So I suppose you have no preference for anything above CD quality..? any tests which prove 24/96, 24/192 or 24/384 sounds better than CD quality and also explaining why?
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MQA Sounds Really Good!
- Thread starter watchnerd
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So I suppose you have no preference for anything above CD quality..? any tests which prove 24/96, 24/192 or 24/384 sounds better than CD quality and also explaining why?
There are multiple scientific AES papers regarding MQA and it is worthwhile to study them, especially since they contain many references to other scientific studies. But the only way to prove to yourself if you prefer HD above CD quality is to listen to it. The same counts for MQA
Or the way it distorted the original HD recordings digital stream. MQA IS a lossy process.
Well yes, I mean they can do whatever they want with the master they get so I just pointed out to the master itself
PS: I know it's lossy and I pinted out in my post indeed and I don't see the point in it while we have many other formats and well lossless files that we can stream easily now (I live in a little city in Italy and we will have 1Gbit internet in a month). Plus I suck at ABX even Ogg q6 vs the origial so I can't say a word about fidelity ahah
Try these files :
HiRes Download - test bench
We invite you to join us in this evaluation of future consumer delivery formats. FLAC is a lossless encoding of WAV-files derived directly from our production original used for the SACD and Pure Audio Blu-ray. All resolutions and encodings are derived from the same original DXD source files. Please send us an e-mail and share your experience on your practical experience with these file formats. Enjoy the music!
http://www.2l.no/hires/
On Tidal meanwhile in MQA resolution:
- over 100 albums in 24/352.8
- over 2000 albums in 24/192
- over 6000 albums in 24/96
MQA purifies the Master Tapes and gets rid of the cumulated A/D - D/A brickwall filter effects. Only frequencies above 48 kHz are lossy folded
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Sounds like someone has been in the Coolaid. Hint: master tapes are all analog in general.
I have downloaded a few just to listen to them in past. I just admitted that I can't even ABX Ogg q6 with the original... What difference could I hear with most of these uncompressed files?
Anyway form a quality point of view I would get the 352.8Khz 24bit stereo ones wich are close(?) to the original DXD format but again I couldn't have(ear) any benefit(improvement) from it.
No I can't, at least not without investing in some alternate playback infrastructure and source (the basis of my denial), which I'm not anticipating at this time as being one of those little things I'm likely to do, being sufficiently amused with what I have, and absolutely not subscribing to Tidal, as it currently exists (described elsewhere).
So, for the sake of no further argument, lets go with "No I won't." as an equivalent alternative to "No I can't".
I've heard MQA and non-MQA at a "club" meeting (twice) without knowing I had and nothing tweaked my twang, and I probably heard plenty of MQA at the one and only Audio show I attended, without any ear perking moments I could attribute to the format, so, at this point can't/won't applies to me.
I'll just try to stay out of MQA discourses starting now.
Have fun.
This intrigues me very much
The improvements are very small and you will need to listen via a high-end system or - headphones. What intrigues me is the 9x smaller MQA file sounds just as good as the original DXD file
The improvements are very small and you will need to listen via a high-end system or - headphones. What intrigues me is the 9x smaller MQA file sounds just as good as the original DXD file
OP
watchnerd
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Are you asking if I can pass a DBT test between a high resolution FLAC (say 24bit/192khz) and the same via a level-matched, perfectly done downsample to 16bit/44.k1khz FLAC?
Nope.
I've tested, and I can't. Or haven't yet.
As for listening to MQA....well, I'm the thread OP.
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Interesting read about MQA:
"We wanted to understand one of the paradoxes of digital audio, which was why 96kHz sounds better than 48, and 192 sounds a little better than 96, and 384 sounds a little better than 192, and 768 can actually sound better than 384! Each time we’ve got a doubling of data rate, but we’ve got the rapidly diminishing return of incremental sound quality.
That’s a puzzle because we can’t really hear frequencies above 20kHz. From the pure frequency-domain point of view, hi-res sampling doesn’t seem to make a lot of sense, which is why you see some skeptics who don’t actually think it through to the next stage.
And yet, we observe that this is the case and understood by producers, and recording engineers. If you said to a recording engineer, “You can’t hear an improvement with 192k because theory says you can’t,” they’d think you were nuts. They hear it clearly; it’s part of their everyday experience.
In general, higher rates sound better because when you sample at a high rate, there’s less ringing in the file and the ringing is shorter. But the file is still full of ringing. Every sample is a mini-transient that ignites that system. The higher rates sound better because the ringing is shorter, and there’s more chance to ameliorate arithmetic noise in the filters and quantizers. (In fact we separated these two mechanisms in our recent listening tests described at AES last October.)
Acoustically, this is like a sheer curtain on top of the audio. If we did the same kind of processing in the visual domain, we’d think it was a joke. The idea that every edge has got parallel lines near it because of the way the filters ring would be unacceptable. The mess we’re in has been a mixture of lack of understanding combined with pragmatism.
Imagine where we started. It’s weird that faster and faster sampling can sound better. We puzzled about it. And then a light bulb went on. We concluded that the problem is a mismatch between how (specifically linear) PCM samples and reconstructs music, and the human neuro system."
"It’s hard to think of conventional PCM as anything but crude and inefficient. A lot of those bits are baggage.
Oh, total baggage. People like 24-bit LPCM in the studio because it gives more room for error. It allows smaller steps but can also code 60 to 80dB below silence and below the thermal noise of air! It’s ridiculous in terms of distribution channel capacity to transmit information far below silence; somewhere down there data’s being used that has no meaning.
Similarly, there’s a lot of unused coding area in the first octave above 20kHz, and even more around the next octave if it’s 192kHz, and even more than that if it’s 384 kHz; these areas have no signal, simply benign inaudible noise. The higher the sample rate the more inefficient it gets. It gets progressively, preposterously inefficient."
http://www.theabsolutesound.com/articles/meridians-master-quality-authenticated-the-interview/
"We wanted to understand one of the paradoxes of digital audio, which was why 96kHz sounds better than 48, and 192 sounds a little better than 96, and 384 sounds a little better than 192, and 768 can actually sound better than 384! Each time we’ve got a doubling of data rate, but we’ve got the rapidly diminishing return of incremental sound quality.
That’s a puzzle because we can’t really hear frequencies above 20kHz. From the pure frequency-domain point of view, hi-res sampling doesn’t seem to make a lot of sense, which is why you see some skeptics who don’t actually think it through to the next stage.
And yet, we observe that this is the case and understood by producers, and recording engineers. If you said to a recording engineer, “You can’t hear an improvement with 192k because theory says you can’t,” they’d think you were nuts. They hear it clearly; it’s part of their everyday experience.
In general, higher rates sound better because when you sample at a high rate, there’s less ringing in the file and the ringing is shorter. But the file is still full of ringing. Every sample is a mini-transient that ignites that system. The higher rates sound better because the ringing is shorter, and there’s more chance to ameliorate arithmetic noise in the filters and quantizers. (In fact we separated these two mechanisms in our recent listening tests described at AES last October.)
Acoustically, this is like a sheer curtain on top of the audio. If we did the same kind of processing in the visual domain, we’d think it was a joke. The idea that every edge has got parallel lines near it because of the way the filters ring would be unacceptable. The mess we’re in has been a mixture of lack of understanding combined with pragmatism.
Imagine where we started. It’s weird that faster and faster sampling can sound better. We puzzled about it. And then a light bulb went on. We concluded that the problem is a mismatch between how (specifically linear) PCM samples and reconstructs music, and the human neuro system."
"It’s hard to think of conventional PCM as anything but crude and inefficient. A lot of those bits are baggage.
Oh, total baggage. People like 24-bit LPCM in the studio because it gives more room for error. It allows smaller steps but can also code 60 to 80dB below silence and below the thermal noise of air! It’s ridiculous in terms of distribution channel capacity to transmit information far below silence; somewhere down there data’s being used that has no meaning.
Similarly, there’s a lot of unused coding area in the first octave above 20kHz, and even more around the next octave if it’s 192kHz, and even more than that if it’s 384 kHz; these areas have no signal, simply benign inaudible noise. The higher the sample rate the more inefficient it gets. It gets progressively, preposterously inefficient."
http://www.theabsolutesound.com/articles/meridians-master-quality-authenticated-the-interview/
I've listened to them with my K712 Pro, Objective2 and Khadas TB: the problem is me not the equipment
About the size and what you said before on the folding. I haven't studied how MQA works but I have to disagree with you when you say that it only goes lossy after 48kHz because if so the MQA file should be at least the same size (and actually bigger considering that it adds the part of the 48kHz+ data but in lossy mode) of the respective lossless file (FLAC, ALAC or any other) but it is not.
About the size and what you said before on the folding. I haven't studied how MQA works but I have to disagree with you when you say that it only goes lossy after 48kHz because if so the MQA file should be at least the same size (and actually bigger considering that it adds the part of the 48kHz+ data but in lossy mode) of the respective lossless file (FLAC, ALAC or any other) but it is not.
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Actually there is one AES paper about MQA from McGill university which concluded MQA wasn't heard. The particulars of that are interesting.
There are several AES papers about things somewhat related to why there is MQA, but like everything MQA you are a victim of bait and switch tactics.
Every time someone gushes about MQA, they are completely unable to back it up with anything other than biased sighted tests. Not an ABX or MUSHRA test in sight, no critical analysis of the codec (because it's secret and proprietary), no facts.
Just blind faith in marketing materials.
Just blind faith in marketing materials.
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Why does anyone pay attention to this convoluted nonsense? There's a lot of unused coding area so we throw it away and its "spirit" remains? An A/D with inadequate anti-aliasing filters has lost all the information beyond fs/2 total recovery of all information that was below fs/2 exactly violates first principles.
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Nope, no preference...
Love to see the tests that 'prove' I'm missing out...
Correct! There was no significant statistical difference found between the large 24/96 files and the more compact MQA files. The test needs tp be done with 24/192 and higher vs MQA as well. Some hear HD others don't but more tests need to be done.
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