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MQA technical analysis


mansr

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Remember that high-frequency noise? The cause of it is the renderer dithering the output to no more than 20 bits (depending on parameters in the LSB metadata) with a 5-tap filter. This a frequency response plot of that filter when applied at 192 kHz:

 

mqa-ns-filter.png

 

Look familiar?

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OK, it is a firmware bug in the Meridian DAC firmware. Played exactly same non-MQA track twice, bit-perfect. Only difference being that between the two playbacks I played MQA version of the track. It seems to leave the noisy MQA rendering process active for the subsequent non-MQA track playbacks. So do not trust much on the listening impressions or such with this DAC when comparing MQA and non-MQA tracks, results depend on which order you play the tracks. LED indications are correct, but the real behavior is not.

 

For both cases, the DAC's leaky upsampling filter is the same. Remember the source is 96 kHz non-MQA track which by definition doesn't have any content above 48 kHz frequency.

 

Before playing any MQA tracks:

[ATTACH=CONFIG]32598[/ATTACH]

 

After playing an MQA track:

[ATTACH=CONFIG]32599[/ATTACH]

 

Exactly same track, exactly same bits going to the DAC in both cases.

 

Oh the fun of doing this research work, debugging Meridian's firmware as we go. :D

 

May be there is especial initialization of the DAC is need?

 

Such behaviour of DAC with any playback software (in bit perfect mode, of course)?

 

Possibly it is damaged DAC.

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I think it shouldn't be doing it... But maybe the decoder always outputs 2x rate, regardless of input. But the hardware decoder knows from the stream metadata that it originated from 44.1 and displays that in the DAC display while still in reality putting out 88.2.

 

So I have another data point for this. I got another version of tidal, some test version due to a bug I found in mine. This version doesn't seem to detect my Explorer 2 as an MQA dac like the release version. In this one, if I do not have "passthrough MQA" checked, then it will output 88.2 or 96khz to the explorer2 and the blue light will not light up.

 

I believe in the previous version, it detected the MQA dac and sent the MQA original data to it regardless of checkbox setting. Because people thought with the checkbox unchecked the dac was receiving 88.2 or 96khz data and still saw the blue light, they thought it preserved the MQA stream. But I think it was just the checkbox not taking effect with the explorer 2.

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I think it shouldn't be doing it... But maybe the decoder always outputs 2x rate, regardless of input. But the hardware decoder knows from the stream metadata that it originated from 44.1 and displays that in the DAC display while still in reality putting out 88.2.

The decoded stream metadata includes the original sample rate (prior to encoding). Clearly it must also be present in the metadata of the undecoded file.

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Hmmh, what 10m? For that number I get 25 cm.

Interesting. We are agreed aren't we that the overriding design goal of MQA is claimed to be to introduce no more temporal blur (whatever that means) than 10m of air. Taken literally as an aim to mimic 10m of air as a filter it woud imply no material filtering at all until the frequencies got pretty high.

It's a while since I looked at the MQA blurb/patents so I can't remember exactly how they cacluated the equivalence but I seem to remember that it involved having a very small number of filter taps. I also vaguely recall possibly a filter width of 10 us, although that seems impossible to me in pcm. I can't find the direct link but this article in sound on sound copies some of the MQA graphs and quotes the claims as being to introduce no more than 10us of time smear, possibly less. I don;t really understand how this is possible for any meaningful filter even at 192khz

 

MQA Time-domain Accuracy & Digital Audio Quality |

 

In any event, I think MQA would cheerfully admit to having narrow filters as a design goal.

 

How did you calculate 25 cm- that sounds more like the latency of 10m of air.

You are not a sound quality measurement device

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Interesting. We are agreed aren't we that the overriding design goal of MQA is claimed to be to introduce no more temporal blur (whatever that means) than 10m of air. Taken literally as an aim to mimic 10m of air as a filter it woud imply no material filtering at all until the frequencies got pretty high.

It's a while since I looked at the MQA blurb/patents so I can't remember exactly how they cacluated the equivalence but I seem to remember that it involved having a very small number of filter taps. I also vaguely recall possibly a filter width of 10 us, although that seems impossible to me in pcm. I can't find the direct link but this article in sound on sound copies some of the MQA graphs and quotes the claims as being to introduce no more than 10us of time smear, possibly less. I don;t really understand how this is possible for any meaningful filter even at 192khz

 

MQA Time-domain Accuracy & Digital Audio Quality |

 

In any event, I think MQA would cheerfully admit to having narrow filters as a design goal.

 

How did you calculate 25 cm- that sounds more like the latency of 10m of air.

 

The trouble with short filters is that they are not steep enough to avoid serious amounts of aliasing. What air has to do with anything completely escapes me. We're talking about a digital signal.

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What air has to do with anything completely escapes me.

 

I did not read the texts implied, but what I get from it is that air itself will cause said blur, and probably MQA is (or should be) designed such that MQA does not cause more blur than 10m of air implies. A sort of : we sit closer than 10m to our speakers anyway.

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I did not read the texts implied, but what I get from it is that air itself will cause said blur, and probably MQA is (or should be) designed such that MQA does not cause more blur than 10m of air implies. A sort of : we sit closer than 10m to our speakers anyway.

 

That's flawed thinking. The distortions caused by digital filters are completely different from those caused by air.

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I did not read the texts implied, but what I get from it is that air itself will cause said blur, and probably MQA is (or should be) designed such that MQA does not cause more blur than 10m of air implies. A sort of : we sit closer than 10m to our speakers anyway.

 

So this is just to trick your brain into believing you are closer to the source than, possibly, the microphone itself? If this is the case I do not like it. For instance: in a concert hall I sit often way more than 10mt from the orchestra...

 

 

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The trouble with short filters is that they are not steep enough to avoid serious amounts of aliasing. What air has to do with anything completely escapes me. We're talking about a digital signal.

I'm not intending to advocate MQA's position, but I have spent/squandered enough time to vaguely grasp what they are saying. One has to start from the premise that there exists something called time blur/smear which is crudely understood as correspondign to the impulse response of a filter (which as far as I am concerned has little meaning outside the mathematical equivalence of time and frequency domains.)

They argue that the ultimate aim of an end to end recording/reproduction chain should be to have no more time smear than 10m of air (ie the inevitable time smear experienced by a person 10 away from the source of sound. The problem of course is that involved in audio terms virtually no filtering at all, which is going to give one real problems in satisfying the requirements of the sampling theorem with a sample rate less than 500khz? 1 Mhz? probably more.

The underpinning of the system as i understand it involves accepting aliasing. It advocates one form of perfectionism at the expense of accuracy in the conventional sense.

 

I find it intriguing that as regards this time domain stuff, there exist at the moment different schools of thought involving essentially the opposite points of view, each of which have ardent adherents claiming huge audible benefits. The Watts school seems to regard increasing quality as involving hugely wide filters with 10s, hundreds maybe even millions of taps. At the opposite extreme, advocates of the nos (or rather sample and hold) school want zero tap filters. Both claim better time domain accuracy.

 

Meanwhile the question remains as to how

1. this gets you anywhere with a recording made with an old microphone

2. this gets through your speakers

3 it gets through the impulse response of the pinna.

4. this get through the cochlea into ones brain

You are not a sound quality measurement device

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T What air has to do with anything completely escapes me. We're talking about a digital signal.

I think it is intended to represent a sort of outer limit reality check on what might be necessary for realistic sound based on the idea of what might happen in the real world when hearing a real sound. of course you can say why 10 why not 5 etc.

 

I think it could have some use in drawing to the attention of analog fundamentalists that sound waves don't really exist in air in any appreciable quantity beyond the upper kilohertz range. I looked this up once since no one ever seems to refer to it but IIRC beyond a point (somewhere between 500khz- 1 mhz IIRC) or so sound waves won't make it through 1cm of air. So there must reach a sample rate at which in any reasonable sense all sound information is captured.

 

Of course there is a cogent argument that that limit is twice the rate of the highest frequency you can hear, but let's not go there..

You are not a sound quality measurement device

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Both claim better time domain accuracy.

 

Is that so ? I thought only the NOS guys do ?

Personally I don' see how long(er) filters can imply a better time accuracy.

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Is that so ? I thought only the NOS guys do ?

Personally I don' see how long(er) filters can imply a better time accuracy.

It's easy if you bear in mind that time domain accuracy has no particular meaning. If you have a frequency spectrum then that is a representation in the freuqncy domain. If you plot pressure or voltage against time then that is a time domain representation. As a matter of mathematics the two are interchangeable, so strictly any inaccuracy in one domain must mean some sort of inaccuracy in the other. But that doesn;t mean that there a unique type of inaccuracy in the time domain.

 

NoS avocates get worked up about the impulse response of a filter which represents the behaviour of the filter in the time domain. This does represent a sort of time blur when faced with a signal of infinite bandwidth; but there are no signals in the world with infinite bandwidth so why should that matter?

 

In any event a sample and hold dac is demonstrably inaccurate in its reproduction of every point in the time domain between the sampling instants. (so yes Mr Watts says that his long long filters are more accurate in the time domain.) At best the NoS claim to better time domain accuracy is largely confined to their reproduction of imaginary dirac pulses and perhaps perhaps some transients (even then only in some respects).

You are not a sound quality measurement device

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The trouble with short filters is that they are not steep enough to avoid serious amounts of aliasing.

 

The crux of MQA, when dealing with original rates in excess of 96k, is to allow aliasing, but within limits. Particularly in the baseband the filters are tuned so that the aliasing there does not exceed the programme's innate noise or another suitable masking threshold. In the 24-48kHz band the situation is more dire.

 

What air has to do with anything completely escapes me.

 

It is a marketing slogan.

 

"Our new digital system does as much damage as 10 m of air. Quick. Rush."

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I'm not intending to advocate MQA's position, but I have spent/squandered enough time to vaguely grasp what they are saying. One has to start from the premise that there exists something called time blur/smear which is crudely understood as correspondign to the impulse response of a filter (which as far as I am concerned has little meaning outside the mathematical equivalence of time and frequency domains.)

They argue that the ultimate aim of an end to end recording/reproduction chain should be to have no more time smear than 10m of air (ie the inevitable time smear experienced by a person 10 away from the source of sound. The problem of course is that involved in audio terms virtually no filtering at all, which is going to give one real problems in satisfying the requirements of the sampling theorem with a sample rate less than 500khz? 1 Mhz? probably more.

The underpinning of the system as i understand it involves accepting aliasing. It advocates one form of perfectionism at the expense of accuracy in the conventional sense.

 

A perfect reconstruction filter is an infinite sinc function. Any finite-length filter is an approximation.

 

Air, on the other hand, is a lossy medium for sound waves with both linear and non-linear distortions, although at frequencies, distances, and levels involved in music reproduction, both are negligible (it starts getting interesting at a few hundred kHz).

 

Equating the distance through air with a corresponding filter length is simply preposterous. If one wished to model the effects of air on a sound wave, a longer filter would be more accurate regardless of the distance modelled.

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A perfect reconstruction filter is an infinite sinc function. Any finite-length filter is an approximation.

I quite agree. But it is fair to say that a perfect filter at an audible corner frequency would be a real problem. This is well known by those who devise perceptual codecs. If the nyquist frequency of the system is treated as being audible in some way then there has to be a trade off between the band-limiting requirements of the sampling theorem and the practical problem of audibility of filter ringing.

 

So what is the cut off point. 20Khz, 40Khz, 80 khz? The former is rational and based on considerable evidence. Above there, it's conjectural at best.

You are not a sound quality measurement device

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As a matter of mathematics the two are interchangeable, so strictly any inaccuracy in one domain must mean some sort of inaccuracy in the other.

 

So "fools rush in," but:

 

Since the relationship of time domain and frequency domain accuracy is that of conjugate variables, wouldn't inaccuracy in one domain increase or decrease opposite to inaccuracy in the other? Thus Fokus' description of allowing frequency domain inaccuracy (aliasing) to some extent that is hopefully masked, in order to allow what is thought to be better time domain accuracy (I suppose intended for the sake of transients, percussion, instrumental and vocal attacks - all the inharmonic stuff).

One never knows, do one? - Fats Waller

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The crux of MQA, when dealing with original rates in excess of 96k, is to allow aliasing, but within limits. Particularly in the baseband the filters are tuned so that the aliasing there does not exceed the programme's innate noise or another suitable masking threshold. In the 24-48kHz band the situation is more dire.

 

So "the situation is more dire" where most of the recorded music is?

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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So "fools rush in," but:

 

Since the relationship of time domain and frequency domain accuracy is that of conjugate variables, wouldn't inaccuracy in one domain increase or decrease opposite to inaccuracy in the other? Thus Fokus' description of allowing frequency domain inaccuracy (aliasing) to some extent that is hopefully masked, in order to allow what is thought to be better time domain accuracy (I suppose intended for the sake of transients, percussion, instrumental and vocal attacks - all the inharmonic stuff).

All of this is kind of fine as long as we enter into a tradeoff discussion in the knowledge that

 

-there does not exist a single meaning to the expression time domain accuracy- it's largely marketing bah

-this whole transient accuracy thing should be demonstrated by reference to actual musical events not dirac pulses.

if you go deep into the MQA Q and A stuff soemwhere or other Staurt finally attempts to pin some meaning to time domain accuracy in terms of the ability to resolve two events. However, IIRC they did not look like two music events.

 

Real sound events do not start and finish within microseconds and if our senses were disturbed by timing uncertainty at that level I don;t think anyone would ever have liked LPs.

You are not a sound quality measurement device

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I quite agree. But it is fair to say that a perfect filter at an audible corner frequency would be a real problem. This is well known by those who devise perceptual codecs. If the nyquist frequency of the system is treated as being audible in some way then there has to be a trade off between the band-limiting requirements of the sampling theorem and the practical problem of audibility of filter ringing.

 

So what is the cut off point. 20Khz, 40Khz, 80 khz? The former is rational and based on considerable evidence. Above there, it's conjectural at best.

 

Indeed, the idea with digital audio is to place the Nyquist frequency above the audible range. The CD standard assumed an upper limit to audibility of 20 kHz and allowed ~2 kHz for the filter transition band. Now audiophiles argue that higher frequencies are in fact audible in some way or other and must thus be preserved. If one believes this to be true, one should not ever resample to a rate where the Nyquist frequency is within the range one deems important. It is mathematically impossible to do so with a finite filter without introducing ringing at the lower Nyquist frequency. Using an aliasing filter to avoid ringing will only push artefacts even further down into the audible range. If you think high frequencies are important, leave the sampling rate high. If you want to use lossy compression to reduce the file size, there are ways of doing that without resampling to a lower rate.

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So "fools rush in," but:

 

Since the relationship of time domain and frequency domain accuracy is that of conjugate variables, wouldn't inaccuracy in one domain increase or decrease opposite to inaccuracy in the other? Thus Fokus' description of allowing frequency domain inaccuracy (aliasing) to some extent that is hopefully masked, in order to allow what is thought to be better time domain accuracy (I suppose intended for the sake of transients, percussion, instrumental and vocal attacks - all the inharmonic stuff).

 

You're thinking of a different kind of inaccuracy. Given a signal, it's value is more well-defined over a narrower time interval while it's frequency content is better defined over a longer period. In the extreme, for an infinitesimal instant (a single sample in discrete time), the frequency spectrum is undefined while the value is exact. Over an infinite interval, the spectrum is exact while the value reduces to an average.

 

Now if we settle on some balance between these opposing goals, an error in one domain translates to an equivalent error in the other. I think this was what adamdea was getting at.

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You're thinking of a different kind of inaccuracy. Given a signal, it's value is more well-defined over a narrower time interval while it's frequency content is better defined over a longer period. In the extreme, for an infinitesimal instant (a single sample in discrete time), the frequency spectrum is undefined while the value is exact. Over an infinite interval, the spectrum is exact while the value reduces to an average.

 

Now if we settle on some balance between these opposing goals, an error in one domain translates to an equivalent error in the other. I think this was what adamdea was getting at.

I quite agree with you about the time/ frequency uncertainty problem. But that wasn't what I was getting at. the uncertainty tradeoff would have increasing precision in one domain leading to reduced precision in the other.

My point was simply that you cannot get something wrong in the frequency domain without also getting it wrong in the time domain (this is trite: if it is right in one it must be right in the other). If you allow aliasing (or unsurpressed spectral imaging) you must be creating some sort of error in the time domain. However you might in fact be happy to have that error if it allows you to be more accurate in another respect. The NoS dac does not (just) trade off time domain against frequency domain it trades off one sort of time domain error against another. AND SO MUST MQA

You are not a sound quality measurement device

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Please read again.

 

My music does not reside in the 24-48kHz band. Maybe Batman's does?

 

Oh, I was thinking of sampling rates.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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