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Just now, adamdea said:

Shadders please,

you are arguing yourself round in a circle.The property which MQA is targeting is not dispersion in the sense you are using it. QED.

The MQA argument is about some alleged time property of filters shown here in the slide attached by manisandher. Whatever it is, a linear phase filter has it. It is not dispersion in the sense you use it. That may mean that somewhere along the line the MQA literature has used the word in one sense and then again in another. No surprises there  2. Temporal Response End-to-End.jpg

Hi,

I am not going round in circles. I has asked for a explanation, received the explanation, and have challenged the explanation.

This is engineering. Dispersion is an engineering term.

The figure 14 clearly states it is : ""Impulse responses: upper and middle as dB magnitude and lower as amplitude"

So, it is NOT a non linear phase response, NOR is it dispersion.

 

You have stated "Whatever it is, a linear phase filter has it".  Has what exactly ?

 

It is claimed as per your diagram, and as per AES paper figure 14, that it is temporal blur (dispersion). It is not. It is just a magnitude plot of the impulse response. The AES paper states it is an impulse response. An impulse response is NOT dispersion.

 

Just because MQA Ltd states it is "blur" does not mean it IS "blur".

 

Regards,

Shadders.

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By the way, this 1997 paper by dcs also uses the heading "dispersion" to refer to the effects of ringing in a linear phase filter. (very similar to the  MQA graphic in my previous post). 

http://www.cirlinca.com/include/aes97ny.pdf

 

Obviously it is irrelevant whether this is a correct usage of the engineering term dispersion. It is the phenomenon from which we are supposed to need saving.

You are not a sound quality measurement device

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39 minutes ago, Fokus said:

Pioneered by Craven and for a while picked up in a few audio magazines.

 

These plots make it easy to depict anything that is not the main central lobe as distortion, at levels of -20dB, no less.

 

You have to scare the audience before presenting a solution.

Yeah, I've never seen such plots anywhere else. From a mathematical and engineering perspective, it is uninteresting.

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5 minutes ago, adamdea said:

By the way, this 1997 paper by dcs also uses the heading "dispersion" to refer to the effects of ringing in a linear phase filter. (very similar to the  MQA graphic in my previous post). 

http://www.cirlinca.com/include/aes97ny.pdf

 

Obviously it is irrelevant whether this is a correct usage of the engineering term dispersion. It is the phenomenon from which we are supposed to need saving.

Hi,

If you examine page 4, it states "Energy Dispersion at Different Sample Rates  - Figure 6 shows the energy associated with the transient response".

This plot is a filter response to an impulse - transient. Again - it does not show that the audio signals in the relevant bandwidth undergo dispersion.

As long as the audio signal is in the linear phase region, then there is NO dispersion. An impulse contains ALL frequencies up to infinity (there are practical limitations). The figures in the paper you have referenced do NOT show dispersion occurring at audio frequencies, just the difference in impulse response.

Regards,

Shadders.

P.S. There is a cable company that in their demonstration (youtube) stated that they have "discovered" a new form of distortion/noise. This was 2009 - still no proof, or rewriting of the text books.

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17 minutes ago, Shadders said:

Hi,

If you examine page 4, it states "Energy Dispersion at Different Sample Rates  - Figure 6 shows the energy associated with the transient response".

This plot is a filter response to an impulse - transient. Again - it does not show that the audio signals in the relevant bandwidth undergo dispersion.

As long as the audio signal is in the linear phase region, then there is NO dispersion. An impulse contains ALL frequencies up to infinity (there are practical limitations). The figures in the paper you have referenced do NOT show dispersion occurring at audio frequencies, just the difference in impulse response.

Regards,

Shadders.

P.S. There is a cable company that in their demonstration (youtube) stated that they have "discovered" a new form of distortion/noise. This was 2009 - still no proof, or rewriting of the text books.

As I understand it the energy which is or might be redistributed in time (I am not going down the road of using the word dispersion) is in the transition band of the filter. Of course, for a sensible anti alias or anti imaging filter  it is therefore energy at inaudible frequencies and that energy can only arise in the first place to the extent that it is in the signal. It is therefore grossly exaggerated by a notional dirac input 

You are not a sound quality measurement device

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I should point out that this is very much the subject of my post over on Pinkfish (itself referring to an article by Archimago) which has been made the subject of another thread on this forum. I have also challenged Jim Austin on the Stereophile forum for his use of impulse responses to (purport to) analyse  the "time domain" behaviour of MQA and non MQA filters. 

O tempora o mores.

You are not a sound quality measurement device

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2 minutes ago, adamdea said:

As I understand it the energy which is or might be redistributed in time (I am not going down the road of using the word dispersion) is in the transition band of the filter. Of course, for a sensible anti alias or anti imaging filter  it is therefore energy at inaudible frequencies and that energy can only arise in the first place to the extent that it is in the signal. It is therefore grossly exaggerated by a notional dirac input 

That is my understanding of what MQA is claiming as much as one can understand their tales mixing real with fantasy.  That long steep filters disperse energy in the transition band over time more than shallow short filters.  Then you make up the tale about air.  Air is absorbing frequencies, but if you graph the amplitude it can look like a filter dispersing energy.  

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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17 minutes ago, adamdea said:

As I understand it the energy which is or might be redistributed in time (I am not going down the road of using the word dispersion) is in the transition band of the filter. Of course, for a sensible anti alias or anti imaging filter  it is therefore energy at inaudible frequencies and that energy can only arise in the first place to the extent that it is in the signal. It is therefore grossly exaggerated by a notional dirac input 

Hi,

Yes - this was the statement behind my original post many pages back - the filter on the front end of the ADC has a cut off near the 1MHz frequency - and that ensures that any statement of "blur" is inconsequential - it does not happen with ADC's from the mid 1990's onwards that are Sigma-Delta.

In fact, a filter with a cut off at or near 96kHz will not cause "blur" since there are no audible signals above 50kHz. The filter will be linear phase at this region and above (to a point - filter spec dependent).

Regards,

Shadders.

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9 hours ago, esldude said:

So what does Lee have to say now that Bob has said up to 17 bits of resolution?

 

On the Hoffman forums he believes 17 bits is taken out of context because Stereophile said 23 bits in some past article.

Roon Rock->Auralic Aria G2->Schiit Yggdrasil A2->McIntosh C47->McIntosh MC301 Monos->Wilson Audio Sabrinas

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1 minute ago, Dr Tone said:

 

On the Hoffman forums he believes 17 bits is taken out of context because Stereophile said 23 bits in some past article.

Well the good thing is saying something so foolish will not cause his credibility to take a hit.  There isn't any left.  

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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14 minutes ago, esldude said:

That is my understanding of what MQA is claiming as much as one can understand their tales mixing real with fantasy.  That long steep filters disperse energy in the transition band over time more than shallow short filters.  Then you make up the tale about air.  Air is absorbing frequencies, but if you graph the amplitude it can look like a filter dispersing energy.  

As I understand it air is somewhat elastic and behaves like a filter (not linear phase or even minimum phase). Interestingly this has the effect that for all practical purposes there is no such thing as sound in air over a few hundred kHz. (can't remember the exact figure but somewhere before 1 MHz you get to the point where 1cm of air attenuates at 90dB or more). Anything you find above a certain level in a recording is an electronic artefact.

You can repeat something like that argument above an order of magnitude lower for most mics (though they might be minimum phase)

You are not a sound quality measurement device

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9 minutes ago, Shadders said:

Hi,

Yes - this was the statement behind my original post many pages back - the filter on the front end of the ADC has a cut off near the 1MHz frequency - and that ensures that any statement of "blur" is inconsequential - it does not happen with ADC's from the mid 1990's onwards that are Sigma-Delta.

In fact, a filter with a cut off at or near 96kHz will not cause "blur" since there are no audible signals above 50kHz. The filter will be linear phase at this region and above (to a point - filter spec dependent).

Regards,

Shadders.

Yes. The issue is the effect or otherwise of the anti alias filters and anti imaging filters required to get to and from the distribution sample rate. These are unproven and dubious at 16/44 , entirely implausible at x/96 and comically absurd at x/192.   But the issue is the effect on the filter of the small to zero  amount of program material in the transition band of a sharpish LF filter ie around 20-24Khz for 16/44 40-55 khz for x/96 and maybe 80-120 khz for x/192 

You are not a sound quality measurement device

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1 hour ago, Dr Tone said:

 

On the Hoffman forums he believes 17 bits is taken out of context because Stereophile said 23 bits in some past article.

 

That is not what I said. I said, "In other articles I have read like in Stereophile, Bob claims it is around 23 bits so I think I need to see if there is a way to reconcile and that we are not taking him out of context."

 

So I am saying we need to investigate why the discrepancy exists and make sure it is not taken out of context.  I am clearly open to the possibility that Stuart meant it on the 17 bits comment.

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47 minutes ago, adamdea said:

But the issue is the effect on the filter of the small to zero  amount of program material in the transition band of a sharpish LF filter ie around 20-24Khz for 16/44 40-55 khz for x/96 and maybe 80-120 khz for x/192 

Hi,

For the 44.1kHz/48kHz sample rate, then possibly the energy near 20kHz is an issue. For 96kHz, the energy at 40kHz+ is approx -80dB to -100dB down from full scale. Two issues - it is too low to hear, and no one can hear in this range. (i am examining Hifi News Hi-Res download plots).

If the ADC is a Sigma-Delta - then none of this is an issue. Only an issue for recordings at the lower sample rates.

Brian Lucey said he uses the Pacific Microsonics ADC at 44.1kHz, i have Depeche Mode's - Spirit album - everything is clear and precise - a good recording like most of Depeche Mode's.

Regards,

Shadders.

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31 minutes ago, Lee Scoggins said:

So I am saying we need to investigate why the discrepancy exists and make sure it is not taken out of context.  I am clearly open to the possibility that Stuart meant it on the 17 bits comment.

 

Stuart admitted it's 17 bits:
 

 

Designer of the 432 EVO music server and Linux specialist

Discoverer of the independent open source sox based mqa playback method with optional one cycle postringing.

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22 minutes ago, Shadders said:

Hi,

For the 44.1kHz/48kHz sample rate, then possibly the energy near 20kHz is an issue. For 96kHz, the energy at 40kHz+ is approx -80dB to -100dB down from full scale. Two issues - it is too low to hear, and no one can hear in this range. (i am examining Hifi News Hi-Res download plots).

If the ADC is a Sigma-Delta - then none of this is an issue. Only an issue for recordings at the lower sample rates.

Brian Lucey said he uses the Pacific Microsonics ADC at 44.1kHz, i have Depeche Mode's - Spirit album - everything is clear and precise - a good recording like most of Depeche Mode's.

Regards,

Shadders.

I think we are probably in agreement provided it is understood that all pcm recordings (and probably almost all dsd recordings) are at "a lower sample rate". The immediate  output of the adc's modulator is really neither here nor there (where are the multibit 14Mhz sample rate recordings?).  

The amusing thing about the HFNRR plots is that  PM actually notes in places that most piano recordings have no information high enough to warrant hi rez. Does that stop people buying them?

 

The amusing point is that I can discern no difference in the enthusiam expressed for  hi rez recordings with payload in the range that might  make a difference from recordings with nothing there (most old recordings); same with sighted filter preferences. What does that tell us?

You are not a sound quality measurement device

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Hi,

As per the AES paper 2014, Figure 3 shows the 8x cascaded 2nd order Butterworth filters. So I have used LTSpice to simulate and reproduce the impulse response. The results are as per the graph. The AES figure is NOT an impulse response but a pulse which is 6us duration.

image.thumb.png.056b31fc7a3de11e198ef0c528202305.png

 

The next graphs is an FFT of this pulse. As can be seen, the frequency range far exceeds the 20kHz bandwidth, and at levels which will never be reached by music.

image.thumb.png.2d6a165382a04db4335858c48d73a3a3.png

In fact, using an impulse/pulse response is thoroughly misleading, as it severely exaggerates dispersion that can be caused by low frequency filters such as 22.05kHz on music material, but will NOT occur by 44.1kHz/48kHz filters (since energy content above 30kHz is negligible and cannot be heard).

Regards,

Shadders.

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9 minutes ago, adamdea said:

 

The amusing point is that I can discern no difference in the enthusiam expressed for  hi rez recordings with payload in the range that might  make a difference from recordings with nothing there (most old recordings); same with sighted filter preferences. What does that tell us?

That any quality difference is in all the other aspects of good recording/mastering practice which doesn't require hirez.  And/or the idea of there being more in hirez makes people hear more regardless of whether there even is more there.  Like old recordings released in hirez that are sourced from 40-50 year old reel tape.  

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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1 minute ago, esldude said:

That any quality difference is in all the other aspects of good recording/mastering practice which doesn't require hirez.  And/or the idea of there being more in hirez makes people hear more regardless of whether there even is more there.  Like old recordings released in hirez that are sourced from 40-50 year old reel tape.  

mais oui. It has always tickled me that the terrible flaw with Meyer and Moran was that unbeknownst to the participants (who IIRC were asked to bring their favourite sacds) some of them were upsampled 16/44. 

You are not a sound quality measurement device

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20 minutes ago, esldude said:

I have run completely out of patience with Stuart answering questions.  He has let sites like this one post questions, and he has answered them.  But he has been evasive and not at all forthcoming on most of the common questions.  Done so repeatedly and it has definitely been intentional.  He clearly has no intent of being transparent about MQA or answering those questions.  He is the fellow who could easily have cleared up most of the same questions at the very beginning and he did not wish to do so. 

 

If he answered the questions honestly it would turn off more potential users, reducing the chance of it taking hold and him making allot of money from licensing.  It's no coincidence that most MQA QA events have been cancelled recently, he knows the public other than Lee are catching on and might ask questions he doesn't want to answer.

Roon Rock->Auralic Aria G2->Schiit Yggdrasil A2->McIntosh C47->McIntosh MC301 Monos->Wilson Audio Sabrinas

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