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MQA Deep Dive - I published music on tidal to test MQA

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danadam

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Umm, that is tag information, not watermark.
What do you mean? mqascan reads data embeded in pcm samples. @Cebolla could have converted those flacs to wavs, run the wavs through mqascan and the output would be the same.
 

sandymc

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The mansr tool returns MQA information embedded in the PCM stream - what other watermark are you after?

Maybe we're not talking about the same thing thing. I'm talking about the stuff that's encrypted into the "hidden" channel.
 

mansr

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In my view, one of the main ideas of MQA (even a paradigmatic change in that), and in open contradiction with the test methods used here, is that the space of the file doesn't need to be agnostic of the content it will register, because you are wasting much of that space, and that agnostic behavior forces (via unwanted filters) to smear the signal of the music to be recorded.
If instead what is to be registered is MUSIC, that information occupies only a fraction of the whole possible bitspace of, say, 96K x 24b file. That space is the famous orange triangle shown in MQA graphics. This limited space of real music is not mere speculation: it has been backup by the statistic of thousands of recordings, as well as the very physics of music (vibrational behavior of chords, surfaces, whatever), and countless research data by third parties.
All that wasted space is precisely what lossless compression algorithms like FLAC take advantage of. Their advantage over MQA is that they don't break down catastrophically if the input doesn't fit the expected model.
 
U

UKPI

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those vert soft filters allow to maintain time coherence in all frequencies (any signal has three components: amplitude, frequency and time. PCM doesn't deal the best way with the last one). Then, phases of harmonics are not smeared with what would, otherwise, be brickwall-like filters.
With time coherence guaranteed
MQA assumes that PCM will contain the timing errors
the theory of MQA says there is another, much more relevant reason for higher samplings: to gain space for softer slopes of the low-pass filters needed, for the "deblurring" reasons explained above. In that way they achieve the time coherence aimed.

You are continuing to ignore all the rebuttals for that "time smearing" and "filter ringing" claim made in this thread, such as:

Here we go again... Again with the bezier filtering... So again: those filters are not phase-linear and do not avoid aliasing. If you think they do, please bring some proof!

At least they are almost free from aliasing and do not distort phase as those are (for the most part) phase linear. You cannot have soft slope filters without aliasing!

It's almost like us standing in front of a red car, and you keep telling us it's blue! One of us is colourblind for sure ;)
TLDR - The test is valid, but the correct result may be unexpected without some understanding of the mathematical model behind PCM. A PCM "impulse" represents a normalized sinc function, not an impulse. Similar math applies to the edges of a square wave.

Impulses and square waves are mathematical constructs that exist in neither the analog domain nor in PCM encoding. They can be approximated by both up to the limits of the hardware (analog) or sample rate (PCM).

The Wikipedia Nyquist sampling theorem introduction gives a great explanation of the true meaining of PCM samples. The sampling theorem is the mathematical basis for PCM, basically the mathematical model showing that sampling works, and the constraints under which it works. In this model, each sample represents a normalized sinc function, and the sum of the normalized sinc functions represented by all the samples gives the reconstruction. So when we see "pre-ringing" or "time smearing" in an impulse (or square wave) response of a reconstruction filter, it is not a deficiency or artifact of the filter; it is the correct reconstruction according to the mathematical model that defines PCM encoding.
You explained it well.

By the way, I created a 16 bit 44.1kHz stereo PCM signal whose both channels contain a representation of an impulse signal after going through a linear phase brickwall reconstruction filter. One channel has a delay of approximately 1.031 microseconds compared to the other. One will easily find this difference by upsampling the file to 44100*22=970200Hz sampling rate and comparing the position of the resulting samples. This difference can go much smaller, (to picoseconds range) but I just stopped at this difference because Audacity (a free audio editing program) doesn't support resampling a track to over 1MHz sampling rate. That would make it difficult for people with less technical knowledge to check this out by themselves.

Obviously, this difference is smaller than the value of necessary time resolution mentioned in the concluding remarks of that convention paper for MQA (10 microseconds).



So, please:
Try to stop regurgitating and start arguing.
 

AdamG

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As previously announced but delayed. This thread is closed to further comment. It will remain available as a reference. New MQA threads can be opened. However, they will need to more focused on specific topics/aspects of MQA functionality.
 
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GoldenOne

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Ok, let's merge these, reopen it, and see what happens...
Thank you.
I think that closing all MQA threads is not a good way to go. Discussion may need to be moderated as with anything but blocking the discussion entirely is not helpful to anyone and does not look particularly good on ASRs part either given MQA's previous attempts to censor other forums.

I (and others I'm sure) appreciate the re-opening of this thread
 

BDWoody

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I think that closing all MQA threads is not a good way to go

Of course you do...

We'll see how it goes.

Keep it together people...
 

bboris77

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I applaud the decision to reopen this thread as I was in the middle of typing my response when it got closed :p

Here it is:

@GoldenOne, amazing work again. Methodically and patiently deconstructing MQA's response and demonstrating various flaws and outright falsehoods in their counterarguments. I also appreciate the fact that you acknowledge and concede certain points showing your maturity. You even go on to offer an olive branch to offer to further work with them to demonstrate potential benefits and value of MQA as an intelligent lossy format.

The nutty thing in this whole MQA debate is that there really is no need for another lossy format in the current day and age where FLACs are easily streamable. Lossless digital PCM files from either original analogue masters or PCM-based digital masters are as good as it gets for all music recorded up to today. If someone wants to come up with a new technological solution to improve the overall fidelity of audio reproduction, they will have to come up with a new superior way of recording/digitizing music as well.
 

tmtomh

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