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Anyone able to ABX MQA audio, through an MQA certified DAC V.S. similar performing DAC without MQA?

Tks

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I understand without the full unfolding process, you're not being given access to the "full" file itself supposedly, and are getting virtually less than CD quality in term of bit depth from the audio being played.

But as dumb as MQA may be, I was looking at people asking whether the D90 is worth it, over the D70S DAC's from Topping (the latter which are all MQA certified I believe). And was wondering if it would be better to go for the cheaper D70S, and at least reap whatever parts of MQA encoded files are holding back if you don't have a DAC that can properly decode the entire file properly?

Seeing as how I there's no one out there that's going to ABX the two DACs in general from a performance perspective anyway?
 

Soniclife

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I doubt anyone can, but to clarify what exactly are you wanting compared.
1) Full MQA in DAC Vs no MQA unfolding.
Or
2) Full MQA in DAC Vs first unfold in software sent to no MQA DAC.

To me option 2 is the most relevant, as the tidal app and roon both apply the first unfold in software. I see zero chance of option 2 being audible.
 
OP
Tks

Tks

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Either would be fine. But ideally option 1, seeing as how I don't care for Roon, and if I wanted to stream I'd use Spotify instead of Tidal. It also gives the ABX more credence if we're at least able to establish people can hear the difference in the most favorable scenario (the scenario where no unfolding MQA magic is happening at all).
 

kevin1969

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I have an MQA DAC and it's hard to tell the difference between the CD version and the HR version on Tidal. If I ever come across another DAC I will try it out.
 

Soniclife

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Either would be fine. But ideally option 1, seeing as how I don't care for Roon, and if I wanted to stream I'd use Spotify instead of Tidal. It also gives the ABX more credence if we're at least able to establish people can hear the difference in the most favorable scenario (the scenario where no unfolding MQA magic is happening at all).
What does Spotify and MQA have to do with each other?
 
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Tks

Tks

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What does Spotify and MQA have to do with each other?

There are no MQA titles on Spotify, and I don't use Roon, so that combination of my personal circumstance doesn't allow me to ABX partial MQA unfold. I have an LG phone that allows for full unfold, but I personally couldn't ABX between it enabled to fully unfold vs disabled.

MQA ?
Really ?

I mean the reason I ask is, if MQA is played without any unfolding, I was wondering if anyone at least could attest to a difference as I couldn't personally.

You would think with the claims made about MQA, people would at least be able to ABX a difference with back to back comparisons (not sound quality differences, but any tells between one file and the other - I'm not even asking if you could tell which is which, I'm simply wondering if people are capable of doing even less than that.. basically if they're able to tell there's a audible difference between the files at all).

I'm not an MQA proponent or something. I've gone on telling people how MQA basically is lossy nonsense, and because of such, couldn't really compare to lossless 16-bit FLAC. So I'm extending an open mind about someone hearing this difference in some fashion due to there being a blatant bit-depth loss between the two (you lose like what? 3-bits with MQA?). So I would at least expect someone to hear some audible difference of any kind. Let alone a difference that gives away which is which.
 
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norcalscott

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@Tks I'm a longtime lurker here, learning a lot from these forums.

I've tried to do some ABX with Tidal MQA versus FLAC (that I own) of the same title (with same remaster version). This is actually difficult to do as far as getting the same exact version. I've been able to find a few tracks that allow for this to an extent.

My equipment is an LG G7 on which I typically use USB Audio Player Pro with the MQA license applied and all the settings that are recommended for the LG Quad DAC, which, as you mentioned can do the full unfold. I recently just acquired a HiBy R5 Saber, which can also do a full unfold using both USBAPP and the built in HiBy music player. My headphones include Philips Fidelio X2 (originals made in Austria, if that makes a difference), Beyer DT770 80 Ohm / modded cable, and a couple of different pairs of Koss Porta Pros (1 pair with Yaxi pads and 1 pair with modded Grado Pads).

For MQA tracks, I prefer LG G7 or HiBy R5=>Tidal=>USBAPP Bit perfect mode=>Headphone

For non MQA I play a FLAC from storage on either player=>USBAPP Bit Perfect mode=>Headphone

Alternatively, I can play a FLAC as above, but use the LG G7 with the TempoTec Sonata HD DAC

My methodology for doing the ABX is to play these identical tracks several times over and over alternating between MQA and non MQA playout paths.

I'm in my 50's and due to too many concerts in the late 70's and 80's, my hearing is not what you would consider pristine. For me, below are my opinions after more than a few sessions of doing testing:

- I give a slight edge to the sound quality of a signal path that I didn't mention above:
LG G7=>Tidal MQA=>USBAPP Bit perfect mode, doing the first MQA unfold=>TempoTec Sonata DAC (does not do any further MQA unfolds)=>Porta Pros with Grado Pads

To be honest, I don't feel my equipment quality allows for detecting too much difference at all between MQA and non-MQA FLACs of the same remaster. Some of these new remasters that Tidal has released sound phenomenal to me, MQA or not. I'm having a great time re-discovering music that I haven't listened to in some time, which has been remastered on Tidal. I'm not a firm believer in MQA, but I do feel that for streaming audio, what Tidal is doing there is really compelling from a sound quality point of view.

Not sure if any of this is useful information for you.
 

norcalscott

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He's asking about something simpler than that: MQA file with unfold vs the same MQA file without unfold.
Ah, OK. That is relatively simple to do on my R5 Saber. I've done this before but have not done serious AB testing with it.

Effectively, disabling bit perfect mode on USBAPP causes MQA files on Tidal to playback at 44.1 resolution. This is confirmed with the Saber resolution readout in the status bar.
 
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Tks

Tks

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Ah, OK. That is relatively simple to do on my R5 Saber. I've done this before but have not done serious AB testing with it.

Effectively, disabling bit perfect mode on USBAPP causes MQA files on Tidal to playback at 44.1 resolution. This is confirmed with the Saber resolution readout in the status bar.

Yeah this is what I tried. Played back the same file with bit perfect so it engages the full MQA unfolding. And then turnning off bit perfect so it resorts to playing the MQA track without any unfolding. I could not tell a lick of difference.
 

norcalscott

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I haven't been able to see any conclusive difference in my testing between decoded and unfolded MQA tracks, but I saw this post on Reddit today that has some interesting data on the differences between Tidal FLAC (non-MQA) and MQA files. Not completely sure I agree with the results from a listening perspective, but interesting nonetheless. Reddit link
 

Jimbob54

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He's asking about something simpler than that: MQA file with unfold vs the same MQA file without unfold.

That should be fairly easy for anyone with the inclination that uses Tidal and Roon. Not exactly a quick switch but within a few seconds you can disable the MQA core decoder in Roon and then play the "folded" version. Though I have a feeling that something in the unfolding process affects the output volume so might still not be like for like.
 

RayDunzl

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I went to the Florida Audio Expo twice.

I'm certain that I heard MQA and non-MQA, thinking about it later.

Don't ask which rooms were playing MQA and which weren't, because nothing stood out to make me notice either way.

Same with electronics, no piece distinguished itself from other examples.

Speakers, on the other hand, in my opinion, often had interesting signatures.
 

danadam

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I haven't been able to see any conclusive difference in my testing between decoded and unfolded MQA tracks, but I saw this post on Reddit today that has some interesting data on the differences between Tidal FLAC (non-MQA) and MQA files. Not completely sure I agree with the results from a listening perspective, but interesting nonetheless. Reddit link
It was posted here on ASR 4 minutes earlier: https://www.audiosciencereview.com/forum/index.php?threads/testing-mqa-is-it-worse-than-flac.21735/
 

norcalscott

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I went to the Florida Audio Expo twice.

I'm certain that I heard MQA and non-MQA, thinking about it later.

Don't ask which rooms were playing MQA and which weren't, because nothing stood out to make me notice either way.

Same with electronics, no piece distinguished itself from other examples.

Speakers, on the other hand, in my opinion, often had interesting signatures.
I agree with you, just one thing to add from my personal experience using Tidal - I brought over a lengthy playlist from my previous streaming service (GPM) into Tidal - a list of my "favorites" over the years, about 3,000 tracks give or take. I like to just shuffle that while I work, playing through headphones. Many tracks in that playlist are non-MQA, but Tidal put the MQA versions in if it was available when I did the import.

Most of the time I just listen, not looking at my DAP's display, and to my old ears the tracks sound really good. Occasionally, however, I will hear something that is just different, maybe better, maybe more depth, but something that makes me think that this is the best version of the track I have experienced. These are nearly always MQA tracks when I look at the display to check.

Maybe it's just a better master, but there is something there that is making a difference to me. I know this isn't exactly blind testing, but I have had this happen a few dozen times using different headphones. Two recent tracks I noticed this with are Help Me - Joni Mitchell, MQA version and I'll be your lover too - Van Morrison - 2013 Master edition, MQA version. There is an acoustic guitar section in that song that gives me chills when it comes through, really enjoyable.
 

bhobba

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I understand without the full unfolding process, you're not being given access to the "full" file itself supposedly, and are getting virtually less than CD quality in term of bit depth from the audio being played.

I do not believe MQA is dumb. Paul McGowan, a noted critic of MQA, compared it and the original 192k masters. They were very close. He was impressed by how close the two were. As he said, 'After all, this is a pretty amazing process that allows streaming services to send high-resolution audio without degradation – no small feat.' That should be with nearly inaudible degradation from his listening tests.

The problem with MQA is that they claim it is better than the original (I will not go into why that is claimed, but they make the claim). As Paul said, 'If all the hype and hoopla had merely stated the results were indistinguishable from the original, I might be jumping up and down with how close they got it. Few systems have the resolving power of Music Room One, and the fact they got close after folding the music into a smaller file size is quite an achievement.'

The real issue with MQA is not that it is a lousy format - it is, as Paul said, quite an achievement. It is the BS marketing hoopla around it making claims better than the original etc. But the MQA people are hardly alone in that tactic.

Technically it is an interesting way to transmit modern recordings. Mostly the original performance these days is captured using some one-bit high sampling rate such as 10XDSD. However, MQA claims the audible information is contained in a triangle found in figure 7 of the following:

To transmit it efficiently, you first run it through a slow roll-off 32-bit filter flat to 20k. It slowly rolls off to be about 8db down at 48k and keeps on reducing even higher frequencies. Since it is all noise above 48khz, that is not a worry, nor likely is it 8db down at 48khz. It is just conjecture; probably (at least hopefully) they have done blind listening tests to prove it. Suppose you have a 192k record, ing and chuck away every second sample, then you get a 96k stream. But this chucking away has a consequence - information above 48khz is reflected into the 0-48khz region. However,r because of the shallow filter,r this information lies below the noise floor hence will not make any audible difference -you are just changing the noise a bit. You can do the same for 384k to convert it to 192k and the high DSD rates recorded. So all the audible information has been captured at 96k.

Now you can locate the noise floor and figure out how many bits you need to reproduce the triangle's music. Add a couple of bits to the safe side, then reduce it to those bits using dithering techniques. Look dithering up if it is not something you understand. From what I have read, it can be anything from about 15 to 18 bits. This means the stuff that has been reflected has been chopped off. It then, for compatibility reasons, adds and subtract samples next to each other. Compress the removed information in the bits below what is chopped off,f, and if played back at 48k 24 bits, it will simply sound like noise. But to get the 96k back, use those bottom bits to recover the difference information, so you get the original back by adding and subtracting. In practice, it uses what is called a quadrature filter - but the principle is the same. We know the slow roll-off filter used and can figure out the best filter to upsample it back to some high bitrate. The filter and chucking away process is done using the triangle sampling they discussed in the link above. The inverse of that is simple linear interpolation. It's just an approximation of what was chucked away, but since it is all noise, who cares. The high sampling rate has all the bits except the top bit thrown and dithering applied so that in the audio band, you get the original bit depth. It makes reconstructing the audible signal easy. You pass it through a simple analog filter. The PS Audio guys give use an 80k 24db filter including an output transformer in the network. A friend of mine takes a DAC output and passes it through just a transformer relying on the natural roll-off of the transformer. Either way, the idea is to introduce the least audible processing to get that very high DSD delivered to the listener. In practice, more complex functions are used in downsampling to 96k and upsampling than simple triangles and linear approximation called b-splines. When encoding MQA the encoder decides on the best filter and transmits that with the 48/24 file. I think all this merging into the bottom bits is silly - transmit the 96k using Flac.

OK, that is what MQA is supposed to do with modern recordings. You usually use a sharp cutoff sinc filter to reduce the one bit to say 192k and transmit it. The MQA claim is in doing that it introduces time smear. The shallow filter introduces negligible time smear. But the MQA people forget one thing - Shannons SamplingTheoremm. If you take a bandlimited signal and upsample it using a sinc filter, then the bandlimited signal is reproduced precisely. That is what the DSD and Chord DAC's do. There is no need to try and recover an approximation of just noise. Only a blind listening test can determine which is best. But MQA certainly is a tricky way of doing it.

Note, however, when audio is made into MQA, what is supplied is usually the 48k, 96k, 192k or even DXD master rather than the original very high bitrate recording. That will likely have used a sinc filter that will introduce what MQA call time smear. When converted to MQA, that time smear will remain, so their claimed advantage of reducing time smear to really low levels does not hold. It would have to upsample it to some high frequency and rely on om Shannons Sampling Theorem then apply the MQA process. That is where the controversy starts. People like Rob Watts claim, unless you use his 1 million tap filter to get an exact reconstruction to 16 bits, then audible inaccuracies are introduced. Most engineers would claim with modern filters, such inaccuracies are inaudible. Only a blind listening test can tell. But measurements show Rob certainly has produced a perfect sync filter to at least 16 bits accuracy. However, he has never published the math showing you need a million taps that require a lot of processing power.

Just a personal opinion, I think MQA has gone in the wrong direction in trying to be compatible with existing standards. One can apply the slow roll-off filter and use a new compression method invented by Microsoft:
https://www.microsoft.com/en-us/research/wp-content/uploads/2016/02/Malvar_DCC07.pdf

You chop off everything below 18 bits in the frequency domain and transmit at the lowest rate where all frequencies above it are zero after chopping off to 18 bits. Any loss of resolution in the frequency domain is much harder to detect than in the time domain. It will likely produce better quality and smaller files and capture the information in the small number of recordings that do not fit into the triangle.
 

Blumlein 88

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I do not believe MQA is dumb. Paul McGowan, a noted critic of MQA, compared it and the original 192k masters. They were very close. He was impressed by how close the two were. As he said, 'After all, this is a pretty amazing process that allows streaming services to send high-resolution audio without degradation – no small feat.' That should be with nearly inaudible degradation from his listening tests.

The problem with MQA is that they claim it is better than the original (I will not go into why that is claimed, but they make the claim). As Paul said, 'If all the hype and hoopla had merely stated the results were indistinguishable from the original, I might be jumping up and down with how close they got it. Few systems have the resolving power of Music Room One, and the fact they got close after folding the music into a smaller file size is quite an achievement.'

The real issue with MQA is not that it is a lousy format - it is, as Paul said, quite an achievement. It is the BS marketing hoopla around it making claims better than the original etc. But the MQA people are hardly alone in that tactic.

Technically it is an interesting way to transmit modern recordings. Mostly the original performance these days is captured using some one-bit high sampling rate such as 10XDSD. However, MQA claims the audible information is contained in a triangle found in figure 7 of the following:

To transmit it efficiently, you first run it through a slow roll-off 32-bit filter flat to 20k. It slowly rolls off to be about 8db down at 48k and keeps on reducing even higher frequencies. Since it is all noise above 48khz, that is not a worry, nor likely is it 8db down at 48khz. It is just conjecture; probably (at least hopefully) they have done blind listening tests to prove it. Suppose you have a 192k record, ing and chuck away every second sample, then you get a 96k stream. But this chucking away has a consequence - information above 48khz is reflected into the 0-48khz region. However,r because of the shallow filter,r this information lies below the noise floor hence will not make any audible difference -you are just changing the noise a bit. You can do the same for 384k to convert it to 192k and the high DSD rates recorded. So all the audible information has been captured at 96k.

Now you can locate the noise floor and figure out how many bits you need to reproduce the triangle's music. Add a couple of bits to the safe side, then reduce it to those bits using dithering techniques. Look dithering up if it is not something you understand. From what I have read, it can be anything from about 15 to 18 bits. This means the stuff that has been reflected has been chopped off. It then, for compatibility reasons, adds and subtract samples next to each other. Compress the removed information in the bits below what is chopped off,f, and if played back at 48k 24 bits, it will simply sound like noise. But to get the 96k back, use those bottom bits to recover the difference information, so you get the original back by adding and subtracting. In practice, it uses what is called a quadrature filter - but the principle is the same. We know the slow roll-off filter used and can figure out the best filter to upsample it back to some high bitrate. The filter and chucking away process is done using the triangle sampling they discussed in the link above. The inverse of that is simple linear interpolation. It's just an approximation of what was chucked away, but since it is all noise, who cares. The high sampling rate has all the bits except the top bit thrown and dithering applied so that in the audio band, you get the original bit depth. It makes reconstructing the audible signal easy. You pass it through a simple analog filter. The PS Audio guys give use an 80k 24db filter including an output transformer in the network. A friend of mine takes a DAC output and passes it through just a transformer relying on the natural roll-off of the transformer. Either way, the idea is to introduce the least audible processing to get that very high DSD delivered to the listener. In practice, more complex functions are used in downsampling to 96k and upsampling than simple triangles and linear approximation called b-splines. When encoding MQA the encoder decides on the best filter and transmits that with the 48/24 file. I think all this merging into the bottom bits is silly - transmit the 96k using Flac.

OK, that is what MQA is supposed to do with modern recordings. You usually use a sharp cutoff sinc filter to reduce the one bit to say 192k and transmit it. The MQA claim is in doing that it introduces time smear. The shallow filter introduces negligible time smear. But the MQA people forget one thing - Shannons SamplingTheoremm. If you take a bandlimited signal and upsample it using a sinc filter, then the bandlimited signal is reproduced precisely. That is what the DSD and Chord DAC's do. There is no need to try and recover an approximation of just noise. Only a blind listening test can determine which is best. But MQA certainly is a tricky way of doing it.

Note, however, when audio is made into MQA, what is supplied is usually the 48k, 96k, 192k or even DXD master rather than the original very high bitrate recording. That will likely have used a sinc filter that will introduce what MQA call time smear. When converted to MQA, that time smear will remain, so their claimed advantage of reducing time smear to really low levels does not hold. It would have to upsample it to some high frequency and rely on om Shannons Sampling Theorem then apply the MQA process. That is where the controversy starts. People like Rob Watts claim, unless you use his 1 million tap filter to get an exact reconstruction to 16 bits, then audible inaccuracies are introduced. Most engineers would claim with modern filters, such inaccuracies are inaudible. Only a blind listening test can tell. But measurements show Rob certainly has produced a perfect sync filter to at least 16 bits accuracy. However, he has never published the math showing you need a million taps that require a lot of processing power.

Just a personal opinion, I think MQA has gone in the wrong direction in trying to be compatible with existing standards. One can apply the slow roll-off filter and use a new compression method invented by Microsoft:
https://www.microsoft.com/en-us/research/wp-content/uploads/2016/02/Malvar_DCC07.pdf

You chop off everything below 18 bits in the frequency domain and transmit at the lowest rate where all frequencies above it are zero after chopping off to 18 bits. Any loss of resolution in the frequency domain is much harder to detect than in the time domain. It will likely produce better quality and smaller files and capture the information in the small number of recordings that do not fit into the triangle.
Uh-huh!
 
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