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Do MQA DACs benefit non-MQA music?

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Sal1950

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Would you be able to explain why there wouldn’t or couldn’t be a change in SQ by the way a MQA DAC handles a signal?
FYI, I'm not ignoring your question here, but a number of more knowledgeable members have joined with comments on the subject. Read the contributions so far by @mansr and @Kal Rubinson minimally.
 
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Maybe I am misunderstanding some things. But a sound in nature doesn’t actually “build up”. But some electronic impulse response filters induce this unnatural build-up or pre-ring. This isn’t as much as a problem for the post-ring. But I’m just trying to learn, so...
This is something that really, really annoys me. I totally get that, if you know nothing about digital audio, an impulse response from a linear phase (FIR) reconstruction filter looks like any sound you play through that filter will have some sort of pre-echo. And magical thinkers like Hans Beekhuyzen and his peers will dream up the weirdest stuff around that misunderstanding.
Pre-ringing is not a real problem and minium phase filters are not the solution to that imaginary problem. First of all, that "ringing" is way above the audible range in frequency and extremly low energy, and second, and I cannot stress this enough, the sampling theorem states that the perfect reconstruction filter for a PCM signal is the steepest FIR-Filter possible. That impulse response is not some envelope shaping effect that is applied to your music, it is the impulse response of the mathematically perfect filter to restore a digitally sampled signal. Also MQA is a scam.
 

dananski

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Or watch the obligatory Monty video:

Thanks for the link, that was really well demonstrated. It's a bit of a tangent but could I ask something here? I keep wondering this whenever I think about digital-analogue signal reconstruction and the linked video reinforced a lot of my understanding but didn't touch on my main concern.

I believe I understand from Shannon's sampling theory how it's possible to reproduce an analogue signal from the digital samples that is perfect up to the Nyquist frequency (and bit depth). But when you capture the analogue signals in the recording process, surely the samples you take are in the time domain and will have some of their amplitude contributed from frequencies beyond the sample rate. So when later reproducing the signal up to Nyquist frequency, i.e. a lower frequency, the samples contain unknown amounts of unwanted contributions from higher frequencies. So how can the reproduction be true to the original - even within the band we care about?

I don't think this is guaranteed to be naturally filtered out simply by being an input above the sample rate. As a thought experiment to eliminate that, you could throw a few cycles of a high frequency signal on top of the one you're recording at the time of a single sample and cause that sample to be off by an arbitrary amount.

My own hypotheses would be:
  • The high frequencies present when recording are typically too small in amplitude to make any noticeable difference to the samples.
  • The samples are not taken instantaneously by the ADC but are continuous capture. Like taking photos with exposure time = 1 / sample rate rather than instant photos. This would be like blurring of a video feed, but here it just blurs out anything high frequency.
  • Some kind of magical trickery allows you to fix the samples after they are taken.
This ended up being a longer tangent than I originally intended and perhaps I should've started a new thread. But if you have seen/read lots of technical material on audio engineering, perhaps you can direct me to an appropriate source?
 
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I'm not even sure that MQA benefits MQA ... it just seems like such a bad idea and poorly executed too.
 

Don Hills

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...
I believe I understand from Shannon's sampling theory how it's possible to reproduce an analogue signal from the digital samples that is perfect up to the Nyquist frequency (and bit depth). But when you capture the analogue signals in the recording process, surely the samples you take are in the time domain and will have some of their amplitude contributed from frequencies beyond the sample rate. So when later reproducing the signal up to Nyquist frequency, i.e. a lower frequency, the samples contain unknown amounts of unwanted contributions from higher frequencies. So how can the reproduction be true to the original - even within the band we care about?
...

A requirement of the sampling theorem is that all components of the input signal must be below half the sampling rate. To satisfy this, all frequencies greater than half the sampling rate must be filtered out before sampling. Your scenario thus cannot occur.

https://en.wikipedia.org/wiki/Anti-aliasing_filter
 

pablolie

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The D in DAC doesn't matter as much as the AC. The coding is digital. If lossless - great. If not - check how much it really matters. MQA gets maligned by some, celebrated by others... but it's just a digital encoding system, and if the "_AC" is up to it, it doesn't matter...
 

dananski

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A requirement of the sampling theorem is that all components of the input signal must be below half the sampling rate. To satisfy this, all frequencies greater than half the sampling rate must be filtered out before sampling. Your scenario thus cannot occur.

https://en.wikipedia.org/wiki/Anti-aliasing_filter
Ah cool, so this is actually implemented as part of the ADC process, potentially with oversampling then discarding higher frequencies, but I suppose there are also physical measures that can reduce HF noise, like electronics or the limited frequency response of a microphone. I suppose even though it can't be perfect, the only things that'll get through into the resultant samples would be things both higher in frequency than the oversampling rate, and high enough in amplitude to still impact a sample after heavy attenuation. So perhaps high intensity electrical/radio interference, hopefully the sort of thing that couldn't happen in a studio unless you were trying to make it.
 

tonycollinet

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Ah cool, so this is actually implemented as part of the ADC process, potentially with oversampling then discarding higher frequencies, but I suppose there are also physical measures that can reduce HF noise, like electronics or the limited frequency response of a microphone. I suppose even though it can't be perfect, the only things that'll get through into the resultant samples would be things both higher in frequency than the oversampling rate, and high enough in amplitude to still impact a sample after heavy attenuation. So perhaps high intensity electrical/radio interference, hopefully the sort of thing that couldn't happen in a studio unless you were trying to make it.
As stated by @Don Hills the frequencies above half sampling are filtered out before sampling. This is done using an analogue low pass filter. It is not sufficient to rely on limited bandwith of (eg) microphones since then you'd have to adapt the sampling rate for the source of highest frequencies in the signal chain.

Though as stated in the article, it is possible to oversample compared to the required bandwidth, then apply a lower band low pass filter digitally. It is still a requirement though to have an analogue low pass filter to cut out frequencies higher than half the over sampling rate.
 
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Mnyb

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This is something that really, really annoys me. I totally get that, if you know nothing about digital audio, an impulse response from a linear phase (FIR) reconstruction filter looks like any sound you play through that filter will have some sort of pre-echo. And magical thinkers like Hans Beekhuyzen and his peers will dream up the weirdest stuff around that misunderstanding.
Pre-ringing is not a real problem and minium phase filters are not the solution to that imaginary problem. First of all, that "ringing" is way above the audible range in frequency and extremly low energy, and second, and I cannot stress this enough, the sampling theorem states that the perfect reconstruction filter for a PCM signal is the steepest FIR-Filter possible. That impulse response is not some envelope shaping effect that is applied to your music, it is the impulse response of the mathematically perfect filter to restore a digitally sampled signal. Also MQA is a scam.

Yes and that actually no such ringing would occur with any music signal :)
You have to specifically provoke this (to test the impulse response as a proxy for how the filter is implemented , a part of testing ) with an impulse that is not a "legal" signal if correctly sampled and bandwidth limited .
 

Roland68

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Reputable YouTube reviewers and in forums it has been said that MQA equipped DACs have the ability to improve non-MQA files because of the MQA DAC’s impulse response filters used in the MQA rendering process. I have noticed regular FLAC files seem to have more depth, better separation, better soundstage, better dynamics, and bass response when using DAC’s certified for MQA. Is there any science behind these claims than an MQA DAC can improved regular FLAC sound quality?
Maybe it's something else?
I listened to four different DACs that came onto the market without MQA and then got MQA in the next generation, so a total of 8 DACs.
Actually, I wanted to find out for myself whether MQA has any advantage for my sound experience, which I can only deny.
But there was another difference for me. The DACs with MQA had 2 advantages for me, the resolution and clarity (with non MQA files) was "better", not worlds, but it was noticeable.
The second point was more interesting to me. The slight differences/sensitivities (also here only nuances) with different USB cables no longer occurred with the MQA devices, it didn't matter at all.
And the only difference was that the non-MQA DACs used an XMOS XU208, while the MQA DACs used the XMOS XU216.

Even if many people claim that the XU208 is the end of the road, maybe the XU216 will bring a little more "cleanliness" to the USB interface.

With all other inputs (i2s, SPDIF, AES) I could not find these differences between MQA and non-MQA devices.
 

BDWoody

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Actually, I wanted to find out for myself whether MQA has any advantage for my sound experience

So, you set up a proper listening test, so you'd have meaningful results?

The slight differences/sensitivities (also here only nuances) with different USB cables no longer occurred with the MQA devices, it didn't matter at all.

Nuances between USB cables? I see...

Have you seen this? Might be helpful.

 

firedog

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I don’t think the MQA DAC unfolding process is even triggered by non-MQA content. I have a MQA DAC dongle and the MQA light does not activate for Any non-MQA content. So what you are speculating seems rather improbable at best. But this is just my opinion and about as trustworthy as your source.
It's not an issue of folding or unfolding. MQA DACs use specific MQA filters. Many of them also may playback non-MQA material with those same filters, even if the light isn't activated.
Sounds like you don't actually know how MQA works.
 

firedog

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I’m referring to minimum-phase filters which have only a post-echo in their impulse response. Although you have other undesirable artifacts.

The way I understand it is if you increase sample frequenciy to 192kHz or 384kHz you reap dividends for impulse response filters, because as filter slopes relax, so does the degree of pre-or post-ringing.

I also understand higher sampling frequencies improve time domain rather than frequency domain. A number of scientific studies have found human auditory system is extremely sensitive to timing. We can discern tiny differences in timing between sounds played into both ears. I believe it was 6µs that we’re able to discern differences. 16/44.1 only gives us 23µs time resolution. This improvement in timing is where I believe MQA benefits music.
Those are all audiophile truisms. And they aren't true and show misunderstanding of how digital audio works.
Hi-res or upsampling might sound slightly better, not because of ringing of any type, but b/c there could be fewer aliasing artifacts introduced during the DA conversion.
The idea that CD doesn't give enough time resolution is also false. The people saying that don't know how to do the calculations.
MQA doesn't improve timing of reproduction. Just b/c a claim is made, doesn't mean it's true. MQA don't actually define what they mean by blur and can't show that MQA reduces it.
 

AdamG247

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We are absolutely Not starting another MQA thread to rehash all the previous arguments. Thread closed.
 
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