Jump to content
IGNORED

MQA is Vaporware


Recommended Posts

9 minutes ago, Fitzcaraldo215 said:

Greater timing accuracy is what is conveyed by hi rez

 

If you study Monty's video's from xiph.org, you would know that band limited PCM has already infinite timing resolution.

Those hammering on time-smear should watch from 21:00 as many times until they get it.
 

 

Link to comment

 

10 minutes ago, mansr said:

MQA discards everything above 48 kHz and replaces it with rubbish. There are only two possibilities here:

  1. Frequencies above 48 kHz are audible.
  2. Frequencies above 48 kHz are not audible.

This leads to two possible conclusions:

  1. The mangling by MQA causes audible distortion.
  2. MQA is pointless.

There really are no other options.


A twist from their paper:
 

 
Quote

Even though some musical instruments produce sounds above 20 kHz [53] it does not necessarily follow that a transparent system needs to reproduce them; what matters is whether or not the means used to reduce the bandwidth can be detected by the human listener.

 

 
Link to comment

I don't think anyone is seriously debating in favor of ultrasonic frequencies being audible, except that widely discredited Japanese study from a long time ago.  However, there are many who do believe that hi rez recording offers sonic benefits for a number of reasons.  

 

I do, and I do not believe ultrasonic frequencies are audible.  But, I concede that it may be possible that hi rez conveys additional timing information about the sound.  So, I don't think MQA has itself somehow debunked the hi rez audio it supports.  Actually, it is the other way.  Filtering above audibility also conveys advantages, as does increased bit depth.

 

That old Meyer-Moran study you linked to has been widely debunked on peer review, and Meyer himself subsequently stated after intense criticism of their methodology that his paper had not been a true scientific study.  I have read it and it is riddled with procedural errors.  

 

 

 

 

Link to comment
16 minutes ago, soxr said:

 

If you study Monty's video's from xiph.org, you would know that band limited PCM has already infinite timing resolution.

Those hammering on time-smear should watch from 21:00 as many times until they get it.
 

 

Seen it.  But, if you believe that, and if you believe Stuart is lying in peer reviewed papers that have been out there for a number of years for possible refutation, then you should not ever buy anything with higher than RBCD resolution.

 

Personally, I have not bought a CD or RBCD file in over 10 years.  I think hi rez sounds much better.  Whether MQA also does remains to be determined, 

Link to comment
9 minutes ago, Fitzcaraldo215 said:

Seen it.  But, if you believe that, and if you believe Stuart is lying in peer reviewed papers that have been out there for a number of years for possible refutation, then you should not ever buy anything with higher than RBCD resolution. ...

 

Not quite. You don't have to buy anything with higher than RBCD resolution, but if your musical enjoyment is increased by more than the added cost (purchase, storage), then it's worth it to you. As for Stuart's papers, they appear to me to be lacking in actual measurements of the audibility of the described phenomena. 

"People hear what they see." - Doris Day

The forum would be a much better place if everyone were less convinced of how right they were.

Link to comment
1 hour ago, soxr said:

A good paper, and I had read it before.  However, I cannot locate the quote in it that you found so offensive.  

 

A worthwhile read, and a key underpinning of much of MQA.  You or others may squalk about it.  But, look at the date.  I am not aware of any AES members or other qualified persons who have refuted any of it since then.

Link to comment
1 hour ago, soxr said:

 

If you study Monty's video's from xiph.org, you would know that band limited PCM has already infinite timing resolution.

Those hammering on time-smear should watch from 21:00 as many times until they get it.
 

 

Timing resolution, in the sense of getting the timing at the right instant in time, was never  part of MQA's claimed solution.  Rather, their desire in the time domain is to reduce temporal blur or time smear.

 

If you look at the Stuart paper you cited, look at Fig. 14.  Or you can scroll down to Fig. 4 here:

 

The timings all have the same midpoint, but the spread as a result of traditional filtering is wider, more blurred, as a function of lower sampling rate.  Note that hi rez is considerably better than RBCD in Fig. 4 of the Q&A.

 

Unfortunately, Monty does not really touch on the esoterica of filtering to the extent MQA does. 

 

Don't believe it?  Then why has no one refuted it in their own testing?  But, by all means, see if you can reproduce Stuart's results or not.

Link to comment
6 minutes ago, Fitzcaraldo215 said:

Timing resolution, in the sense of getting the timing at the right instant in time, was never  part of MQA's claimed solution.  Rather, their desire in the time domain is to reduce temporal blur or time smear.

What's the difference?

Link to comment
14 hours ago, PeterSt said:

 

Peter. I just looked and I will rephrase what Fokus already told, but which possible goes beyond you and others :

The Nightfly is presented to you as an MQA album in native 48KHz. This means that it won't be 96KHz or more. It is and remains 48KHz.

 

The impulse response would be poor because of this, normally. Say that it is just CD quality and for CD quality to reconstruct, steep filtering is necessary in order to let drop the frequency to 0dB at 24KHz (half of 24KHz). This implies a high amount of ringing and that is your smear.

Might it be helpful, here's the spectrograph of the 2nd track :

 

spctr-UnicodeTrack0001.thumb.png.739f1bab4861e6992ec6ae8ee42e2652.png

As you can see, apart from some anomalies, no data above 24KHz. Not even noise, which at least proves that their workflow regarding this 48KHz is OK. It also proves that nothing is faked here (no fake hires).

 

One more thing for additional confusion :

I was promised that "no Hires" MQA albums were going to be there just the same. The benefit ? that ADC thing. This is an example of at least the existence of MQA without being Hires (I found some more).

The confusion is of course and again about the "where" of the ADC "counteraction".

 

Peter

 

 

 

 

Two questions from a non-technical follower of this MQA thread:

1. Does ringing show up at the frequency extreme, 24 Khz in your example,or elsewhere in the signal?

2. I have the Nighfly Trilogy box set which included a DVD of the Nightfly, ripped to 48KHz, might this be the same used to produce the MQA file?

 

Jim

Jim

Link to comment
17 minutes ago, mansr said:

What's the difference?

The signal (transient) may occur at the correct moment in time, but be altered (usually lengthened) in duration. 

 

It's all theoretical, anyway. As I understand it, the only signals that will be modified ("blurred") by conventional filtering are invalid ones (containing components greater than the Nyquist frequency). They shouldn't be present in the digital domain in the first place.

"People hear what they see." - Doris Day

The forum would be a much better place if everyone were less convinced of how right they were.

Link to comment
4 hours ago, Don Hills said:

 As I understand it, the only signals that will be modified ("blurred") by conventional filtering are invalid ones (containing components greater than the Nyquist frequency). They shouldn't be present in the digital domain in the first place.

 

Correct, but only for the DAC side: these signals that trigger reconstruction filter ringing should not be present. But in reality they are, because the dominant type of AA filter used during music production is half-band, meaning that it is only a small multiple of 6dB down at Nyquist.

 

Moreover, the DAC side is not really the problem here. Look at the ADC side, where signals exceeding Nyquist are present prior to AA filtering. And thus these signals will make the AA filter ring.

 

For clarity: I believe, strongly (and based on experiments), that this ringing is not audible, provided it happens at a frequency not audible to the listener.

Link to comment
5 hours ago, Fitzcaraldo215 said:

Timing resolution, in the sense of getting the timing at the right instant in time, was never  part of MQA's claimed solution.  Rather, their desire in the time domain is to reduce temporal blur or time smear.

 

 

 

True. They keep hammering on this ill-defined time smear, showing graphs of filter responses to illegal (at the DAC side) and unrealistic (at the ADC side) stimuli.

 

They claim that this is highly significant. They offer no proof. The studies they cite also offer no proof (go ahead, go through them in detail).

 

What proof would be needed?

 

Easy: a formal listening test (for once not botched by procedural errors and technical mistakes, please) whether a sampling rate above 1x rate is audible.

 

That's all.

 

People have been trying this for ages. And yet, today there is no definite result.

 

Highly significant?

 

Link to comment
1 hour ago, Fokus said:

...

Moreover, the DAC side is not really the problem here. Look at the ADC side, where signals exceeding Nyquist are present prior to AA filtering. And thus these signals will make the AA filter ring.

 

For clarity: I believe, strongly (and based on experiments), that this ringing is not audible, provided it happens at a frequency not audible to the listener.

 

I was under the impression that aliasing has long been a solved problem in ADCs used for music production, ever since NOS ADCs were replaced by delta-sigma.

 

Does it bother anyone that the term "ringing" does not accurately describe the problem anyway? How can a FIR Filter "ring"? It has no feedback loop. "Ringing" is a damped oscillation. That would imply that, for example, a signal at Nyquist (the classic +1, -1, +1, -1 digital signal) would result in more output than a signal at a lower frequency. That doesn't happen. The response of the filter to an impulse may look like a damped oscillation, but it isn't. It's simply what happens to a signal when you band-limit (remove the higher frequency components of) it. (Gibbs effect.)

"People hear what they see." - Doris Day

The forum would be a much better place if everyone were less convinced of how right they were.

Link to comment
8 hours ago, Fitzcaraldo215 said:

Filtering above audibility also conveys advantages, as does increased bit depth


Sampling at higher rates than 44.1K has some advantages: the AA filter can be outside the treshold of human hearing range, it can be less steep, ... and any DSP artefacts can be shifted to ultrasonic range where you can't hear them.

So basically highres is not about more music content (bit depth mainly determines noise floor), but about avoiding audible artefacts in the baseband as you have more places to hide mistakes into parts of the data that can't be heard.



 

Link to comment

'ringing' is the widely accepted term in the fields of signal theory and signal processing. It appears in the time domain whenever the steepness of a feature in a frequency domain function exceeds a certain limit, and vice versa. Gibbs is just a subset of this phenomenon, conventionally restricted to a Fourier series, not a transform.

 

You don't need feedback. Passive analogue filters of >2 order can ring.

 

Link to comment
5 minutes ago, Fokus said:

'ringing' is the widely accepted term in the fields of signal theory and signal processing. ..

 

Ok. At least it's descriptive. I can't think of a better name for it. 

"People hear what they see." - Doris Day

The forum would be a much better place if everyone were less convinced of how right they were.

Link to comment
2 hours ago, soxr said:


Sampling at higher rates than 44.1K has some advantages: the AA filter can be outside the treshold of human hearing range, it can be less steep, ... and any DSP artefacts can be shifted to ultrasonic range where you can't hear them.

So basically highres is not about more music content (bit depth mainly determines noise floor), but about avoiding audible artefacts in the baseband as you have more places to hide mistakes into parts of the data that can't be heard.



 

Hi,

Just to check on this, are we saying that current recording/mastering is at such a high sample rate that the filters used do NOT have temporal smear ?

Can it be confirmed that the temporal smear is in fact "group delay" of the filter - and the filters used for recording/mastering have an all pass filter response for those frequencies that the ear is interested in ?

As such, MQA can really only "tart up" previous recordings and at the same time implement aliasing which is detrimental ???.

Regards,

Shadders.

Link to comment
27 minutes ago, Shadders said:

Can it be confirmed that the temporal smear is in fact "group delay" of the filter -

 

No. Typical AA and AI filters are linear phase, thus with a constant group delay.

 

'time smear' is the 'problem' that is suggested when you look at the system response to an  unit impulse.

Link to comment
4 hours ago, Fokus said:

 

Correct, but only for the DAC side: these signals that trigger reconstruction filter ringing should not be present. But in reality they are, because the dominant type of AA filter used during music production is half-band, meaning that it is only a small multiple of 6dB down at Nyquist.

 

Moreover, the DAC side is not really the problem here. Look at the ADC side, where signals exceeding Nyquist are present prior to AA filtering. And thus these signals will make the AA filter ring.

 

For clarity: I believe, strongly (and based on experiments), that this ringing is not audible, provided it happens at a frequency not audible to the listener.

 

Would ultrasonic ringing be subject to the same potential problem as ultrasonic aliasing, intermodulation distortion?

 

Other than intermodulation distortion, is there anything you know of that would potentially cause time domain distortions in audible range transients?

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.

Link to comment

Next questions, for anyone who knows:

 

1. Does the output of the MQA filtering in the DAC (a) go through the same upsampling to 8x rates and SDM as any other signal, or (b) bypass the normal upsampling and SDM?

 

2. If the answer to 1(a) is yes, does someone (Archimago or anyone else) have measurements showing any difference in analog output of a DAC vs. non-MQA?

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.

Link to comment
52 minutes ago, Fokus said:

 

No. Typical AA and AI filters are linear phase, thus with a constant group delay.

 

'time smear' is the 'problem' that is suggested when you look at the system response to an  unit impulse.

Hi,

I was of the understanding that an all pass filter had a constant time delay across the frequency band of interest, hence no smearing/delay/phase change in the frequency band of interest.

I would have thought that linear phase, which introduces a change in phase (albeit linearly) with regards to frequency, means, as an example, that at 100Hz there is no phase change (no delay), and at 10kHz there is a 45deg phase change, hence a delay when compared to 100Hz.

Is it that linear phase filters provide minimal temporal smear, but it still exists ?

Regards,

Shadders.

Link to comment
8 minutes ago, Shadders said:

Hi,

I was of the understanding that an all pass filter had a constant time delay across the frequency band of interest, hence no smearing/delay/phase change in the frequency band of interest.

I would have thought that linear phase, which introduces a change in phase (albeit linearly) with regards to frequency, means, as an example, that at 100Hz there is no phase change (no delay), and at 10kHz there is a 45deg phase change, hence a delay when compared to 100Hz.

Is it that linear phase filters provide minimal temporal smear, but it still exists ?

Regards,

Shadders.

 

In linear phase filters, time through the filter does not vary by frequency.  In minimum phase filters, group delay is minimized but varies by frequency.

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.

Link to comment
7 minutes ago, Shadders said:

I would have thought that linear phase, ..., that at 100Hz there is no phase change (no delay), and at 10kHz there is a 45deg phase change, hence a delay when compared to 100Hz.

 

 

Linear phase = constant (time) delay over the entire frequency band.

 

If you have 45 degrees at 10kHz, then you'll find 0.45 degrees at 100Hz. No phase distortion. No temporal distortion.

 

It are the minimum phase filters so beloved by Meridian/MQA that cause phase distortion. Not that this matters much.

 

 

Link to comment

Create an account or sign in to comment

You need to be a member in order to leave a comment

Create an account

Sign up for a new account in our community. It's easy!

Register a new account

Sign in

Already have an account? Sign in here.

Sign In Now



×
×
  • Create New...