MQA disappointing

It seems like a good and lively debate though!I didn’t know beating a dead horse was either good or lively.

I didn’t know beating a dead horse was either good or lively.

Deleted for good order

So I’ve tried to read this whole thread. I’m not an electrical engineer, nor a sound engineer. I can’t make heads or tails out of the more abstruse elements of the disagreement (well, that’s not quite true, I can sort of follow along, but definitely feel like a layperson).

What I am is a guy who subscribed to the Best Buy Tidal $120/year sub, with first year at $90 and already organized his collection. I had some hopes of trying out an MQA DAC at some point just to see what my ears think vs the Schiit I have today. But I also don’t know what my ears are missing on some releases that are MQA-only or MQA-defaulting, when Roon does all the work of converting to my 24/192 capable DAC. I could go to the trouble of getting Quboz instead, and on some days I’m tempted.

Request: can someone summarize in a somewhat dispassionate fashion what the sonic downsides (potential/theoretical and/or known/measured) are or might be of listening to Tidal MQA content downsampled to Redbook or 24/192 non-MQA DACs like the Modius (leaving aside whether or not you think the modius is resolving enough / good enough - insert whichever DAC you want here with the same basic capabilities)? Like I get the basics that there are some bits missing vs a truly lossless encoding. But once you get to analog, what might the difference be?

And if this has already been done, if you can point me to it I’d be very appreciative.

Thanks!

Audiophile are staunch anti MQA campaigners, so hardly an unbiased piece there. They actually ban people who make positive comments about MQA or hold different views. How do I know? Obvs…

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I really suggest you try yourself. You’ll probably get a polarised view here. MQA sounds great, through my system. Listened to the Labeque sisters play Philip Glass in MQA through Tidal yesterday (I’m on a free trial), and it was fabulous. On my system (full MQA render but I rarely use it).

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I’d say that the issue has been with the intersection of marketing and process, for example ■■ and his acolytes trying to redefine “losslessness”, which seems to still be the discussion the heroically patient @Jacques_Distler was having here.

That I can tell, MQA is an anachronistic and needlessly restrictive answer to a non-question, whose only competitors are long-established and well documented open standards. Many of the arguments I’ve heard otherwise have come from anecdotal reports by a crowd who seem having trouble articulating much beyond a noun, a verb, and a Meridian model number. The one serious study of the question - the Canadian one presented at AES, an organisation of which ■■ is a fellow - came back, inconclusive, a couple of years after the public launch of MQA. Call me a troll if you’d like, but this, to me, is a much better rational explanation of obfuscation than purported competition will ever be.

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ACK! A stupid algebra mistake below. Corrected now.

  • There’s no free lunch. If the objective is to minimize bandwidth, then (on information-theoretic grounds) you’ll do better applying compression to the whole signal, rather than treating the high frequency part of the bandsplit separately.
  • While I agree that the information content of the high frequency part of the bandsplit is lower, that’s not nearly enough. Here’s an experiment you can do to see that’s the case. All of these manipulations are trivially accomplished in ffmpeg or your favourite tool.
    1. Take a 24/96 source file. Let’s call it “2496.wav”.
    2. Downsample it to 24/48. Let’s call the result “2448.wav”
    3. Reduce the bit-depth of both files to 16 bits. Call the resulting files “1696.wav” and “1648.wav”.
    4. Now FLAC compress both of the reduced bit-depth files. Let’s call the resulting files “1696.flac” and “1648.flac”.
    5. size(1648.wav)-size(1648.flac) is how much you can save by compressing the low-frequency part of the bandsplit. size(1696.wav)-size(1696.flac) is how much you can save by compressing the whole thing.
    6. So the difference:
      size(1696.wav)-size(1696.flac)-size(1648.wav)+size(1648.flac)
      is an overestimate of how much you can save by just compressing the high-frequency part of the bandsplit. In other words, size(1696.flac)-size(1648.flac) is an underestimate of how much space you require to store the losslessly-compressed high-frequency part of the bandsplit.
    7. How much space do you have available? That’s
      size(2448.wav)-size(1648.wav)
      assuming (incorrectly) that we can devote all of the LSBs to this task.
    8. So now we just need to compare
      size(1696.flac)-size(1648.flac)
      with
      size(2448.wav)-size(1648.wav)
      to see whether the space required exceeds the space available.
    9. In other words, compute
      size(1696.flac)-size(1648.flac)-size(2448.wav)+size(1648.wav)
      If that number is negative, you win. If it’s positive, I win.

I await the results of your experiment.

You can’t both laud the “scientific” claims for MQA and deride independent efforts to test those claims as “trolling”. Sorry, but that’s not how science works.

In any case, there’s an experiment for you to do, to test one of these claims. If you are really committed to the scientific methods, you’ll post the results here.

Edit: Oh, and don’t cheat, by starting with a 24/96 file that was up-sampled from a 24/48 or, even worse, 16/44.1 source. Start with something with some actual ultrasonic content.

Edit 2: And, just so that you can’t complain that testing your claims scientfically is “too much work”, here’s a shellscript to automate the creation of the files listed above:

#!/bin/sh
ffmpeg -i "$1" -acodec pcm_s24le -ar 96000 2496.wav
ffmpeg -i "$1" -acodec pcm_s24le -ar 48000 2448.wav
ffmpeg -i "$1" -acodec pcm_s16le -ar 96000 1696.wav
ffmpeg -i "$1" -acodec pcm_s16le -ar 48000 1648.wav
ffmpeg -i 2496.wav -acodec flac -bits_per_raw_sample 24 -ar 96000 2496.flac
ffmpeg -i 1696.wav -acodec flac -bits_per_raw_sample 16 -ar 96000 1696.flac
ffmpeg -i 2448.wav -acodec flac -bits_per_raw_sample 24 -ar 48000 2448.flac
ffmpeg -i 1648.wav -acodec flac -bits_per_raw_sample 16 -ar 48000 1648.flac
ls -l

Just call it on any 24/96 file in any format (FLAC, ALAC, WAV, …).

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That’s me!! We love our music though and most importantly we have ears. Theory vs actual sound I guess. Most Meridian users were short changed by MQA btw. Also short change when Sooloos 2 became Roon.

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Jacques,
Looking over your posts top to bottom, including the new flac example, this appears to be the crux:

It is just not possible to fit losslessly-compressed 96kHz data into the 8 LSBs of a 48kHz file.

You’re assuming that data at a 96 kHz rate is being compressed. That’s not how lossless bandsplitters work. A 96 kHz file becomes two 48 kHz files after bandsplitting. The HF part is at a 48 k data rate and has much lower amplitude than the LF section. It may not be more than 8b above the noise floor in many areas. So even a factor of 2 compression may be adequate, but the predictive coding plus difference signal tack might be better than 2 given the shape of this signal.

Lossless bandsplitting is the basis for perfect reconstruction filterbanks, for example using quadrature mirror filters.

Although your flac comparison might be instructive, the objective in MQA wouldn’t be just to do optimal compression. The band split provides the 48 kHz LF data needed for redbook, but then can be recombined losslessly with the HF part to restore a 96kHz signal.

It’s described in the MQA patent if you want the link.

I am fully aware of that. Hence the somewhat more elaborate computation that I suggested.

FLAC uses both predictive coding and compression of the residual. If you have a scheme for encoding the high-frequency part of the bandsplit in fewer bits than FLAC can achieve, then I (and the rest of the digital audio world) would like know what that scheme is.

Just because they file a patent claiming to have invented a perpetuum mobile doesn’t make it true.

There is no free lunch. Some of the best minds in the world have worked on lossless compression algorithms. For your claim to be true, Bob Stuart’s compression algorithm has to be better than the very best available.

It’s more than instructive. It provides an upper bound on what lossless compression of just the high-frequency part of the bandsplit can achieve. The actual performance will be worse.

I suggest you either do the experiment or explain why the reasoning above is incorrect.

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Was this linked already? It’s from 2016 about the it initial announcement of Warner Bros MQA

Here’s an interesting piece from it.

Sometimes even a CD rip might indeed be used as a master, but that only happens when it’s the best recording available. Let’s say a label has gone out of business and the master recordings have been lost, but the artist still wants her music converted to MQA. More relevant to the Warner project is when early digital master tapes are unusable, or when the proprietary machine they were recorded on is beyond repair.

CD-quality masters? That’s hardly high-resolution.
Sure, but it’s about the music, right? Stuart indicates that MQA is not about high resolution in the usual sense; it’s about authenticity. “As far as we’re concerned, anything from a cylinder forward is legitimate as long as it’s the definitive statement about a recording,” Stuart told me. “If a recording is important enough, and all there is is a 78, that’s where we start. . . We’re really concerned about producing the definitive thing,” not the thing with the highest bit depth or sampling rate.

The cognitive dissonance is strong. Look at what he said while peddling yet another proprietary scheme. If the standard was published in that machine, we’d have no problem. Now we are staring at history repeating itself. No more proprietary crap.

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So here’s where I’ll stick my neck out. IF (and this is a big if, because I don’t think it’s actually true, but I want to make a point), IF there was a commitment to actually gather all the best back-catalog recordings, wax cylinders, etc - to create a global catalog of all the old recordings, and to be ‘fully archival’ and clear about which were the canonical recordings, I would rate that very highly as a positive. So much so, that I would be willing to pay for it, and to have others pay for it. I don’t like proprietary formats one iota. But… this article suggests that there’s the core of an idea there to do something that I think is a very important one - and one which (i think anyways) requires the idea of provenance / authentication / traceability. I’m not sure that gets the light of day much, and I don’t think I’ve seen evidence that the MQA folks are actually doing that work, or funding it, or even encouraging it. It’s possible you could create some kind of distributed ledger of non-proprietary formats of archival recordings too, and maybe that’s the really important project that some of us should do. But because there’s so much copyright issue involved, I don’t know how it could be done without the publishers involved. Welcome anyone’s thoughts. But I think that in a world where copyrights exist and are meaningful (and that’s a world I’m happy to live in, albeit they can be taken too far, eg sampling lawsuits) it’s hard to conceive of a way that has verification / provenance / traceability and involves copyright owners, that doesn’t involve their ability to get paid.

There’s a whole thread discussing provenance and MQA, which might be of interest to you…

This one is simple, It’s for @Chrislayeruk!!! :rofl: :joy: :innocent:

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That’s easy, MQA delivers high quality music that is free from Temporal blur doing no more damage than a few meters of air, in a downloadable streamable package making efficient use of the available bandwidth whilst also being backward compatible with existing playback equipment.
The resultant music file is authenticated to be authentic and can be signed off by the Artist/Owner as the official studio sound.

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Legend. :rofl:

Too bad they don’t make MQA generating soundboards for The Barn shows, huh?

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There’s an idea to ponder… We could do with an MQA plugin for the DAW… One day… One day :joy:

I’ve been into music and hifi for 40 years and I’d never heard of “temporal blur” until MQA came along. I still don’t know what it sounds like. Anyone care to enlighten me?