Hugo M Scaler by Chord Electronics - The Official Thread
Jul 10, 2023 at 1:40 PM Post #17,386 of 18,448
Out of scientific curiosity I am wondering how many people are actually listening to the M-Scaled music while writing on head-fi about high fidelity reproduction of sound :wink:
Hmm, I am not sure what information you are actually asking for?
If you mean WHILE typing OR basing what one writes on the Hugo Mscaler thread on actually listening to music VIA an Mscaler or not? My answer to "no 1" would be a firm NO.
I do not listen to music while typing messages. I don't think I ever have in any context here or elsewhere.
When I listen to music, I listen to music.
I think I have made it clear though that I mostly but not always, listen to music via an Mscaler and also find it essential for my musical enjoyment and the increased level of HIGH FIDELITY it brings to my listening experiences of mainly well recorded unamplified instrumental music of several genres, not only Western Classical or using another more modern term Art Music , my two main choices. But sometimes also other music just for fun. Having travelled many countries of the World I also enjoy Latin American and lots of Asian music, both Popular, Folk and Classical Music.


Cheers CC
 
Jul 11, 2023 at 5:45 AM Post #17,387 of 18,448
Thanks for the recommendations. I can’t find Anna Thorwaldsdottir but maybe I am not searching right (on Qobuz or Tidal)? I have been to live concerts and heard instruments, but probably enough enough or many types/manufacturers/spaces to be able to discern the different tones and timbres. Serious question while I am listening to some of your violin recommendations: when you listen to this music reproduced, does your enjoyment come from how well the reproduction meets the standard for live that you know, or is it more direct enjoyment of what you are hearing? (If that even makes sense.)
She might be easier to find if you spell her name correctly: Thorvaldsdottir.
 
Jul 11, 2023 at 7:31 AM Post #17,388 of 18,448
very confused by all this “high fidelity” talk. Mscaler is an upsampling box that adds no new relevant information other than allowing the use of a more gradual sloped reconstruction filter. However, given Chord dacs already have thousands of taps on that reconstruction filter the additional benefit from additional upsampling is next to zero in terms of making the sound “high fidelity“. Science says you only need to sample at 2x the band-limited frequency to capture an analog signal perfectly in discrete digital format. I am with science on this one.
 
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Jul 11, 2023 at 7:44 AM Post #17,389 of 18,448
She might be easier to find if you spell her name correctly: Thorvaldsdottir.
Oops, with a blush I have to confess that I am guilty of spreading the wrong spelling with a w instead of v in her name.
Sloppy of me, and my only feeble excuse is that I must have subconsciously been thinking of the German word Wald which means forest.
Or at least it used to mean forest ,but who knows maybe just as High Fidelity seems to have taken on a new meaning for some here, maybe Wald means something else too for some these days?
Anyway Mea Culpa and yes Anna Thorvaldsdottir's music can be found on Quboz and also on DGG, Youtube and other platforms online. Tidal? I don't know ,it is a service I do not use myself.
And Andrea Tarrodi's work Liguria was performed at the BBC Proms 2017 if my memory serves me right?
Cheers CC
 
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Jul 11, 2023 at 11:26 AM Post #17,390 of 18,448
very confused by all this “high fidelity” talk. Mscaler is an upsampling box that adds no new relevant information other than allowing the use of a more gradual sloped reconstruction filter. However, given Chord dacs already have thousands of taps on that reconstruction filter the additional benefit from additional upsampling is next to zero in terms of making the sound “high fidelity“. Science says you only need to sample at 2x the band-limited frequency to capture an analog signal perfectly in discrete digital format. I am with science on this one.
It doesn't claim to add new information.
It better reconstructs the existing information as described by Nyquist theory.

The science/theory behind Nyquist is that you can perfectly reconstruct the original signal upto half the sampling frequency if and only if you can PERFECTLY band-limit, ie: instantly and infinitely attenuate at that frequency.
In practice, this isn't possible as it'd require infinite computing power, so we have to compromise.

With the limited compute power available on a DAC, filters typically either start rolling off before Nyquist, or they do not sufficiently attenuate by Nyquist at all.

The general consensus (though this is NOT something that has been scientifically tested and existing evidence actually contradicts it) is that we don't need 'perfect' band limiting anyway because so long as you don't attenuate anything under 20khz, but attenuate 'fully' (by -96dB for 16 bit or -146 for 24 bit) by nyquist, then you will eliminate any unwanted imaging, because no content above nyquist remains, and are doing so in an 'audibly transparent' way, because we can't hear above 20khz.

In practice though, even if this were true, very few DACs actually do this. In fact none of the stock AKM or ESS filters do. Even from companies commonly touted as being 'measurably transparent' such as topping, they don't adhere to Nyquist theory as they all either roll off too early or do not attenuate by the nyquist frequency.

The debate about whether a 'sufficiently transparent' reconstruction (ie: everything under 20khz passed unaltered but nothing remaining above 22.05khz) vs a 'high performance' reconstruction like PGGB, HQP, MScaler etc that attenuates EXTREMELY steeply and can do so extremely close to the nyquist frequency, actually matters, is a different debate. (And I'd recommend looking into the research on that, whilst humans cannot hear above 20khz, we have been shown to be able to perceive time domain differences at 10uS, something that would require a frequency domain bandwidth of about 100khz to describe.)

But in terms of the 'science', actually basically no consumer DACs properly adhere to Nyquist theory by themselves in the first place.
 
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Jul 11, 2023 at 11:57 AM Post #17,391 of 18,448
You have the means, perhaps you can do the experiment yourself and make a YouTube video. hook up a signal generator to an oscilloscope and then output to an A/D and then right back to analog via D/A and then hook the output to another oscilloscope. You will see without fail the signal at the other end of the D/A is a perfect sine wave up to 22khz if you sample at 44.1khz. Any reasonably modern A/D and D/A will do.

Also take a look at this - you don’t need ridiculously large number of coefficients for your FIR filter. http://dx.doi.org/10.1109/82.877143
 
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Jul 11, 2023 at 12:06 PM Post #17,392 of 18,448
As part of that test perhaps you could also let us know if that sine wave gets your feet tapping. :beerchug:
 
Jul 11, 2023 at 12:14 PM Post #17,393 of 18,448
It doesn't claim to add new information.
It better reconstructs the existing information as described by Nyquist theory.

The science/theory behind Nyquist is that you can perfectly reconstruct the original signal upto half the sampling frequency if and only if you can PERFECTLY band-limit, ie: instantly and infinitely attenuate at that frequency.
In practice, this isn't possible as it'd require infinite computing power, so we have to compromise.

With the limited compute power available on a DAC, filters typically either start rolling off before Nyquist, or they do not sufficiently attenuate by Nyquist at all.

The general consensus (though this is NOT something that has been scientifically tested and existing evidence actually contradicts it) is that we don't need 'perfect' band limiting anyway because so long as you don't attenuate anything under 20khz, but attenuate 'fully' (by -96dB for 16 bit or -146 for 24 bit) by nyquist, then you will eliminate any unwanted imaging, because no content above nyquist remains, and are doing so in an 'audibly transparent' way, because we can't hear above 20khz.

In practice though, even if this were true, very few DACs actually do this. In fact none of the stock AKM or ESS filters do. Even from companies commonly touted as being 'measurably transparent' such as topping, they don't adhere to Nyquist theory as they all either roll off too early or do not attenuate by the nyquist frequency.

The debate about whether a 'sufficiently transparent' reconstruction (ie: everything under 20khz passed unaltered but nothing remaining above 22.05khz) vs a 'high performance' reconstruction like PGGB, HQP, MScaler etc that attenuates EXTREMELY steeply and can do so extremely close to the nyquist frequency, actually matters, is a different debate. (And I'd recommend looking into the research on that, whilst humans cannot hear above 20khz, we have been shown to be able to perceive time domain differences at 10uS, something that would require a frequency domain bandwidth of about 100khz to describe.)

But in terms of the 'science', actually basically no consumer DACs properly adhere to Nyquist theory by themselves in the first place.
Audible differences? That I guess is the question that seems to surround not just this claim, but other such claims. We can use theoretical models to demonstrate that at "some level" change is occurring, we can even describe the direction of the change, but when it comes to the intensity of the change, does it reach absolute and obvious audibility? I am sure you are well versed in all such debate positions so I will simply leave it there as I have seen how quickly threads go OT.
 
Jul 11, 2023 at 12:55 PM Post #17,394 of 18,448
You have the means, perhaps you can do the experiment yourself and make a YouTube video. hook up a signal generator to an oscilloscope and then output to an A/D and then right back to analog via D/A and then hook the output to another oscilloscope. You will see without fail the signal at the other end of the D/A is a perfect sine wave up to 22khz if you sample at 44.1khz.
It isn't the same upto 22khz. That'd only be true if we have a reconstruction filter that doesn't attenuate anything at all below the Nyquist frequency.

This is somewhat easy to do, but usually means you're then not attenuating by Nyquist, because you're rolling off too late. So whilst your amplitude for signals under 20khz might be mostly correct, you're also producing unwanted images of the signal and not accurately reconstructing. Filters are a game of tradeoffs unless you throw more and more compute power at the problem (which is kinda what Chord does).

Let's take an example DAC, in this case the ESS 9039 based SMSL SU-9 Pro.
Selecting the 'fast linear' filter, we can see that a 1khz 0dBfs sine outputs 4.192V. I've set the dBrA ref level to this so we can easily see how gain changes.

1689091622108.png


If we now go for 17khz, you can see that the level is sliiiightly lower, also there are some unwanted products showing up above the fundamental. Some of which (the ones at 34khz and 51khz) are just the harmonic distortion of the DAC, but the other one is actually due to the incorrect reconstruction filter.

AudioPrecision.APx500_e7VThn5ouY.png


And if we go for 22khz the main signal is a tiny bit lower as the filter continues to gently roll off, though again only by a tiny amount so that indicates the filter is not attenuating things much at all here, though we now have a substantially higher distortion product just above the fundamental as well.

1689091782771.png


So why is this? Well it's because the filter is not actually adhering to Nyquist, as it is not attenuating fast enough. We can show this by putting 44.1khz white noise through the DAC and observing the output:

AudioPrecision.APx500_93BOFLu515.png


It's hardly attenuating anything under 20khz, as we saw earlier, which is great, but it's not attenuating fully until over 26khz, meaning it's not adhering to Nyquist and is not correctly eliminating erroneous imaging. The cursor at 22.05khz (Nyquist frequency) shows that it's basically not attenuating at all there yet.

This DAC does not actually have any filters that correctly adhere to Nyquist theory. But it does have one filter that attenuates a bit steeper called 'apodizing'. Let's look at the filter response for that (I've changed the Y axis scale to dB for this part to make things clearer in the next step).

AudioPrecision.APx500_egTDJCspuO.png


This one attenuates by 24khz, which is better, though still not correct as by 22.05khz we've only attenuated about 11dB. We can also see that it does this at the expense of rolling off sooner. According to the graph, a 21.5khz signal should be about 5dB lower than a 20khz one, and sure enough:

AudioPrecision.APx500_a55XTuP0Te.png


20khz produces 4.111V.

AudioPrecision.APx500_R4fDVJDRSW.png


but 21.5khz produces 2.272V, 5.15dB lower.
Also note that separately from the amplitude thing, the unwanted products above the fundamental are significantly lower than what we saw with the previous filter, at around 10uV instead of 350uV that we saw previously. This is due to the reconstruction filter more accurately eliminating content above Nyquist.

So, the question is then what happens if we use as close to a 'perfect' reconstruction filter as possible?
Well, I'll do this with PGGB. Same DAC, I'm just upsampling the files to 768khz using PGGB and then feeding that to the DAC rather than just using the DAC's internal reconstruction filter, similar to how the MScaler would. (NOTE: I've set PGGB Gain to -1.0dB for all three files to prevent intersample overs clipping)

Here's 20k:
AudioPrecision.APx500_aLDMLEsK74.png


Here's 21.5k:
AudioPrecision.APx500_8NXoLH8zzQ.png



So whereas we were getting differences of several decibels with the previous filters, this one we're seeing about 0.02dB even that close to Nyquist. Additionally note that the unwanted distortion product we'd seen above the fundamental previously is completely nonexistant here.

And if we look at the whitenoise test:

1689094428724.png


About as close to perfect attenuation as we can get.
You can characterize how a filter will attenuate things by looking at the whitenoise response. A 'perfect' filter will do what the graph above shows. Everything below Nyquist left unaltered and everything above it completely attenuated.
Some would argue that a 'sufficient' filter would be one that leaves everything below 20khz unaltered and fully attenuates by 22.05khz, but very few DACs actually manage to even do that anyway. The vast majority of DACs on the market have filters that do not properly reconstruct the signal according to Nyquist theory. Chord's is one of the few that does.

Chord DAVE+MScaler filter response measured at the analog output of the device:

1689094912073.png
 

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Jul 11, 2023 at 1:04 PM Post #17,395 of 18,448
Audible differences? That I guess is the question that seems to surround not just this claim, but other such claims. We can use theoretical models to demonstrate that at "some level" change is occurring, we can even describe the direction of the change, but when it comes to the intensity of the change, does it reach absolute and obvious audibility? I am sure you are well versed in all such debate positions so I will simply leave it there as I have seen how quickly threads go OT.
This is something I have a video coming on in future.
The TLDR is yes, I personally am able to reliably demonstrate an audible difference between a filter that adheres to the 'sufficient' adherence to Nyquist described in my above post vs a 'close to perfect' one, to a more than statistically significant degree.

What the average hifi enthusiast, average joe off the street etc would/wouldn't be able to hear I've no idea. But in my view if something CAN be audible even if it ends up being something you have to train yourself to be able to hear, it's something that people can therefore have valid reason to want to address/improve.
I don't think many people on this forum are here because they're happy with 'good enough'. Else they'd have bought some decent wireless headphones and called it a day.

The nice thing about filters is that because you can do it digitally, you can demonstrate that with an ABX test that can be remotely verified. You don't actually need special hardware like an MScaler to do it and tools like the foobar ABX plugin plus HQPlayer and PGGB are helpful for this.
 
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Jul 11, 2023 at 1:07 PM Post #17,396 of 18,448
Apples and oranges. ES9038 only has 128 coefficients for their FIR filter. Dave has 164k coefficient. Why don’t you do the math and tell us amplitude variation based on 164k tap FIR filter?

1689095721743.png


would be even better if you can just show us the FIR with 128 taps and 164k taps in matlab.
 
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Jul 11, 2023 at 1:24 PM Post #17,397 of 18,448
Apples and oranges. ES9038 only has 128 coefficients for their FIR filter. Dave has 164k coefficient. Why don’t you do the math and tell us amplitude variation based on 164k tap FIR filter?



would be even better if you can just show us the FIR with 128 taps and 164k taps in matlab.

DAVE Standalone:
1689096163520.png


DAVE with MScaler (Note X-axis scale is zoomed in because filter steepness is considerably better)
1689096208429.png


This was your original comment:
hook up a signal generator to an oscilloscope and then output to an A/D and then right back to analog via D/A and then hook the output to another oscilloscope. You will see without fail the signal at the other end of the D/A is a perfect sine wave up to 22khz if you sample at 44.1khz. Any reasonably modern A/D and D/A will do.
I've demonstrated in the previous post that this is not the case. How far one needs to go in terms of reconstruction filter accuracy is up for debate because there is no conclusive research on it.
You can make your own assumptions, but please do understand that existing evidence supports the idea that human perception can discern incredibly small time domain differences and as a result one could reasonably argue that almost ANY reconstruction filter improvement with redbook content could in theory be audible.
 
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Jul 11, 2023 at 1:28 PM Post #17,398 of 18,448
Your x scale does not line up between the two outputs. Please revise.

also, I note we are talking about beyond 22khz flat pass band at this point and I can’t hear 22khz. Not sure about you.

Also show us the 128 tap filter you used in your first example.
 
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Jul 11, 2023 at 1:33 PM Post #17,399 of 18,448
Your x scale does not line up between the two outputs. Please revise.

also, I note we are talking about beyond 22khz flat pass band at this point and I can’t hear 22khz. Not sure about you.

Also show us the 128 tap filter you used in your first example.
As said, the X-Axis scale was changed, feel free to stretch the image if preferred, or just look at the labelling.
I do not have the DAVE + MScaler here to retest currently.

The 128 tap filter completely depends on what said coefficients are. But just look at any ESS DAC filter results for examples
 
Jul 11, 2023 at 1:38 PM Post #17,400 of 18,448
I note we are talking about beyond 22khz flat pass band at this point and I can’t hear 22khz. Not sure about you.
Neither can I, but the audible effect is not due to us hearing 22khz content directly from a frequency domain perspective (humans cannot hear above generally 20khz). But rather the time domain information that then provides.

As stated earlier, whilst we cannot hear frequency domain information above 20khz, there is evidence showing we can hear time domain differences as low as 10uS which would require a frequency domain bandwidth of about 100khz to describe.

https://phys.org/news/2013-02-human-fourier-uncertainty-principle.html
https://www.ncbi.nlm.nih.gov/pmc/articles/PMC3663869/
 
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