[limpidglitch]

If MQA have improved the A to D process, which is what they claim to have done by reducing time domain distortion, then it certainly can be, even if the result is eventually converted to PCM. I'm not saying it IS better, but there's absolutely no reason why it CANNOT be better.

 
That's exactly what castle says, except he puts the burden of proof on the one set to make a financial gain. You know, the way it is done in markets based on tangible value.

 

[spruce music]

Longer term, the idea is that new music is recorded using MQA. In that case it will be working with analogue input. It will be interesting to see how that sounds.

Something like a visit to a sausage factory I suppose.  Go to a studio to see how recordings get done. There are multiple tracks as in 24 or 48 or big time recordings even more.  Even if you have records of provenance showing exactly which ADC was used how useful is de-blurring time domain when there will be multiple levels of compression, EQ and many other processes performed on the various tracks involved.  Yet you somehow think MQA will be a big boost to fidelity?  Again, it could only help, IF it helps, a tiny minority of recordings. 

 

[RichardTownsend]

Something like a visit to a sausage factory I suppose.  Go to a studio to see how recordings get done. There are multiple tracks as in 24 or 48 or big time recordings even more.  Even if you have records of provenance showing exactly which ADC was used how useful is de-blurring time domain when there will be multiple levels of compression, EQ and many other processes performed on the various tracks involved.  Yet you somehow think MQA will be a big boost to fidelity?  Again, it could only help, IF it helps, a tiny minority of recordings. 

I actually produce my own material and I have an (in the box) studio. I think you're probably right it won't help a lot when there's a lot of digital eq, reverb, compression, use of samples etc, especially in pop and electronica, but for other genres where there are simpler recording techniques it could make a big difference. That would include classical, jazz, folk. Luckily I like those...

 

[castleofargh]

how would it make a big difference? do you have a clear vision of how they would plan to "improve" the signal at the recording?

 

[NeoG]

The divide - as it usually is in audio related matters - sits between the model of reality you prescribe to.
 
One model deals with measured thresholds. No matter where you place MQA on the spectrum, it doesn't matter once it has reached at least the threshold of transparency. The rigorously studied psycho-acoustic model indicates we throw out anything outside of those parameters. In this model, MQA can only be a cash grab or a silly misguided endeavour. Any access to better masters or capturing processes are being locked behind a proprietary wall which can and will be used against you in the name of the mighty dollar.
 
In the other model, there are virtually limitless exponentially reducing benefits to everything you do and every change you make potentially takes you closer to a facsimile of the real thing. The source of truth is your own experiences regardless of whether they line up with popular opinions or are met with derision. We do not fully understand everything and so everything can be a possibility, you need only be open to it. In this model, MQA is another step towards the goal. Any improvements the system can bring will not be turned away, and the creators clearly think the same way about it that you do.
 
One of these models is driven by mass delusion, and the other by a healthy feedback process. But which one is which?

 

[haiku]

Here´s an interesting comment about MQA from Paul McGowan (PS Audio)
 
"I can tell you our point of view, if you’re interested. So far, we too will not support MQA.

It has nothing to do with costs. Which makes sense, if you think about it. What they charge is a big secret and we have yet to be let in on it, but they tell us it won’t “be much”. Fair enough. Let’s call it $10. Heck, call it $20 though I doubt that’s the case.

Do you think $20 more for us or Benchmark matters if it helps sell DACs? Neither we nor any of the manufacturers I know would care about that. Sure, $20 is real money, but in the grand scheme of things, it’s chump change.

I do believe that MQA can make many DACs sound better. You’ve heard it yourself – as has Harley.

But that doesn’t mean IT is a miracle. It only means exactly what it means – that some DACs are helped. Others, like our own, are harmed.

We have taken a great deal of time to play with MQA. Their hardware’s been available to us for quite some time – and in every case our DAC sounds worse. That’s why we haven’t implemented it. There’s nothing anyone could say that would make us add a feature or perceived benefit that would in any way damage what we do. Ain’t gonna happen.

Bob Stuart has an open invitation to show us MQA can work in our DAC. If he or his team does, we’d be the first in line.

It isn’t about cost. It’s about performance. And if the performance isn’t there, then we’re not interested."

This might explain, why others such as Schiit, Linn, Bryston and other High End Companies don´t want to join the MQA Hypetrain. MQA just doesn´t sound good enough with ultra high resolution gear.

 

[RichardTownsend]

how would it make a big difference? do you have a clear vision of how they would plan to "improve" the signal at the recording?

Good question. At the moment, I'm relying on what Bob Stuart has said about this, I have no independent knowledge of my own, and what they have revealed is quite limited. I have seen the plots showing how impulse responses are coded in MQA vs, for example, 192kHz PCM, and the time accuracy is significantly better in MQA. But how this is used to encode better PCM signals has not been revealed. So I don't have a clear vision of how this is done. But I'm hopeful, largely because I've heard comments from respected producers and engineers about improvements to their own material (and most of that was already in the digital domain). But we will only know for sure when the new material arrives.

One thing I've noticed is the the Mytek Brooklyn MQA ADC, which is similar to the one used to encode some of the Warner catalog, samples the incoming audio at 12 MHz. Just a straw in the wind, but it may indicate the timing precision available to the MQA encoder.

In terms of MQA material that has already been encoded from analogue, in Tidal, the omens are good, although it's hard to know what to compare it to. For example, the Doobie Brothers 'Minute by Minute' MQA version says it was remastered in 2016 and so it's likely they have gone back to the original analogue tapes to do this. It sounds wonderful, way better than the standard version. But is this MQA or is it a general improvement in A to D? I Don't know. In general, material on Tidal MQA from the analogue era sounds great, and it really shows up the limitations of early digital recordings. But again we don't know how much MQA has contributed to that.

 

[astrostar59]

If you dig around, there are MQA versions of the same album release as the 44.1 versions. So far, in those cases MQA sounds better to me.
 
On PS Audio's Paul McGowan, it is complicated IMO. I wonder how much of the mystery is the permanent up sampling aspect of his DAC? I don't pretend to know the answer, but many DACs use different filters and up sampling techniques to then output the analogue. Just maybe those very processes are negating the MQA approach in the DAC?
 
This subject is gigantic and goes to the heart of digital and the 2 lane paths manufacturers have taken. Lately more seemingly are releasing non upsampled R-2R implementations, some discrete, others multiship. And some of those R-2R oddly still upsample (C1 DAC). God knows. Isn't in DAC up sampling looked down on by the MQA team? I read a bit of text in that QA on computer audiophile that seemed to be saying the 'time smearing' was part of it. 
 
But I can say honestly, in my small time tests feeding 24/96K in my system, it sounds better than 44.1, and seems as good or better than many of my HD tracks. And there is no cost impact to me to do that which is why I am positive about it. Unless I change DACs I won't come across the 2nd and 3rd stage processing that MQA can do in a compliant DAC. So won't know if it is good or bad.
 
It will interesting to see more feedback from users on MQA coming through. There is too much political intrigue, we need more solid feedback from owners systems IMO.
 
Time will tell if manufacturers pick up on the MQA capability aspect. I am convinced it will sort itself out as these things seem to in time.
 
I personally think it will as many audio fans will want CD or better quality streaming. Eveyone now knows MP3 is bad quality. If they don't those clients won't buy a Tidal hifi subscription anyway. The goal posts have moved and the days we have to pay for MP3 garbage as streaming or iTunes purchases has gone IMO. How the market for bought HD tracks will go I wonder.
 
But oddly I think the vinyl market will keep going, and the recent revival will gain momentum. Go figure.

 

[ThomasHK]

  If you dig around, there are MQA versions of the same album release as the 44.1 versions. So far, in those cases MQA sounds better to me.
 

 
Can you give us a list? On Tidal it's impossible to tell. Thanks!

 

[Pokemonn]

i have heard MQA files vs PCM file at Fujiya Tokyo Headphone festival before.
MQA file sound absolutely noticeable better/smoother/musical than conventional PCM file. it was very easy to heard differences between them.

 

[castleofargh]

 

how would it make a big difference? do you have a clear vision of how they would plan to "improve" the signal at the recording?

Good question. At the moment, I'm relying on what Bob Stuart has said about this, I have no independent knowledge of my own, and what they have revealed is quite limited. I have seen the plots showing how impulse responses are coded in MQA vs, for example, 192kHz PCM, and the time accuracy is significantly better in MQA. But how this is used to encode better PCM signals has not been revealed. So I don't have a clear vision of how this is done. But I'm hopeful, largely because I've heard comments from respected producers and engineers about improvements to their own material (and most of that was already in the digital domain). But we will only know for sure when the new material arrives.

One thing I've noticed is the the Mytek Brooklyn MQA ADC, which is similar to the one used to encode some of the Warner catalog, samples the incoming audio at 12 MHz. Just a straw in the wind, but it may indicate the timing precision available to the MQA encoder.

In terms of MQA material that has already been encoded from analogue, in Tidal, the omens are good, although it's hard to know what to compare it to. For example, the Doobie Brothers 'Minute by Minute' MQA version says it was remastered in 2016 and so it's likely they have gone back to the original analogue tapes to do this. It sounds wonderful, way better than the standard version. But is this MQA or is it a general improvement in A to D? I Don't know. In general, material on Tidal MQA from the analogue era sounds great, and it really shows up the limitations of early digital recordings. But again we don't know how much MQA has contributed to that.

 
I whine about the impulse thing some pages back(or was it another topic?) do not just accept everything they say because it feels ok with the example they show. music is not just how fast an impulse response can stop. that's what they focus on so they show the objective data that makes only that part clear. marketing being marketing.
not only many people would argue that what they propose is improved music, but making impulse responses look good is nothing new under the sun:
https://www.ayre.com/pdf/Ayre_MP_White_Paper.pdf
care for the "white paper" name because while it's a simple example of a few possibilities, the interpretations and claims about improvements are still only the Ayre guy's opinion. not facts.
let's look at page 1 vs page 2. the impulse is clearly better on 2 "wow much time accuracy, such impress, woof woof!" the maximum frequency in the signal basically just goes higher. by the time the attenuation for proper band limiting is reached, we're factually at a higher frequency. if the signal had just been filtered with the first filter but at higher frequency, it would have looked better too.
in the end it's a very basic and simple logic. to reconstruct an ideal impulse, you need to "stack" an infinite number of frequencies, just like with square waves. we see one signal but it's composed of so many sine waves of the right amplitude at the right phase. when you low pass the impulse you remove the highest frequencies, the ones that go up and down the fastest, so the ones that can "draw" vertical lines the best. so you can look at the impulse and think "OMG it's ringing the timing is ruined". or you can think "ok so it's the same sound without some ultrasounds that I can't hear anyway".  both are saying the same thing. if the record is a guitar and a dog whistle, filtering the ultrasounds out will make us lose most of the whistle, all the sounds we weren't hearing anyway. but the guitar is just fine. when we filter out some ultrasounds that's what we're doing, removing some signals that never really concerned us humans in the first place.
 
one argument against my views on this, is about transient response. if we want to draw something with an instantaneous rise, we need crazy high frequencies. that is true. but here is my tiny little problem. do we notice the difference if some ultrasounds are missing and the signal doesn't climb as fast as it should? well I can think of a simple enough testing method. CD vs highres. more frequencies, much improved transient response. again it's a fact. but if it's so beneficial how come we have such a hard time telling both apart in a blind test? to me failing a blind test disproves the need for perfect transient reproduction.
and it's about the same with filters, when they are clearly out of the audible range, we tend to fail blind tests. and it's obvious why, the ringing or whatever we call it, is happening outside our hearing range.  so why do we notice a change when we try different filters on some DACs? why is a MQA dac really changing the sound? why are Ayre bothering with their stuff? well the answer is on those graphs from the previous link. on page 2 the more gentle filter that improves time accuracy, it starts in the audible range and clearly rolls off the trebles in a perfectly audible way. time domain=wins frequency domain=loses. is that better sound? from an objective point of view it's obviously not, even if we were to disregard the extra aliasing. we have a 2 axis signal and we mess one up to make the second look good. then we show marketing stuff about only the one looking good. and this concept is used over and over by everybody including MQA. in fact even the compression process of MQA is losing bit depth for more samples. they can't pull resolution out of a hat, so they move it around from one variable to the next.
 
back to the Ayre paper again, the third page is a well known, well liked kind of filter for the audiophile. because the ringing is delayed and many people think pre ringing is unnatural(fair enough). here there is phase shift so it's a bad filter from a time perspective, ringing plus shift. yet many people say that they like it better (I have a DAC working that way and one working like the first page, I can't tell the difference. but hey I'm not golden ear and 16khz is about as far as I go ^_^).  so already we have many people who do not agree that time should be the only focus right here in the audiophile world.
 
the last page on the PDF isn't too far off from the meridian apodizing filter concept (included in MQA DACs). the PDF forgets to mention how it still has most of the elements deemed negative in page 2. like upper frequency roll off and extra aliasing(=more distortions), and some phase shift from the page 3 example. because like MQA, Ayre focus on what is important to them, not to the signal fidelity. they subjectively find that it's good. and why not, subjectivity is taste, some like justin bieber, some like celine dion. IMO those choices are just that, choices and cannot be called higher fidelity.
 
 
/!\ warning etc.
Now this is pure speculation, but the ideas I can think about for the ADC/recording bit:
we can take note of where the band limiting is done based on the ADC, then reencode the signal in MQA applying a more gentle filter starting at lower frequency(so cut the end again to change the shape of the cut). something along the idea of page 4. that way the so called "bad" filter that was picked by professionals because they're idiots and only MQA knows better(I assume that's how they think), is mostly cut out of the signal.
I don't know if that's what they wish to do, but it would seem to fit the "make impulses great looking again". the upper freqs that were fine are now rolled off, the distortions levels have increased, there is probably a little phase shift, but hey, the impulse looks great! the consumer can dream of better sound if we only ever discuss time domain stuff.
 
that was my wild guess about how MQA planned to "improve" the sound from ADCs.  the rest of the perceived improved audio I guess goes back to mastering and never needed MQA to be done right(or wrong).

 

[RichardTownsend]

I whine about the impulse thing some pages back(or was it another topic?) do not just accept everything they say because it feels ok with the example they show. music is not just how fast an impulse response can stop. that's what they focus on so they show the objective data that makes only that part clear. marketing being marketing.
not only many people would argue that what they propose is improved music, but making impulse responses look good is nothing new under the sun:
https://www.ayre.com/pdf/Ayre_MP_White_Paper.pdf
care for the "white paper" name because while it's a simple example of a few possibilities, the interpretations and claims about improvements are still only the Ayre guy's opinion. not facts.
let's look at page 1 vs page 2. the impulse is clearly better on 2 "wow much time accuracy, such impress, woof woof!" the maximum frequency in the signal basically just goes higher. by the time the attenuation for proper band limiting is reached, we're factually at a higher frequency. if the signal had just been filtered with the first filter but at higher frequency, it would have looked better too.
in the end it's a very basic and simple logic. to reconstruct an ideal impulse, you need to "stack" an infinite number of frequencies, just like with square waves. we see one signal but it's composed of so many sine waves of the right amplitude at the right phase. when you low pass the impulse you remove the highest frequencies, the ones that go up and down the fastest, so the ones that can "draw" vertical lines the best. so you can look at the impulse and think "OMG it's ringing the timing is ruined". or you can think "ok so it's the same sound without some ultrasounds that I can't hear anyway".  both are saying the same thing. if the record is a guitar and a dog whistle, filtering the ultrasounds out will make us lose most of the whistle, all the sounds we weren't hearing anyway. but the guitar is just fine. when we filter out some ultrasounds that's what we're doing, removing some signals that never really concerned us humans in the first place.

one argument against my views on this, is about transient response. if we want to draw something with an instantaneous rise, we need crazy high frequencies. that is true. but here is my tiny little problem. do we notice the difference if some ultrasounds are missing and the signal doesn't climb as fast as it should? well I can think of a simple enough testing method. CD vs highres. more frequencies, much improved transient response. again it's a fact. but if it's so beneficial how come we have such a hard time telling both apart in a blind test? to me failing a blind test disproves the need for perfect transient reproduction.
and it's about the same with filters, when they are clearly out of the audible range, we tend to fail blind tests. and it's obvious why, the ringing or whatever we call it, is happening outside our hearing range.  so why do we notice a change when we try different filters on some DACs? why is a MQA dac really changing the sound? why are Ayre bothering with their stuff? well the answer is on those graphs from the previous link. on page 2 the more gentle filter that improves time accuracy, it starts in the audible range and clearly rolls off the trebles in a perfectly audible way. time domain=wins frequency domain=loses. is that better sound? from an objective point of view it's obviously not, even if we were to disregard the extra aliasing. we have a 2 axis signal and we mess one up to make the second look good. then we show marketing stuff about only the one looking good. and this concept is used over and over by everybody including MQA. in fact even the compression process of MQA is losing bit depth for more samples. they can't pull resolution out of a hat, so they move it around from one variable to the next.

back to the Ayre paper again, the third page is a well known, well liked kind of filter for the audiophile. because the ringing is delayed and many people think pre ringing is unnatural(fair enough). here there is phase shift so it's a bad filter from a time perspective, ringing plus shift. yet many people say that they like it better (I have a DAC working that way and one working like the first page, I can't tell the difference. but hey I'm not golden ear and 16khz is about as far as I go ^_^).  so already we have many people who do not agree that time should be the only focus right here in the audiophile world.

the last page on the PDF isn't too far off from the meridian apodizing filter concept (included in MQA DACs). the PDF forgets to mention how it still has most of the elements deemed negative in page 2. like upper frequency roll off and extra aliasing(=more distortions), and some phase shift from the page 3 example. because like MQA, Ayre focus on what is important to them, not to the signal fidelity. they subjectively find that it's good. and why not, subjectivity is taste, some like justin bieber, some like celine dion. IMO those choices are just that, choices and cannot be called higher fidelity.


/!\ warning etc.
[COLOR=B22222]Now this is pure speculation[/COLOR], but the ideas I can think about for the ADC/recording bit:
we can take note of where the band limiting is done based on the ADC, then reencode the signal in MQA applying a more gentle filter starting at lower frequency(so cut the end again to change the shape of the cut). something along the idea of page 4. that way the so called "bad" filter that was picked by professionals because they're idiots and only MQA knows better(I assume that's how they think), is mostly cut out of the signal.
I don't know if that's what they wish to do, but it would seem to fit the "make impulses great looking again". the upper freqs that were fine are now rolled off, the distortions levels have increased, there is probably a little phase shift, but hey, the impulse looks great! the consumer can dream of better sound if we only ever discuss time domain stuff.

that was my wild guess about how MQA planned to "improve" the sound from ADCs.  the rest of the perceived improved audio I guess goes back to mastering and never needed MQA to be done right(or wrong).

Thanks Castle. I'm not really qualified to discuss this, I would like to be, but there you are I did note that Bob Stuart himself has said that MQA currently has timing resolution equivalent to 768kHz PCM so it may well be in a few years that standard PCM of equivalent time resolution will be produced, when bandwidth is cheaper. Certainly I do not believe it's possible to give 96kHz or 192kHz PCM equivalent time resolution to 768kHz PCM whatever is done to it, but what exactly they have done is still a bit of a mystery.

 

[NeoG]

I whine about the impulse thing some pages back(or was it another topic?) do not just accept everything they say because it feels ok with the example they show.

I would also point out that the impulse response has come up before with DSD, but the advertised impulse was a lie because they measured synthetic unfiltered samples without the mandatory output filter.

So you need to take this marketing material with a grain of salt until it can be independently verified.

 

[astrostar59]

  If you dig around, there are MQA versions of the same album release as the 44.1 versions. So far, in those cases MQA sounds better to me.
 
On PS Audio's Paul McGowan, it is complicated IMO. I wonder how much of the mystery is the permanent up sampling aspect of his DAC? I don't pretend to know the answer, but many DACs use different filters and up sampling techniques to then output the analogue. Just maybe those very processes are negating the MQA approach in the DAC?
 
This subject is gigantic and goes to the heart of digital and the 2 lane paths manufacturers have taken. Lately more seemingly are releasing non upsampled R-2R implementations, some discrete, others multiship. And some of those R-2R oddly still upsample (C1 DAC). God knows. Isn't in DAC up sampling looked down on by the MQA team? I read a bit of text in that QA on computer audiophile that seemed to be saying the 'time smearing' was part of it. 
 
But I can say honestly, in my small time tests feeding 24/96K in my system, it sounds better than 44.1, and seems as good or better than many of my HD tracks. And there is no cost impact to me to do that which is why I am positive about it. Unless I change DACs I won't come across the 2nd and 3rd stage processing that MQA can do in a compliant DAC. So won't know if it is good or bad.
 
It will interesting to see more feedback from users on MQA coming through. There is too much political intrigue, we need more solid feedback from owners systems IMO.
 
Time will tell if manufacturers pick up on the MQA capability aspect. I am convinced it will sort itself out as these things seem to in time.
 
I personally think it will as many audio fans will want CD or better quality streaming. Eveyone now knows MP3 is bad quality. If they don't those clients won't buy a Tidal hifi subscription anyway. The goal posts have moved and the days we have to pay for MP3 garbage as streaming or iTunes purchases has gone IMO. How the market for bought HD tracks will go I wonder.
 
But oddly I think the vinyl market will keep going, and the recent revival will gain momentum. Go figure.

 

 

I whine about the impulse thing some pages back(or was it another topic?) do not just accept everything they say because it feels ok with the example they show.

I would also point out that the impulse response has come up before with DSD, but the advertised impulse was a lie because they measured synthetic unfiltered samples without the mandatory output filter.

So you need to take this marketing material with a grain of salt until it can be independently verified.

Where does an R-2R NOS DAC without a filter sit in all this? Audio Note, Zanden, others. If we go high resolution do we even need a filter? And do we need to upsample at all if said filter is requiring that to implement a harsh brickwall filter near the limit of Redbook. I don't fully understand it. I like some DS DACs which have filters and upsample to insane levels, but I also like (prefer) ones that don't upsample at all. IME and YMMV.
 
Maybe, just, getting access to higher resolution and 'clean' copies of some of those classic studio masters, and the label / studios using current tech not early digital tech when CD first came out, we are getting to the point where the way a DAC operates can be simplified? i.e play the file bit perfect as is. I have no idea, this is my theory thrown out there for comment. But I do know I like what I hear on MQA and Tidal so far. It is good, a move in the right direction IMO and to me is no extra cost (with a standard hifi account).

 

[NeoG]

  Where does an R-2R NOS DAC without a filter sit in all this? Audio Note, Zanden, others. If we go high resolution do we even need a filter? And do we need to upsample at all if said filter is requiring that to implement a harsh brickwall filter near the limit of Redbook. I don't fully understand it. I like some DS DACs which have filters and upsample to insane levels, but I also like (prefer) ones that don't upsample at all. IME and YMMV.

 
At the moment we don't really know anything about what happens in hardware when the MQA process kicks in. The mechanism in which they claim to achieve greater "time resolution" is unknown to the general public. I don't think anyone outside of the creators of MQA could answer that question right now.
 
I put "time resolution" in quotes because it's a glossy term that doesn't really describe what they are solving (hence people's scepticism, who are in the know). Dithered digital audio has time resolution out the wazoo even at 44/16. You can position a transient with far smaller spacing than 1 sample and far tighter spacing between channels than you can hear. It's honestly giving people the wrong impression. People's heads explode when they try to imagine it, but them's the breaks.