[SlimDevices: Audiophiles] Internet Blind Test: Linear vs. Minimum Digital Filters...

[Archimago]

Hello everyone, it's up!

I know some people have strong feelings about digital upsampling filters
and there's especially the notion of minimum phase filters being
superior or more natural sounding than the typical linear filters used
in most DACs.

Here's an opportunity to try it out for yourself and let me know if you
hear a difference!

Note that the files are high-resolution 24/176 so please make sure your
DAC is capable of this bit- and sample-rate.

Test will conclude on June 25th so please submit your results by that
time.

Instructions and test download:
http://archimago.blogspot.com/2015/04/internet-blind-test-linear-vs-minimum.html

Thanks,
Archimago



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[Wombat]

If you really used SoX with the steep filter of 99% it isn't exactly
real-world. The high amount of ringing introduced with 99% is completely
maintained in the 176kHz upsampled signal.
A DAC playing back the 44.1kHz signal never has such a steep filter
imho.



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[Archimago]

In doing this we can see if there is any significant preference among
respondents!

I want to know, EVEN with this amount of (pre-)ringing, whether
suppression with the minimum phase setting actually results in a
significant difference detected.

1. Is there significant *preference* in a naturalistic sample for one
setting vs. another?

2. Are listeners able to consistently "prefer" one variety over another
among the 3 samples? Ie. They keep preferring the same type of filter?
How many of these potential "golden ears" are there? Of course it's also
possible that the folks with these *systems* are particularly
susceptible to high frequency nonlinearities.

3. Can we detect cohort effect among musicians, production folks, and
even audio reviewers? I have been told for example that musicians and
those who record and produce music may be able to detect the differences
better...

In any event, the filter effect is clear. Trading strong pre-ringing at
22kHz for high frequency phase shift. Is there an audible difference and
can some people consistently tell?

We can deal with the idea of suppressing ringing and such in a future
test. One variable at a time :-).

BTW: I resample with these steep filters all the time. So I have vested
interest in changing how I do things based on the results!



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[Wombat]

My little critic shouldn't invalidate anything of your well done test.
It is only that many use filters with a transition band of 1-2 kHz to
avoid any problems steep filters may have.
On the other hand there are no real convincing arguments there is really
a problem, only some marketing papers, anectodes or faulty experiments
:)



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[jimmypowder]

I can tell the difference with nearfield monitors .In a hifi environment
,I doubt it.


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[Archimago]

jimmypowder wrote:
> I can tell the difference with nearfield monitors .In a hifi environment
> ,I doubt it.

JIMMY:

17939

Please, have a listen on the nearfield monitors! Let me know what you
hear!


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[Mnyb]

Good luck with this ! I can't be a test subject this time due to the
24/96 "limit" of my system .

However is any comparison done to the original ? I say if what if any of
the converted files sounds different ? They're really should not then we
might have got in the territory of pleasantly colouring artefacts . The
ones sounding different from the original is broken ?

Does any of the filters have a frequency drop inside of 20kHz ? There
have been many SoX filters suggested by some users over the years , some
quite unorthodox often made on the fly by for example convert.conf
settings in lms or by other means with other systems . Some of these
give a slight attenuation of the treble ? I always wondered if we just
have a fancy tone control here .



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[Archimago]

Mnyb wrote:
> Good luck with this ! I can't be a test subject this time due to the
> 24/96 "limit" of my system .
>
> However is any comparison done to the original ? I say if what if any of
> the converted files sounds different ? They're really should not then we
> might have got in the territory of pleasantly colouring artefacts . The
> ones sounding different from the original is broken ?
>
> Does any of the filters have a frequency drop inside of 20kHz ? There
> have been many SoX filters suggested by some users over the years , some
> quite unorthodox often made on the fly by for example convert.conf
> settings in lms or by other means with other systems . Some of these
> give a slight attenuation of the treble ? I always wondered if we just
> have a fancy tone control here .

Although suboptimal, you can certainly still give it a try Mnyb.
Squeezebox will downsample to 88kHz (?96) and assuming you're just using
a standard linear filter, you'll still maintain the general "effect".

No frequency drop in 20kHz as demonstrated in the frequency response
curve on the blog page. That's definitely one variable that had to be
controlled. Straight up STEEP filters with no significant frequency
response anomaly up to 20kHz except for "nasty" pre-echo ringing with
the linear setting and unavoidable phase distortion with minimum
setting.



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[Wombat]

Fun fact to me with your test is the idea not using a 192kHz source
against a lowpassed one. You may do better as a famous AES paper lately
claimed to :)
Well, not exactly but one of the reasons in the AES paper differences
may be heard is still the possibility of IM of music content in the
ultra hard metal tweeter.
Your test has no IM content only the added ringing at the filters
frequency. Brillant!



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[Archimago]

Wombat wrote:
> Fun fact to me with your test is the idea not using a 192kHz source
> against a lowpassed one. You may do better as a famous AES paper lately
> claimed to :)
> Well, not exactly but one of the reasons in the AES paper differences
> may be heard is still the possibility of IM of music content in the
> ultra hard metal tweeter.
> Your test has no IM content only the added ringing at the filters
> frequency. Brillant!

Don't know about the "brilliant" part...

I figure it was just obvious in order to isolate the variables :-).

Now if someone out there can explain to me what kind of "Filter
responses tested were representative of anti-alias filters used in A/D
(analog-to-digital) converters or mastering processes" settings these
people used, I would be most appreciative as I do not have access to
said "famous AES paper":
http://www.aes.org/e-lib/browse.cfm?elib=17497

I'm also a bit confused as to why this paper even bothers to mention
16-bit quantization and dithering at all... How's that supposed to fit
into the title "Audibility of Typical Digital Audio Filters in a
High-Fidelity Playback System"?

Perplexed...



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[Wombat]

Archimago wrote:
> Don't know about the "brilliant" part...
>
> I figure it was just obvious in order to isolate the variables :-).
>
> Now if someone out there can explain to me what kind of "Filter
> responses tested were representative of anti-alias filters used in A/D
> (analog-to-digital) converters or mastering processes" settings these
> people used, I would be most appreciative as I do not have access to
> said "famous AES paper":
> http://www.aes.org/e-lib/browse.cfm?elib=17497
>
> I'm also a bit confused as to why this paper even bothers to mention
> 16-bit quantization and dithering at all... How's that supposed to fit
> into the title "Audibility of Typical Digital Audio Filters in a
> High-Fidelity Playback System"?
>
> Perplexed...

No one really knows. The paper talks about a filter using a 500Hz
transition band realized with Matlab. People that know much more about
this stuff wanted to create own files but the paper does not include the
exact parameters you need. Matlab must have tons of possibilities
there.
It is a kind of joke that in a so called peer reviewed paper the data
for verifying the findings it is about are missing :)
The paper makes no claims about 16bit being inferior to higher bit
depth. They even used a bad dither method that no real-world resampler
would use today.
This may be also just another move to build a reason MQA does not need
all bits.
Like mentioned before elsewhere. This paper has many critics for just
being a Meridian marketing paper and buy-in at the AES.



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[Archimago]

Wombat wrote:
> No one really knows. The paper talks about a filter using a 500Hz
> transition band realized with Matlab. People that know much more about
> this stuff wanted to create own files but the paper does not include the
> exact parameters you need. Matlab must have tons of possibilities
> there.
> It is a kind of joke that in a so called peer reviewed paper the data
> for verifying the findings it is about are missing :)
> The paper makes no claims about 16bit being inferior to higher bit
> depth. They even used a bad dither method that no real-world resampler
> would use today.
> This may be also just another move to build a reason MQA does not need
> all bits.
> Like mentioned before elsewhere. This paper has many critics for just
> being a Meridian marketing paper and buy-in at the AES.

Thanks for the confirmation on this.

That's really quite ridiculous! Deservedly a -shame -for the AES for
publishing this if indeed there was some kind of scholarly peer-review
process applied and missed such an obvious omission.

Maybe they should print / publish a followup where the authors are asked
to submit details of the filters, or a retraction if these basics of the
experiment set-up cannot be provided?!



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[arnyk]

Archimago wrote:
> Thanks for the confirmation on this.
>
> That's really quite ridiculous! Deservedly a -shame -for the AES for
> publishing this if indeed there was some kind of scholarly peer-review
> process applied and missed such an obvious omission.
>
> Maybe they should print / publish a followup where the authors are asked
> to submit details of the filters, or a retraction if these basics of the
> experiment set-up cannot be provided?!

More information here:


Google:

"AES Convention Papers Forum" and a few keywords from the paper title.

I'd post the link if I could but I am currently prohibited from doing so
by the forum software.


Which is the AES papers forum for the paper that we are discussing
(readible by anybody, postable by AES members)


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[Wombat]

Welcome Mr. K. :)



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[garym]

Wombat wrote:
> Welcome Mr. K. :)



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[Julf]

Wombat wrote:
> Welcome Mr. K. :)

This is turning into a rather remarkable sub-forum. :)



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[adamdea]

Wombat wrote:
> No one really knows. The paper talks about a filter using a 500Hz
> transition band realized with Matlab. People that know much more about
> this stuff wanted to create own files but the paper does not include the
> exact parameters you need. Matlab must have tons of possibilities
> there.
> It is a kind of joke that in a so called peer reviewed paper the data
> for verifying the findings it is about are missing :)
> They even used a bad dither method that no real-world resampler would
> use today.
> This may be also just another move to build a reason MQA does not need
> all bits.
>

Is this the paper where they used rectangular pdf dither not triangular
despite the fact that Stuart knows fully well that rectangular PDF
dither does not remove all the quantisation distortion? I thought that
was simply outrageously intellectually bankrupt. Absolutely up there
with staircase graphs to prove that hi rez is better.


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[arnyk]

adamdea wrote:
> Is this the paper where they used rectangular pdf dither not triangular
>
>

TRUE.

>
> despite the fact that Stuart knows fully well that rectangular PDF
> dither does not remove all the quantisation distortion?
>

Read his explanation for that here:

https://secure.aes.org/forum/pubs/conventions/?ID=416

>
> I thought that was simply outrageously intellectually bankrupt.
> Absolutely up there with staircase graphs to prove that hi rez is
> better.

If you read the articles at the above link, you will see that the RPDF
dither was only one of several such asymmetries.


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[Archimago]

arnyk wrote:
> TRUE.
>
>
>
> Read his explanation for that here:
>
> https://secure.aes.org/forum/pubs/conventions/?ID=416
>
>
>
> If you read the articles at the above link, you will see that the RPDF
> dither was only one of several such asymmetries.

Thank you for the link Mr. Krueger!

Interesting discussion on the paradigm.

As for RPDF dithering as a "probe" for suboptimal DACs, it looks like
they acknowledge "the quantisation and dither tests were reported for
information but are not central to the point of the paper." Okay...

The last sentence of the Stuart reply: "we have continued this series of
experiments using different filters (including both shorter and
minimum-phase designs) and will be reporting these findings in the near
future." I'm sure we'll be looking forward to this paper.

So, I feel like I'm missing something here and curious about practical
implications:

1. As for the actual claims of "The Audibility of Typical Digital Audio
Filters in a High-Fidelity Playback System" per the title, what exactly
did they find here? (Other than suboptimal dithering being audible...)

2. What device uses 16-bit RPDF dithering which would be of any
significance for the hi-fi enthusiast these days?



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[arnyk]

Archimago wrote:
> Thank you for the link Mr. Krueger!
>
> So, I feel like I'm missing something here and curious about practical
> implications:
>
> 1. As for the actual claims of "The Audibility of Typical Digital Audio
> Filters in a High-Fidelity Playback System" per the title, what exactly
> did they find here? (Other than suboptimal dithering being audible...)
>
>

Since they didn't test filters that were "Typical Digital Audio Filters
in a High-Fidelity Playback System" and instead tested filters with a
misch-mash of parameters with inherently flawed listening tests of their
own contrivance; they found out what their inherently flawed listening
tests revealed related to that cats-and-dogs set of filters. The paper
was about straw men.

The other big straw man came from a lack of understanding of ABX testing
as it is practiced today in audio. They referenced a 1950 paper about an
early form of ABX testing that was not the interactive process that is
used in audio today. When corrected, they went back and referenced a
later paper that cited the same 1950 paper. I guess you can't teach old
dogs new tricks.

At any rate their excuse for ignoring ABX testing was a repetition of
the false claim that ABX is necessarily a 2AFC listening test which as a
practical matter is not true. ABX as practiced today is interactive and
therefore many choices about how to execute the test are up to the
listener. The listener has the option of using ABX as a same/different
test, and many say they exercise this option.

The paper's own test methodology locked their listeners into
fixed-length arbitrary samples, while many ABX testers prefer shorter
samples of their choosing. What we know about how people detect audible
differences favors shorter samples of the listener's choosing.

>
> 2. What device uses 16-bit RPDF dithering which would be of any
> significance for the hi-fi enthusiast these days?

I have no idea of any such device being commonly used. Hi Fi
enthusiasts generally only use DACs, and unlike ADCs, dither is not
central to the operation of DACs.

The predominate source of dither involved with a typical piece of hi fi
gear is generally the digital recording. If we adopt the model of
recordings being made with ADCs that have less noise then their sources
which seems reasonable, then the actual dithering noise in the recording
can easily be the background noise from acoustical and analog sources
that are an inherent part of the recording process. Analog noise
generally has a Gaussian PDF which more strongly resembles TPDF dither
than RPDF dither. 5


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[adamdea]

In the meantime I've just noticed this post on HA (I think Wombat
participated in the thread) which very neatly shows the two effects of a
steep filter namely pre-ringing and pre-echo (and post for each too).
The middle row (impulse) just shows ringing (the blurry horizontal line
showing spuriae in the transition band). This can also be seen in the
top row (castanets) The bottom row (tone sweep) shows this again with
also pre- and post echoes being the diagonal purple lines. I'm not quite
sure why the impulse (middle row) and casanets do not show pre and post
echo though.

http://www.hydrogenaud.io/forums/index.php?s=&showtopic=93588&view=findpost&p=895337


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[Archimago]

adamdea wrote:
> In the meantime I've just noticed this post on HA (I think Wombat
> participated in the thread) which very neatly shows the two effects of a
> steep filter namely pre-ringing and pre-echo (and post for each too).
> The middle row (impulse) just shows ringing (the blurry horizontal line
> showing spuriae in the transition band). This can also be seen in the
> top row (castanets) The bottom row (tone sweep) shows this again with
> also pre- and post echoes being the diagonal purple lines. I'm not quite
> sure why the impulse (middle row) and casanets do not show pre and post
> echo though.
>
> http://www.hydrogenaud.io/forums/index.php?s=&showtopic=93588&view=findpost&p=895337

Did the image get fixed? I'm seeing the pre and post-echo as it should
with the steep linear phase filter in the thread after applying the
20kHz filter (2nd column)...



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[Wombat]

Archimago wrote:
> Did the image get fixed? I'm seeing the pre and post-echo as it should
> with the steep linear phase filter in the thread after applying the
> 20kHz filter (2nd column)...
I don't think the picture was different before. The ringing is nicely
shown at the filters frequency where it belongs. Maybe adamdea expected
ringing in the audible band.
These spectral pics are much more clear with that as the typical graph
you see as marketing since ages.



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[utgg]

I'm coming late to all these discussions, so forgive me if I'm covering
old ground here.

As an engineer that has worked for many years in all sorts of fields
involved in signal processing, I've found the difficulties with
wide-band high resolution systems are mostly to do with non-linearities.
I suspect that if people can actually hear a difference between wide
bandwidth input vs. the same thing low-pass filtered - that we know
shouldn't be audible - this is probably a non-linearity defect in either
the reproduction equipment or receiving apparatus (i.e. the ears). Most
likely the ears.

In other words, those that think they've got 'golden ears' because they
can hear a difference maybe shouldn't be too proud - it could well be
because their ears are unusually non-linear, i.e. defective.

It could, of course, be a non-linear defect in the (expensive)
wide-bandwidth amplifier/speaker/listening environment combination being
revealed with this new-fangled high-sample rate source material. That
might be equally problematic....


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[Archimago]

utgg wrote:
> I'm coming late to all these discussions, so forgive me if I'm covering
> old ground here.
>
> As an engineer that has worked for many years in all sorts of fields
> involved in signal processing, I've found the difficulties with
> wide-band high resolution systems are mostly to do with non-linearities.
> I suspect that if people can actually hear a difference between wide
> bandwidth input vs. the same thing low-pass filtered - that we know
> shouldn't be audible - this is probably a non-linearity defect in either
> the reproduction equipment or receiving apparatus (i.e. the ears). Most
> likely the ears.
>
> In other words, those that think they've got 'golden ears' because they
> can hear a difference maybe shouldn't be too proud - it could well be
> because their ears are unusually non-linear, i.e. defective.
>
> It could, of course, be a non-linear defect in the (expensive)
> wide-bandwidth amplifier/speaker/listening environment combination being
> revealed with this new-fangled high-sample rate source material. That
> might be equally problematic....

Thanks for the note UTGG. I've heard a similar comment with
psychoacoustic encoding as well. I remember running into an older fellow
(mid 60's) with sensorineural hearing loss from acoustic trauma a number
of years back who claims he could hear the difference in MP3 even at
high bitrates. He attributed it to his own hearing damage such that the
normal mechanisms of auditory masking didn't "work" for him any more.

Good insight I thought in that he didn't just proclaim that "MP3
sucks!", but rather realized that he could hear things that were
"different" when his own kids and grandkids had no complaints.



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[arnyk]

utgg wrote:
> I'm coming late to all these discussions, so forgive me if I'm covering
> old ground here.
>
> As an engineer that has worked for many years in all sorts of fields
> involved in signal processing, I've found the difficulties with
> wide-band high resolution systems are mostly to do with non-linearities.
> I suspect that if people can actually hear a difference between wide
> bandwidth input vs. the same thing low-pass filtered - that we know
> shouldn't be audible - this is probably a non-linearity defect in either
> the reproduction equipment or receiving apparatus (i.e. the ears). Most
> likely the ears.
>
> In other words, those that think they've got 'golden ears' because they
> can hear a difference maybe shouldn't be too proud - it could well be
> because their ears are unusually non-linear, i.e. defective.
>
> It could, of course, be a non-linear defect in the (expensive)
> wide-bandwidth amplifier/speaker/listening environment combination being
> revealed with this new-fangled high-sample rate source material. That
> might be equally problematic....

The literature related to doing this kind of listening test contains
many examples of attention and inattention to the potential for
nonlinear distortion (e.g. IM) in the monitoring system to cause false
positives.

This pair of sample-rate-testing files contain the results of several
generations of trying to build a listening test that is self-diagnostic
for this problem: 'Link to files for studying the audible effects of
downsampling that are also self-diagnostic for IM'
(http://www.hydrogenaud.io/forums/index.php?showtopic=107570&view=findpost&p=894877)

The way this works is that the primary test is the classic "Keys
Jangling" sound originally recorded at 24/96 and then downsampled to
44/16 and upsampled back to 24/96. Following the keys jangling sound is
a brief low level marker tone followed by ultrasonic test tones
designed to elicit audible IM if there is excess nonlinear distortion in
the monitoring chain.

The intent is that an ABX file comparison tool such as Foobar2K with its
ABX plug-in (all freeware) are used to control the test.

In the basic test (keys jangling, before the audible test tone) the
intent is that you either hear or do not hear a difference. That's the
primary test.

Following the low level test tone is the secondary qualification test
for your monitoring chain. If you hear any difference between the files
in this test segment, then your monitoring chain has audible IM and any
positive results from the primary test are probably the results of that.
IOW, false positives.


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[utgg]

arnyk wrote:
> The literature related to doing this kind of listening test contains
> many examples of attention and inattention to the potential for
> nonlinear distortion (e.g. IM) in the monitoring system to cause false
> positives.
>
> This pair of sample-rate-testing files contain the results of several
> generations of trying to build a listening test that is self-diagnostic
> for this problem: 'Link to files for studying the audible effects of
> downsampling that are also self-diagnostic for IM'
> (http://www.hydrogenaud.io/forums/index.php?showtopic=107570&view=findpost&p=894877)
>
> The way this works is that the primary test is the classic "Keys
> Jangling" sound originally recorded at 24/96 and then downsampled to
> 44/16 and upsampled back to 24/96. Following the keys jangling sound is
> a brief low level marker tone followed by ultrasonic test tones
> designed to elicit audible IM if there is excess nonlinear distortion in
> the monitoring chain.
>
> The intent is that an ABX file comparison tool such as Foobar2K with its
> ABX plug-in (all freeware) are used to control the test.
>
> In the basic test (keys jangling, before the audible test tone) the
> intent is that you either hear or do not hear a difference. That's the
> primary test.
>
> Following the low level test tone is the secondary qualification test
> for your monitoring chain. If you hear any difference between the files
> in this test segment, (or hear anything but silence or a very faint low
> level rushing noise) then your monitoring chain has audible IM and any
> positive results from the primary test are probably the results of that.
> IOW, false positives.

Thanks for that Arnold. I'm glad you guys are doing your best to test
this properly, as a counter to the somewhat warped methods of those with
commercial interests at heart.

As an aside, I've recently updated one of my long term projects
(involved with genuine ultrasonic content) to take advantage of the
ready availability of low cost 192KHz ADC/DAC devices now. I'm really
impressed with the performance, but for me - especially with my ageing
ears - it is utterly pointless (and quite possibly detrimental) to use
such high sample rates for music.


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[jkeny]

Can anyone tell me the logic of this test, please? Is there not an issue
with testing the audible effect of recordings that used linear or
minimum phase filters when listening through a DAC that itself uses one
of these filters?


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[jkeny]

arnyk wrote:
> Yes, but based on past experience with you John, you are incapable of
> appreciating or benefiting from the explanation.
>
> Any reasonable explanation will be bent, folded, spindled, torn and
> mutilated.

Ok, I got it - you can't justify it


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[arnyk]

jkeny wrote:
> Ok, I got it - you can't justify it

Jkeny you seem to think that I am omniscient and control this test. In
fact it originated before the first time I ever posted on this forum.
Therefore it is an undeniable fact that I had no influence over its
parameters, and your continued bulling of me related to it is just more
evidence of the fantasy world in which you seem to live.


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[jkeny]

arnyk wrote:
> Jkeny you seem to think that I am omniscient and control this test. In
> fact it originated before the first time I ever posted on this forum.
> Therefore it is an undeniable fact that I had no influence over its
> parameters, and your continued bulling of me related to it is just more
> evidence of the fantasy world in which you seem to live.

I asked an open question to all objectivists about the validity of this
test - you chose to respond to my post with a slur but no answer.
So I get it - you can't justify it so why bother responding to my post -
let others respond who can answer my question.

As far as I can make out the logic of this test - it seems to me the
equivalent of recording two different DACs & then asking people to play
back this recording through their own DAC to see if they can
differentiate between the two.


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[arnyk]

jkeny wrote:
> I asked an open question to all objectivists about the validity of this
> test - you chose to respond to my post with a slur but no answer.
> So I get it - you can't justify it so why bother responding to my post -
> let others respond who can answer my question.
>

See, jkeny there you go again. How am I possibly so omniscient that I
prevented anybody from responding to your question?

>
> As far as I can make out the logic of this test - it seems to me the
> equivalent of recording two different DACs & then asking people to play
> back this recording through their own DAC to see if they can
> differentiate between the two.
>
> Am I missing something?
>

Yes, jkeny you are missing the fact that the test is not about DACs or
anything like them.

It is about a technical operation that can and often does happen with
nary a DAC in sight.

Therefore your interpretation of the logic of the test is false.

Furthermore, you are holding a perfectly valid kind of test of DACs up
to ridicule.


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[Wombat]

Safe your energy Arnold. This forum has a long history in letting people
talk about their daydreams. Over time here were several bizarre claims
made by well known überears no one should take to serious. Since this is
an "audiophile" sub forum no one really complains and in some way it
makes it a bit special and enjoyable. It even leads to some funny
writings when people explore the twilight zone. Just take it with a
smile :)



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[jkeny]

I don't see an answer in the two previous posts - just a lot of hand
waving.
According to ArchiMago's instructions there are 6 Flac files which he
wants people to play back through their DAC & state a preference. He
gives some useful DAC setup instructions & warnings.

But, as I said, is this test not attempting to differentiate recordings
of two filters by playing them back through a DAC which itself has a
filter?
How can these confounders be handled?


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[Wombat]

No DAC playing back 176.4kHz should touch the audible band where the
filters from upsampling act in these samples. Archimago surely already
has several reports. Please submit your listening results to him with a
detailed description of your setup when you have a 176.4kHz capable DAC.



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[jkeny]

Wombat wrote:
> No DAC playing back 176.4kHz should touch the audible band where the
> filters from upsampling act in these samples. Archimago surely already
> has several reports. Please submit your listening results to him with a
> detailed description of your setup when you have a 176.4kHz capable DAC.

OK, let's clear something up - what is being tested here - pre-echo on
all frequencies in the audio band or the Gibbs effect which only results
in ringing at frequencies around 22.05KHz?

Your post talks about "176.4KHz playback not touching anything in the
audible band" but 22.05KHz is not in the audible band, is it?


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[Wombat]

jkeny wrote:
> OK, let's clear something up - what is being tested here - pre-echo on
> all frequencies in the audio band or the Gibbs effect which only results
> in ringing at frequencies around 22.05KHz?
>
> Your post talks about "176.4KHz playback not touching anything in the
> audible band" but 22.05KHz is not in the audible band, is it?
The test is clearly explained and i don't know what you talk about.
There is no such thing as pre-echo at all frequencies. I doubt you
understand the basics. A DAC playing back 176.4kHz should not have a
filter doing anything to content at 22kHz. For anything higher there is
simply no content that can ring in these samples.



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[jkeny]

Wombat wrote:
> The test is clearly explained and i don't know what you talk about.
> There is no such thing as pre-echo at all frequencies. I doubt you
> understand the basics. A DAC playing back 176.4kHz should not have a
> filter doing anything to content at 22kHz. For anything higher there is
> simply no content that can ring in these samples.

Right, so your statement "No DAC playing back 176.4kHz should touch the
audible band " was meaningless to this test & only confuses matters. A
minimum phase filter

Let's see what Archimago did then - *"I took ~1 minute of these three
24/44 or 16/44 recordings and using SoX, upsampled them to 24/176.4 with
either the linear or minimum phase upsampling algorithm, re-naming one
file A and another B." *

So he used an upsampling algorithm of either minimum or linear phase to
bring these files to 24/176. Now tell me how pre-ringing at the Nyquist
frequency of 88KHz is of interest to anyone & what this test is intended
to do?


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[Wombat]

jkeny wrote:
> Right, so your statement "No DAC playing back 176.4kHz should touch the
> audible band " was meaningless to this test & only confuses matters. A
> minimum phase filter
>
> Let's see what Archimago did then - *"I took ~1 minute of these three
> 24/44 or 16/44 recordings and using SoX, upsampled them to 24/176.4 with
> either the linear or minimum phase upsampling algorithm, re-naming one
> file A and another B." *
>
> So he used an upsampling algorithm of either minimum or linear phase to
> bring these files to 24/176. Now tell me how pre-ringing at the Nyquist
> frequency of 88KHz is of interest to anyone & what this test is intended
> to do?
As before i doubt you understand the basics. Do the test or let it go
but please don't waste my time.



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[arnyk]

jkeny wrote:
> Right, so your statement "No DAC playing back 176.4kHz should touch the
> audible band " was meaningless to this test & only confuses matters. A
> minimum phase filter
>
> Let's see what Archimago did then - *"I took ~1 minute of these three
> 24/44 or 16/44 recordings and using SoX, upsampled them to 24/176.4 with
> either the linear or minimum phase upsampling algorithm, re-naming one
> file A and another B." *
>
> So he used an upsampling algorithm of either minimum or linear phase to
> bring these files to 24/176.
> > > >
> >
> > Correct.
> >
> > > > > > >
> > > Now tell me how pre-ringing at the Nyquist frequency of 88KHz is of
> > > interest to anyone & what this test is intended to do?> > > >
> >
> > That snide comment just shows the monumental amount of relevant
> > technical information that you don't know about upsampling, jkeny.
> >
> > Upsampling is done by zero-filling the input file, followed by a
> > brick wall low pass filter whose corner frequency is based on the
> > sample rate of the original file.
> >
> > The filtering is required because the original file has been
> > expanded into the output file by putting in samples whose value is
> > equal to zero to bring the number of samples up to the new, higher
> > number of samples. This naturally creates aliases of the original
> > audio data. By properly filtering the new file, we end up with a
> > higher sample rate file whose audio information is an excellent
> > approximation of the original flie.
> >
> > In the case of this test, the 44 KHz input data is upsampled to 176
> > KHz by means of zero filling which initially creates 3 aliases of
> > the input data. The 44 KHz low pass brick wall filter which is
> > implemented at a 176 KHz sample rate is supposed to knock out the 3
> > aliases leaving the original data clean and pure. If it is designed
> > well, then all is well.
> >
> > This test is about what constitutes a well designed 44 KHz filter
> > operating at 176 KHz. It is supposed to be inaudible.


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[jkeny]

Thank you for clarifying!


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[Wombat]

Maybe we can agree at least. This test only can show if upsampling with
a linear phase filter can sound different as with a minimum phase filter
and what sound people prefer. It doesn't even test what filter sounds
more like the original.
The files are pretty good at helping to test this without involving the
playback hardware sound to much.



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[adamdea]

Wombat wrote:
> The test is clearly explained and i don't know what you talk about.
> There is no such thing as pre-echo at all frequencies. I doubt you
> understand the basics. A DAC playing back 176.4kHz should not have a
> filter doing anything to content at 22kHz. For anything higher there is
> simply no content that can ring in these samples.
Although your other sentences are quite accurate, there actually is such
a thing as echo (distinct from ringing)
see para 2.1 of this and Graph E relating to each filter.
http://www.nanophon.com/audio/antialia.pdf

That said, I think that these days the FR rippling can be made
arbitrarily small, so that effect would be (dare I say it) insignificant


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[Wombat]

adamdea wrote:
> Although your other sentences are quite accurate, there actually is such
> a thing as echo (distinct from ringing)
> see para 2.1 of this and Graph E relating to each filter.
> http://www.nanophon.com/audio/antialia.pdf
>
> That said, I think that these days the FR rippling can be made
> arbitrarily small, so that effect would be (dare I say it) insignificant
I just did the ringing pic lately like posted before.
http://forums.slimdevices.com/showthread.php?103537-Internet-Blind-Test-Linear-vs-Minimum-Digital-Filters&p=816463&viewfull=1#post816463
The resolution of the pic was -110dB but i have to recheck. Using SoX
with -v there. Pre-echo was not visible to me. In the pics of bandpass
there are indeed these low level patterns most likely due to the extreme
steep experimental filter he designed for this test. You are right.



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[adamdea]

Wombat wrote:
> I just did the ringing pic lately like posted before.
> http://forums.slimdevices.com/showthread.php?103537-Internet-Blind-Test-Linear-vs-Minimum-Digital-Filters&p=816463&viewfull=1#post816463
> The resolution of the pic was -110dB but i have to recheck. Using SoX
> with -v there. Pre-echo was not visible to me. In the pics of bandpass
> there are indeed these low level patterns most likely due to the extreme
> steep experimental filter he designed for this test. You are right.

I'm intrigued as to why the echo only appears with the sweep in the
example and not with the castanets or impulse. I wonder whether there's
more energy in the sweep as it does seem to glow brightly.


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[arnyk]

adamdea wrote:
> I'm intrigued as to why the echo only appears with the sweep in the
> example and not with the castanets or impulse. I wonder whether there's
> more energy in the sweep as it does seem to glow brightly.

That is it! By definition an impulse being infinitesimally narrow but
with the same maximum amplitude as any other kind of test signal,
contains less energy. The castanets being a natural sound can pick up
some duration over the impulse, but by their nature their energy content
is limited. The sine sweep being a continuous tone has duration on its
side.

We've been through this same progression with speaker measurements. Back
in the day we knew we wanted the frequency response whose theoretical
synonym is Impulse Response, so we actually set up spark gaps and there
was your impulse signal to test with. But the amplitude of the test
signal is limited by the max SPL capabilities of the microphone. And
people have to be in the room and ear plugs are a hassle so really loud
noises are less practical. Then we tried pink noise and the like, which
drove up the energy levels by being more of a continuous signal, but the
average energy in noise is still less than it is with the sine wave.
Today, most of the speaker measurement systems are based on a swept sine
wave test signals.


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[adamdea]

Thanks Arny


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[Archimago]

We're 1/2 way through the test period! At present I'm at 35 detailed
responses which is not bad given the demands of the test...

Folks, if you've ever wanted to know whether this whole digital filters
effect with pre-ringing makes a difference for you, I'd highly recommend
giving the test a try and participate in the experiment. Satisfy your
curiosity while contributing with some data! As far as I know, this
would be the largest "public" trial of this sort and I think this can
give us some answers around the audibility of pre-ringing given the
(extreme!) type of filter setting I'm using here...



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[Archimago]

Happy Father's Day to all the dads...

Closing off the test on June 25th! Get your results in if you haven't
yet...



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[Archimago]

Results out!

'Part I: RESULTS'
(http://archimago.blogspot.ca/2015/07/the-linear-vs-minimum-phase-upsampling.html)

'Part II: ANALYSIS & CONCLUSIONS'
(http://archimago.blogspot.ca/2015/07/the-linear-vs-minimum-phase-upsampling_10.html)

Thanks to all who participated. :cool:



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[Julf]

Archimago wrote:
> Results out!
>
> 'Part I: RESULTS'
> (http://archimago.blogspot.ca/2015/07/the-linear-vs-minimum-phase-upsampling.html)
>
> 'Part II: ANALYSIS & CONCLUSIONS'
> (http://archimago.blogspot.ca/2015/07/the-linear-vs-minimum-phase-upsampling_10.html)
>
> Thanks to all who participated. :cool:

Interesting, as always! Thanks!



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[Wombat]

This looks like a lot of work again Archimago and i will read it
carefully.
Thank you very much btw. for mentioning me in the former article ;)
The spectral pics you offer are nice. Audition color sheme seems more
clear as the Audacity pics i tried.



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[Archimago]

Wombat wrote:
> This looks like a lot of work again Archimago and i will read it
> carefully.
> Thank you very much btw. for mentioning me in the former article ;)
> The spectral pics you offer are nice. Audition color sheme seems more
> clear as the Audacity pics i tried.

Thanks Wombat. Appreciate your insights and discussions off the forum
here over the years...



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[Archimago]

Lavorgna posted on this test...

Perverse that the guy bothers to comment on my blog contents yet I've
been banned from responding on his comments section. I guess he and
Plaskin had to resort to censorship at that place.

No choice but to leave a response on my blog: :)
'MUSINGS: Digital Filters Test Discussion'
(http://archimago.blogspot.com/2015/07/musings-digital-filters-test-discussion.html)



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[Wombat]

Archimago wrote:
> Lavorgna posted on this test...
>
> Perverse that the guy bothers to comment on my blog contents yet I've
> been banned from responding on his comments section. I guess he and
> Plaskin had to resort to censorship at that place.
>
> No choice but to leave a response on my blog: :)
> 'MUSINGS: Digital Filters Test Discussion'
> (http://archimago.blogspot.com/2015/07/musings-digital-filters-test-discussion.html)
Nice we come to a similar conclusion. I suggest SoX -b92 -a for some
years now :)



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[arnyk]

Archimago wrote:
> Lavorgna posted on this test...
>
> Perverse that the guy bothers to comment on my blog contents yet I've
> been banned from responding on his comments section. I guess he and
> Plaskin had to resort to censorship at that place.
>

We see similar trends over here on the slim devices forum. Who is it
that wants contrary viewpoints banned from *their* threads over here?


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[Archimago]

Wombat wrote:
> Nice we come to a similar conclusion. I suggest SoX -b92 -a for some
> years now :)

Good stuff :cool:



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[Mnyb]

What did you actually test was it a preference with listeners for a
certain filter ?

Is it addressed what filters gives inaudible differences when
downsampling from a hires original ?

As we before have reached the conclusion that hires can't be heard over
CD-res of what use is filter with a slight euphoric preference ? We all
have tone controls ?

A filter that makes a 16/44.1 down sample indistinguishable from 24/196
must be more correct than a filter than gives a slight euphonic
pleasantness to the sound ?



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[Archimago]

Mnyb wrote:
> What did you actually test was it a preference with listeners for a
> certain filter ?
>
> Is it addressed what filters gives inaudible differences when
> downsampling from a hires original ?
>
> As we before have reached the conclusion that hires can't be heard over
> CD-res of what use is filter with a slight euphoric preference ? We all
> have tone controls ?
>
> A filter that makes a 16/44.1 down sample indistinguishable from 24/196
> must be more correct than a filter than gives a slight euphonic
> pleasantness to the sound ?

No, regarding stuff on settings for downsampling, personally for myself
just an intellectual exercise to find something that doesn't ring much
(especially for those who believe they can hear the ringing/smearing),
frequency response flat to 20kHz (unlike stuff like the Ayre filter),
has minimal aliasing while doing the above, and remains linear phase
because I think linear phase is still "best practice" for accuracy. The
settings seem to be decent "objective tradeoffs" that should not result
in any subjective concerns / criticisms...

I welcome anyone who feels the need to ABX filter settings to go for it
:-).



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[Wombat]

If you asked me Mnyb, for similar reasons. SoX b 91-93 covers the
complete redbook spec from 20-20.000. Its aliasing is only allowed above
the passband. Especialy with some noise shaped dither used the little
aliasing above 20kHz is covered and can't cause any trouble.
To lazy to look it up again but non-linear spectral filter pics of
impulses showed frequency offsets into the audible band below 20kHz.
SoX is the most simple free choice imho.



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[Mnyb]

Archimago wrote:
>
> I welcome anyone who feels the need to ABX filter settings to go for it
> :-).

he he SoX itlself has some to go trough.... :))

Wombat wrote:
> If you asked me Mnyb, for similar reasons. SoX b 91-93 covers the
> complete redbook spec from 20-20.000. Its aliasing is only allowed above
> the passband. Especialy with some noise shaped dither used the little
> aliasing above 20kHz is covered and can't cause any trouble.
> To lazy to look it up again but non-linear spectral filter pics of
> impulses showed frequency offsets into the audible band below 20kHz.
> SoX is the most simple free choice imho.

Ok .

So some minimum phase filters have the frequency response slope before
20k not good if it start way before 20k.

My concern is that’s all there is to it , it's gets slightly softer and
whiz bang it's the next big thing for audiophiles ?



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[Archimago]

Mnyb wrote:
>
> Ok .
>
> So some minimum phase filters have the frequency response slope before
> 20k not good if it start way before 20k.
>
> My concern is that’s all there is to it , it's gets slightly softer and
> whiz bang it's the next big thing for audiophiles ?

Yes. IMO, it looks like this is exactly all it has ever been! We'll see
if Meridian and MQA is the next great thing in this chapter when we get
to see the kind of upsampling and filter settings they're promoting...
Perhaps that will be the pinnacle since Meridian was one of the first to
promote minimum phase and got the press going "ohhh... ahhh..."



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[Wombat]

Archimago wrote:
> Yes. IMO, it looks like this is exactly all it has ever been! We'll see
> if Meridian and MQA is the next great thing in this chapter when we get
> to see the kind of upsampling and filter settings they're promoting...
> Perhaps that will be the pinnacle since Meridian was one of the first to
> promote minimum phase and got the press going "ohhh... ahhh..."
Again i can repeat often enough that all tests that try to promote low
ringing show very, very low statistical value and even then only with
very, very strong ringing filters. A setting with a gentle filter
setting like these mentioned above is very, very likely all you need to
not worry anymore about ringring or voicing every piece of music
different because of some phase thing acting in the audible band.



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[Mnyb]

Wombat wrote:
> Again i can repeat often enough that all tests that try to promote low
> ringing show very, very low statistical value and even then only with
> very, very strong ringing filters. A setting with a gentle filter
> setting like these mentioned above is very, very likely all you need to
> not worry anymore about ringring or voicing every piece of music
> different because of some phase thing acting in the audible band.

So the typical oversampling DAC with a filter thats not your ca 1986
brickwall filter does it roughly rigth ?

I not versed in the exact technical details . I'm certain that there is
some kind of group of good compromises that gets its done like the SoX
settings you use . Actual chip implementations has and still probably
have their own set of problems . Hence the idea now that CPU power cost
nought to do this stuff with proper floating point math in software .



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[Julf]

Mnyb wrote:
> So the typical oversampling DAC with a filter thats not your ca 1986
> brickwall filter does it roughly rigth ?

Indeed.

> I not versed in the exact technical details . I'm certain that there is
> some kind of group of good compromises that gets its done like the SoX
> settings you use.

Yes - I think that is the "way beyond more than good enough" point.

> Hence the idea now that CPU power cost nought to do this stuff with
> proper floating point math in software .

Do we want to go there? I can already see it - "ah, but floating point
is never totally precise, so there is always room for improvement" :)



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
edge that will fool many people" - Paul W Klipsch, 1953
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[Wombat]

Mnyb wrote:
> So the typical oversampling DAC with a filter thats not your ca 1986
> brickwall filter does it roughly rigth ?
>
> I not versed in the exact technical details . I'm certain that there is
> some kind of group of good compromises that gets its done like the SoX
> settings you use . Actual chip implementations has and still probably
> have their own set of problems . Hence the idea now that CPU power cost
> nought to do this stuff with proper floating point math in software .
Surely old designs were already good enough with this but marketing has
to create problems to solve.
You see that even me suddenly wurries about things that most likely not
matter :)
We can phantasy around even more. Lets assume we use a filter that
filters softly at 20kHz for downsampling and i play it at my
Transporter. From a measurement at stereoplay i see it filters 44.1kHz
material steep around 20.8kHz. Since my SoX resampling setting kicks in
at ~20.28kHz the ringing of the 20.8kHz Transporter filter should not
matter because there is no loud content left to trigger but we get very
theoretical here.



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[Mnyb]

Julf wrote:
>
>
> Do we want to go there? I can already see it - "ah, but floating point
> is never totally precise, so there is always room for improvement" :)

Like that's not happening thousands of times for every track in a modern
DAW :D



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[Mnyb]

Wombat wrote:
> Surely old designs were already good enough with this but marketing has
> to create problems to solve.
> You see that even me suddenly wurries about things that most likely not
> matter :)
> We can phantasy around even more. Lets assume we use a filter that
> filters softly at 20kHz for downsampling and i play it at my
> Transporter. From a measurement at stereoplay i see it filters 44.1kHz
> material steep around 20.8kHz. Since my SoX resampling setting kicks in
> at ~20.28kHz the ringing of the 20.8kHz Transporter filter should not
> matter because there is no loud content left to trigger but we get very
> theoretical here.

Yes , but you can not have it as free software tweak you must build an
expensive hardware box then it will be the next thing surely :)



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[Julf]

Mnyb wrote:
> Like that's not happening thousands of times for every track in a modern
> DAW :D

Ssshhhhh! Don't tell them! :)



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
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[Mnyb]

Julf wrote:
> Ssshhhhh! Don't tell them! :)

*cough* can it be so that some plugins and effects in a DAW use Filters
what algorithm are used here ?



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[arnyk]

Mnyb wrote:
> So the typical oversampling DAC with a filter thats not your ca 1986
> brickwall filter does it roughly rigth ?
>

There was a sea change in DAC filters around Y2K and linear phase
filters became far more common.

Of course the only rule is that there are no rules. ;-)

Some modern DACs have switchable filters so you can play DAC filter de
jour with them.


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[Julf]

jkeny wrote:
> OK, let's clear something up - what is being tested here - pre-echo on
> all frequencies in the audio band or the Gibbs effect which only results
> in ringing at frequencies around 22.05KHz?
>

Seems you don't really understand the Gibbs effect either.



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[arnyk]

jkeny wrote:
> Right, so your statement "No DAC playing back 176.4kHz should touch the
> audible band " was meaningless to this test & only confuses matters. A
> minimum phase filter
>
> Let's see what Archimago did then - *"I took ~1 minute of these three
> 24/44 or 16/44 recordings and using SoX, upsampled them to 24/176.4 with
> either the linear or minimum phase upsampling algorithm, re-naming one
> file A and another B." *
>
> So he used an upsampling algorithm of either minimum or linear phase to
> bring these files to 24/176. Now tell me how pre-ringing at the Nyquist
> frequency of 88KHz is of interest to anyone & what this test is intended
> to do?

Julf wrote:
> Seems you don't really understand the Gibbs effect either.

From what I can see,
http://archimago.blogspot.com/2015/04/internet-blind-test-linear-vs-minimum.html

pretty well answered jkeny's question:

"As you can see by the images above, when we send a 16/44 "impulse"
through these reconstruction filters, we create different forms of
high-frequency ringing. Linear phase filters do not result in phase
distortion. However, the "issue" we are to note looking at the typical
linear phase impulse response is the symmetrical ringing as seen above.
It has been suggested that the pre-ringing ("pre-echo") is damaging to
the sound and "unnatural". As a result, minimum phase filter algorithms
became staples of companies like the aforementioned Meridian and later
on Ayre with their well known whitepaper on their filter choice (I see
this is also used in the recent PonoPlayer). Realize however that in
removing the pre-echo, minimum phase filters will result in high
frequency phase shift compared to the linear phase filter as shown here
(from the Infinite Wave website looking at SoX 14.4 VHQ Minimum Phase
vs. Linear phase):"

Perhaps we should ask a question that a reasonable person who knew very
little about digital technology could still answer, such as "What is
unclear about the above paragraph"?


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[Julf]

> Perhaps we should ask a question that a reasonable person who knew very
> little about digital technology could still answer, such as "What is
> unclear about the above paragraph"?

Indeed, but I would also love to hear how "the Gibbs effect only results
in ringing at frequencies around 22.05KHz"...



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[arnyk]

Julf wrote:
> Indeed, but I would also love to hear how "the Gibbs effect only results
> in ringing at frequencies around 22.05KHz"...

Of course, but I'm trying to create a situation where honest sincerity
would work for the person asking the question without making demands on
a resource that seems highly constrained, namely technical competence.

We both know that the Gibbs effect happens in the data domain when you
suddenly cut off all of the higher order harmonics. IOW you can
demonstrate the Gibbs effect perfectly with just math, which is more
fundamental than either analog audio or digital audio. You can do the
math in the time domain or the frequency domain and the results are the
same.

The bottom line is that the math says that when you brick wall low pass
filter a waveform a rapid change in the signal will cause ringing, and
that is how the ball bounces. IMO the cited paragraph covers that quite
clearly and in times past I would be mystified by any failure to
comprehend. However I have been working with placebophiles lately and
I now understand that their biases severely inhibit learning from
reliable sources.

So the question becomes how to overcome the perceptual difficulties of
placebophiles when it comes to technical discussions. My approach this
time was to direct the discussion so that honest sincerity would be the
critical resource, not technical competence.


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[arnyk]

jkeny wrote:
>
> But, as I said, is this test not attempting to differentiate recordings
> of two filters by playing them back through a DAC which itself has a
> filter?
>

Every audio system has many filters in it, some acoustical, some
electrical, some in the original recording, some added by the local
system.

Jkeny, if we let you assert that we can't hear differences in the
presence of any other filter but the one of interest, then we have no
justification for performing any listening tests, including listening
tests of your DACs.

>
> How can these confounders be handled?

By using what some of us (but obviously not yourself Mr. Jkeny) know
about audio technology to ensure that they don't interfere with each
other in a significant way. I've explained this to you before but like
it will predictably happen this time, the technical content shot over
your head, it would seem.


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[jkeny]

arnyk wrote:
> Every audio system has many filters in it, some acoustical, some
> electrical, some in the original recording, some added by the local
> system.
>
> Jkeny, if we let you assert that we can't hear differences in the
> presence of any other filter but the one of interest, then we have no
> justification for performing any listening tests, including listening
> tests of your DACs.
>
>
>
> By using what some of us (but obviously not yourself Mr. Jkeny) know
> about audio technology to ensure that they don't interfere with each
> other in a significant way. I've explained this to you before but like
> it will predictably happen this time, the technical content shot over
> your head, it would seem.

Can you show that the DAC's filter will not have an affect on the
playback of the recordings? This being the premise that underpins the
whole test.
It's not up to me to prove it does - it up to those who are suggesting
this as a valid test to prove this premise is correct

Surely the correct approach is to use a DAC which has switchable filters
& do your blind testing on it?


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[arnyk]

jkeny wrote:
> Can you show that the DAC's filter will not have an affect on the
> playback of the recordings? This being the premise that underpins the
> whole test.
>

Yes. The test runs at 24/192 which is a 96 KHz bandpass and the filters
being studied run around 22 KHz.

>
> Surely the correct approach is to use a DAC which has switchable filters
> & do your blind testing on it?

Wrong again. The design of the test puts the DAC filters up at 96 Khz,
while the test itself runs at 22 KHz. The 74 KHz difference keeps them
out of each other's way. Any reasonable 24/192 DAC filter will have its
brick wall effects well away from the test frequencies.


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[adamdea]

Archimago wrote:
> Did the image get fixed? I'm seeing the pre and post-echo as it should
> with the steep linear phase filter in the thread after applying the
> 20kHz filter (2nd column)...

Wombat wrote:
> I don't think the picture was different before. The ringing is nicely
> shown at the filters frequency where it belongs. Maybe adamdea expected
> ringing in the audible band.
> These spectral pics are much more clear with that as the typical graph
> you see as marketing since ages.
Hi. Only just noticed your replies to my post. I have tried to
illustrate what I meant on this here. I was under the impression that
pre/post ringing and and pre/post echo were two -different- things- the
latter being the time domain result of the rippling in the frequency
response of a filter. I assume that the two purple lines which appear in
the filtered frequency sweep [bottom right] represent pre/post-echo. I
only see this effect on the bottom row, whereas the ringing appears on
the first and second row too.

If I am wrong about this, please show where you think the pre-echo is.
If I have misidentified the purple lines in the effect in the bottom row
right image, then happy to be corrected
18132


+-------------------------------------------------------------------+
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[Archimago]

Wombat wrote:
> My little critic shouldn't invalidate anything of your well done test.
> It is only that many use filters with a transition band of 1-2 kHz to
> avoid any problems steep filters may have.
> On the other hand there are no real convincing arguments there is really
> a problem, only some marketing papers, anectodes or faulty experiments
> :)

No worries Wombat! I totally accept the critique and welcome it since
it's good to know and realistically present the findings (if any!).

Like you said, there are marketing papers out there and certain research
presented (often by parties with vested interests). I'd love to see we,
the consumers weigh in!



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[Wombat]

Archimago wrote:
> No worries Wombat! I totally accept the critique and welcome it since
> it's good to know and realistically present the findings (if any!).
>
> Like you said, there are marketing papers out there and certain research
> presented (often by parties with vested interests). I'd love to see we,
> the consumers weigh in!
Absolutely! If people don't hear day and night differences with this
strong ringing i doubt it ever can become a problem. Just read at CA
about every digit from 0-2000 changes sound "obviously" in the thread
about recommended iZotope settings :)
Unfortunately here is also no 176.4 native support.
Just for illustration a simple pic showing the ringing distribution for
sox 99% 44.1 -> 176.4, L and M


+-------------------------------------------------------------------+
|Filename: ring.jpg |
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[Archimago]

Wombat wrote:
> Absolutely! If people don't hear day and night differences with this
> strong ringing i doubt it ever can become a problem. Just read at CA
> about every digit from 0-2000 changes sound "obviously" in the thread
> about recommended iZotope settings :)
> Unfortunately here is also no 176.4 native support.
> Just for illustration a simple pic showing the ringing distribution for
> sox 99% 44.1 -> 176.4, L and M

Exactly!



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