Original2.wav (post#159). It improves the results quite a bit. Mani wants to evaluate only the DAC perfomance, so this helps get at that, although of course it won't correct for all of the error that the ADC contributes.
 
See the result I got comparing your RME loopback recording to a 0.7hz high passed Original2.wav. It improves the results quite a bit. Mani wants to evaluate only the DAC perfomance, so this helps get at that, although of course it won't correct for all of the error that the ADC contributes.
I’m still confused. Did you see my comparison using original2 and my ADI-2/4 loopback recording? I don’t see any obvious difference in DC that would require additional filtering. What am I missing?
 
I’m still confused. Did you see my comparison using original2 and my ADI-2/4 loopback recording? I don’t see any obvious difference in DC that would require additional filtering. What am I missing?
it isn't DC. It is that the RME ADC has a 0.07hz high pass filter. So filtering the reference file before comparing gives a better match. So this is an alternative to shorting the RME capacitors like @KSTR has done. It isn't a must have feature for DW since you only have to filter the reference once and can do this with sox. Just would be a little more convenient.
 
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it isn't DC. It is that the RME ADC has a 0.07hz high pass filter. So filtering the reference file before comparing gives a better match. So this is an alternative to shorting the RME capacitors like @KSTR has done. It isn't a must have feature for DW since you only have to filter the reference once and can do this with sox. Just would be a little more convenient.

I see. How much better is the match with 0.07Hz filter?
 
I see. How much better is the match with 0.07Hz filter?
I am away from that computer. You see the spectrum of the delta curve in post #159 where everything is below -100dB. I think peaks were like -80dB or -85dB without correcting the reference. By the way I can't say enough time how great your software is. The people on this site are unbelievable.

I did not say that right, should have said correction for the ADC filter applied to the reference.

edit: @pkane here is what I get for a delta with your ADI loopback of Original2.wav, and then with 0.072Hz HP filter correction (this is right channel I think which was a little better than left). Does this look right to you?
1736684930984.png
 
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I see. How much better is the match with 0.07Hz filter?

Right now, I only have the means to apply a 6dB/octave minimum phase HP filter at 0.1Hz to the reference file using Audacity.

Here's the effect it has on an SU-10 null:
1736673231604.png


The effect on a DAVE null is less so, but there nevertheless:
1736673275180.png


To be clear, the compare files used for these nulls (which have been through the DA/AD chain) have not been manipulated in any way.
 
Are there files that demonstrate the need for such a filter? I’m still not sure why the DC removal setting that’s already there wouldn’t do exactly what you are asking.
The file is here:


I've analysed it and there is quite a lot of energy between the lowest musical content at 37Hz and 0Hz. I filtered everything below 15Hz and changed the pitch to listen to it. Its not musical, more a resonance - perhaps an artefact of the room or recording equipment.

What the OP is trying to do is use DW to compare three DACs which are level-matched using their internal DSP volume controls whilst playing this music. They repeatedly get wildly different results at DC unless they filter the source file
 
What the OP is trying to do is use DW to compare three DACs which are level-matched using their internal DSP volume controls whilst playing this music. They repeatedly get wildly different results at DC unless they filter the source file

I wouldn't say "wildly" different, but certainly different: https://audiosciencereview.com/foru...asurements-using-deltawave.59822/post-2190924

However, this has little to zero effect on the null values themselves.

To echo what @pjug has already said, we're not trying to fix any issues with DC offset or anything. We're trying to compensate for the effects of the caps at the analogue inputs of the ADC.
 
I am away from that computer. You see the spectrum of the delta curve in post #159 where everything is below -100dB. I think peaks were like -80dB or -85dB without correcting the reference. By the way I can't say enough time how great your software is. The people on this site are unbelievable.

I did not say that right, should have said correction for the ADC filter applied to the reference.

edit: @pkane here is what I get for a delta with your ADI loopback of Original2.wav, and then with 0.072Hz HP filter correction (this is right channel I think which was a little better than left). Does this look right to you?
View attachment 420518

This looks like this filter has a significant effect on the spectrum up to and above 5kHz, and likely some effect on phase, as well.

With DW, the point is to figure out the difference between original and recorded waveforms. If there's a tiny difference below 1Hz without the 0.07Hz HP filter, i don't see any reason to introduce this filter, with all the potential collateral damage to frequency and phase.

For this particular set of files, again, here are the two spectra, original2.wav and recorded loopback, with no additional filters applied, with just tiny differences below 1Hz all the way to DC (blue is the unprocessed original2.wav):
1736693416322.png


Here's the actual difference between the two spectra:
1736694168969.png


Even if you could match up the two spectra a tiny bit better to correct for the difference below 1Hz, it would have an insignificant effect on the actual RMS null (probably a fraction of a dB), and no effect on PKMetric, which is perceptually weighted.
 
They repeatedly get wildly different results at DC unless they filter the source file
Well, so where are there any two files, original and DAC/ADC recorded, that demonstrate these "wildly different results"? The one link to the file you posted doesn't give me a way to analyze it, since DeltaWave requires two files so the delta can be computed. This one appears to be the reference file. I also need the loop-back recording(s), aka, comparison file(s).
 
I wouldn't say "wildly" different, but certainly different: https://audiosciencereview.com/foru...asurements-using-deltawave.59822/post-2190924

However, this has little to zero effect on the null values themselves.

To echo what @pjug has already said, we're not trying to fix any issues with DC offset or anything. We're trying to compensate for the effects of the caps at the analogue inputs of the ADC.

I see the plots, but can't reproduce any of your results. Can you please share some example loop-back-recorded files that demonstrate the benefit of this HP filter? Raw recordings, please, don't apply the filter to these.
 
With DW, the point is to figure out the difference between original and recorded waveforms. If there's a tiny difference below 1Hz without the 0.07Hz HP filter, i don't see any reason to introduce this filter, with all the potential collateral damage to frequency and phase.
I see what you are saying but if you want to isolate DAC effects and remove ADC effects (mainly the low frequency effects from the 0.7Hz RC filter), how do you do that if you don't model the ADC HP filter and correct for it? Or else short the ADC caps. In my opinion, for purposes of what Mani is trying to do, it is perfectly valid to filter the reference with a filter that is based on a known circuit that is described in the RME user manual. This different from the challenge on the gearspace thread where people were attempting to compare overall loopback performance.
 
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