[Arny Krueger[_4_]]

I have come across several references to an article on the topic of
so-called hi rez audio files
recently. It found it interesting - as is the site whence it came.

"24/192 Music Downloads ...and why they make no sense" is he

http://people.xiph.org/~xiphmont/demo/neil-young.html

[Audio Empire]

On Tue, 6 Mar 2012 16:28:58 -0800, Arny Krueger wrote
(in article ):

I have come across several references to an article on the topic of
so-called hi rez audio files
recently. It found it interesting - as is the site whence it came.

"24/192 Music Downloads ...and why they make no sense" is he

http://people.xiph.org/~xiphmont/demo/neil-young.html


Well written article and there's a lot of truth to it. However in his zeal to
make his point, the author has made a few assumptions that I don't think are
really in evidence.

1) In spite of the author's premise, I really don't think that ANYONE expects
a digital system frequency response beyond 22.5KHz to actually have any (or
at least not very much) actual program content.

(a) Few microphones have any response much above 20 KHz and most large
capsule mikes of the type generally used for recording have a huge resonance
peak somewhere between about 8 and 16 KHz and drop off rapidly above that.

(b) Those in the know realize that sampling rates of 88.2, 96, 176.4, 192,
and 384 KHz have merit simply because they move the Nyquist frequency WAY out
of the audio passband and that 88.2KHz is probably quite far enough with 96
and above merely being overkill.

2) His characterization of "golden ears" shows a basic lack of understanding
of the actual meaning of the term. He's right that nobody has the kind of
super hearing that he characterizes as being the definition of the
golden-eared audiophile. But it's a strawman argument because "super hearing"
is not what being "golden-eared" is all about. Most audiophiles these days
are over fifty. They certainly do not have super hearing. Many can't hear
much over 12 KHz, and if they have been exposed (either in their careers or,
more likely, being exposed to loud electronic rock-n-roll in their youth)
they may not hear that wide a frequency response, and yet they still posses
"golden ears". How is that possible, I hear some of you ask? Because all the
term "golden ears" means is that some sound enthusiasts care enough about the
sound of music to have trained themselves to listen for artifacts in
reproduced music and the equipment used to reproduce it and to identify and
quantify those artifacts. Things like ragged frequency response, different
types of distortion and their origin, problems in recordings and the ability
to tell a real stereo recording from a multi-channeled mono one. That's all
it is. When my friend J. Gordon Holt was in his late 70's, he could still
listen to a stereo system and tell you exactly what was wrong with it! Things
like: the phono cartridge is mis-tracking, the speakers have boomy bass, the
room has standing waves, The amplifier has high-frequency distortion, the
speakers are out of phase (a common one), etc., etc., etc.

3) He is right about 24-bit. It is a much better CAPTURE format than is
16-bit simply because it allows the recordist more headroom. 32-bit
floating-point recording is even better. The ideal LPCM capture format would
be 32-bit/88.2KHz. It moves the Nyquist frequency well outside the passband,
and it gives the engineer lots of headroom.

4) What he really should be attacking is the high-resolution audio content
provider's' insistence on using FLAC to deliver the files. FLAC is so flawed
that the way that these sites use it, it's anything but lossless!

Otherwise, the article is good reading and should be de riguer for anyone
contemplating purchasing high-res downloads.

[Sebastian Kaliszewski]

Audio Empire wrote:
4) What he really should be attacking is the high-resolution audio content
provider's' insistence on using FLAC to deliver the files. FLAC is so flawed
that the way that these sites use it, it's anything but lossless!


Would you care to elaborate on that point? What are the flaws of FLAC?

rgds
\SK
--
"Never underestimate the power of human stupidity" -- L. Lang
--
http://www.tajga.org -- (some photos from my travels)

[Audio Empire]

On Wed, 7 Mar 2012 05:24:16 -0800, Sebastian Kaliszewski wrote
(in article ):

Audio Empire wrote:
4) What he really should be attacking is the high-resolution audio content
provider's' insistence on using FLAC to deliver the files. FLAC is so
flawed
that the way that these sites use it, it's anything but lossless!


Would you care to elaborate on that point? What are the flaws of FLAC?

rgds
\SK


FLAC can be very good, but, according to Dr. Charles Zeilig and Mr. Jay
Clawson who have written several papers about digital audio software,
different software compressors available out there use different FLAC
compression rates and different versions of the FLAC standard. Two of the
most widely used compression schemes for FLAC are FLAC-8 and FLAC-Zero. When
compared to a standard uncompressed WAV file, both FLAC -Zero and FLAC-8
compression settings yielded noticeably and measureably poorer sound. Since
there's really no way for the consumer to know what setting is being used,
Dr. Zeilig recommends that the high-res download industry avoid supplying
downloaded high-resolution music in the FLAC or other compression formats,
if possible.

I am not allowed (by the moderators) to mention my source for this info, but
you can find it online by googling "FLAC vs WAV" or just "Dr, Charles Zeilig
and Jay Clawson".

[Andrew Haley]

Audio Empire wrote:
On Wed, 7 Mar 2012 05:24:16 -0800, Sebastian Kaliszewski wrote
(in article ):

Audio Empire wrote:
4) What he really should be attacking is the high-resolution audio content
provider's' insistence on using FLAC to deliver the files. FLAC is so
flawed
that the way that these sites use it, it's anything but lossless!


Would you care to elaborate on that point? What are the flaws of FLAC?

FLAC can be very good, but, according to Dr. Charles Zeilig and
Mr. Jay Clawson who have written several papers about digital audio
software, different software compressors available out there use
different FLAC compression rates and different versions of the FLAC
standard. Two of the most widely used compression schemes for FLAC
are FLAC-8 and FLAC-Zero. When compared to a standard uncompressed
WAV file, both FLAC -Zero and FLAC-8 compression settings yielded
noticeably and measureably poorer sound.

This doesn't make any sense. FLAC files are bit-for-bit perfect when
uncompressed. How can they sound different? I blame Audiophilia
Nervosa.

Andrew.

[Arny Krueger[_4_]]

"Audio Empire" wrote in message
...
On Wed, 7 Mar 2012 05:24:16 -0800, Sebastian Kaliszewski wrote
(in article ):

Audio Empire wrote:
4) What he really should be attacking is the high-resolution audio
content
provider's' insistence on using FLAC to deliver the files. FLAC is so
flawed
that the way that these sites use it, it's anything but lossless!


Would you care to elaborate on that point? What are the flaws of FLAC?

rgds
\SK


FLAC can be very good, but, according to Dr. Charles Zeilig and Mr. Jay
Clawson who have written several papers about digital audio software,
different software compressors available out there use different FLAC
compression rates and different versions of the FLAC standard.

Zelig and Clawson have made a few contributions to the literature of audio
nervosa, as another poster titled it. They are working outside of their area
of professional endeavor and are in print as having made a large number of
"Exceptional claims" that violate the laws of physics and reason.

Two of the
most widely used compression schemes for FLAC are FLAC-8 and FLAC-Zero.
When
compared to a standard uncompressed WAV file, both FLAC -Zero and FLAC-8
compression settings yielded noticeably and measureably poorer sound.

Since these are both lossless compresison schemes that return bit-perfect
copies of the origional files, it is impossible for their use to result in
measurably poorer sound. Any scheme that finds differences between
arithmetically identical streams of data would ordinarly be considered to be
greviously flawed.

[Sebastian Kaliszewski]

Audio Empire wrote:
On Wed, 7 Mar 2012 05:24:16 -0800, Sebastian Kaliszewski wrote
(in article ):

Audio Empire wrote:
4) What he really should be attacking is the high-resolution audio content
provider's' insistence on using FLAC to deliver the files. FLAC is so
flawed
that the way that these sites use it, it's anything but lossless!

Would you care to elaborate on that point? What are the flaws of FLAC?

rgds
\SK


FLAC can be very good, but, according to Dr. Charles Zeilig and Mr. Jay
Clawson who have written several papers about digital audio software,
different software compressors available out there use different FLAC
compression rates and different versions of the FLAC standard. Two of the
most widely used compression schemes for FLAC are FLAC-8 and FLAC-Zero.

Yet there are no such things. 8, 0, etc are so called compression levels --
those are command line options to set of 5-6 compression algorithm parameters
(i.e. order of linear prediction, block size, use of exchaustive model search,
use and type of so called midside coding (all technicalities of entropy
extracion step), and range (min and max value) of so called Rice partitions
orders -- again purely technical parameters of losless transcoding of fixed bit
length words (numebers) into variable length ones where more frequent ones are
shorter). Those levels translate into speed at which compression occurss and
also influence average compression effectivenes. Decompression speed is very
little affevted byt those, and decompression allways results in the *very*
*same* *bit* *exact* copy of the original audio data.

When
compared to a standard uncompressed WAV file, both FLAC -Zero and FLAC-8

When compared to a standard uncompressed WAV file there are *identical*.
Both "flac -0" and "flac -8" as well as "flac -5" as well as "flac -l 12 -e -M
-b 1152" generate files which decode to *bit* *exact* copies. Those men either
don't have a clue or are deliberately dishonest or the combination of the two.

compression settings yielded noticeably and measureably poorer sound.

The only thing which could vary is so called replaygain feature which can be
turned on (flac file must be processed by another tool from flac-toolset family
which calculates it) -- this feature is automatic *volume* setting per track or
per album. It's a pair of values stored in file header which describe average
(averaged using psychoacouaticak model which is an elaboration on RMS level
measurement) and peak levels. Those allow replay software or hradware to
automatically adjust volume so varius recording play at similar loudness. That
info must be explicitly added (as a result of additional processing) by someone
preparing flac files. End even if present this feature could be ignored or even
compeletly stripped out if one so desires.

ReplayGain has nothing to -0 or -8 or -whatever_digit, not -l nor -r nor -m nor
-M nor -b options (those options allow for finetuning of compression preformance
i.e. speed against effectiveness).

So, if those men wrote about that feature then all would be OK. But they did
not. So again, those men talk about stuff they have no clue about (or in fact
have but are dishonest).

Since
there's really no way for the consumer to know what setting is being used,

It doesn't matter. Result is bit exact.

Dr. Zeilig recommends that the high-res download industry avoid supplying
downloaded high-resolution music in the FLAC or other compression formats,
if possible.

Dr. Zeilig (who apparently has PhD in molecular biology not audio engeneering,
nor acoustics, not even physics nor physiology, and is known of similar antics
since eighties) spouts utter nonsense.


I am not allowed (by the moderators) to mention my source for this info, but
you can find it online by googling "FLAC vs WAV" or just "Dr, Charles Zeilig
and Jay Clawson".

Again, FLAC is fully lossless format - i.e. it's bit exact. You could add WAV
header to some application install file or even to this very text. You can then
compress it using any compressor with any lossless FLAC compression sheme then
uncompress and congert it bvack to WAV it using different uncompressor, then
strip that WAV header and you'd get the exact copy of that file or newsgroup text.


BTW found juicy fragments of that text series by those men, one is he

"Although JRMC reported an accurate rip for all the speeds, and are bit-for-bit
identical at all read speeds, we are still able to detect sonic differences in
the resulting file. We know these results drive engineers crazy. We would love
it if someone could come up with a definitive explanation that could provide
input to software developers."

I have an explanation but it's not to be shown in public forum due to
significant amount of "explicit lyrics" and acustations off either serius mental
shortcommings or serious, bordering criminal, dishonesty.


IOW. all that could be dismissed safely -- they are either dishonest or
incapable or both -- but for the reader all those options mean the same: dismiss
it as it's just a load of nonsense.

rgds
\SK
--
"Never underestimate the power of human stupidity" -- L. Lang
--
http://www.tajga.org -- (some photos from my travels)

[Audio Empire]

On Thu, 8 Mar 2012 16:03:22 -0800, Sebastian Kaliszewski wrote
(in article ):

snip
IOW. all that could be dismissed safely -- they are either dishonest or
incapable or both -- but for the reader all those options mean the same:
dismiss
it as it's just a load of nonsense.

Thanks, I read their "papers" and thought that the "points" system they used
to evaluate playback "quality" was a bit suspicious, but the the articles
seemed so "scholarly" that I accepted the results (until such time as those
results were successfully challenged - as you have done). As for myself I've
never used FLAC. I use Apple Lossless Compression in iTunes to rip my CDs for
my iPod Touch and my Logitech Squeezebox. In an admittedly flawed DBT*
between the ALC file and the CD itself, I could detect no difference between
the two with multiple samples tried over multiple sessions.

*The CD player and the Squeezebox Touch fed the same SPDIF inputs of a DAC
via a digital input selector switch, and both played through the same
amplifier at the same volume control settings through the same input, etc. A
second party switched between the two digital inputs using a remote control.
I couldn't see him (he was sitting behind me) and I had no idea when or even
if he switched inputs. Not scientific, perhaps, but both sources were at
EXACTLY the same level. My trusty HP3400A audio voltmeter assured me of that.




rgds
\SK

[Andrew Haley]

Audio Empire wrote:

4) What he really should be attacking is the high-resolution audio
content provider's' insistence on using FLAC to deliver the
files. FLAC is so flawed that the way that these sites use it, it's
anything but lossless!

What is wrong with FLAC?

Andrew.

[Arny Krueger[_4_]]

"Audio Empire" wrote in message
...
On Tue, 6 Mar 2012 16:28:58 -0800, Arny Krueger wrote
(in article ):

I have come across several references to an article on the topic of
so-called hi rez audio files
recently. It found it interesting - as is the site whence it came.

"24/192 Music Downloads ...and why they make no sense" is he

http://people.xiph.org/~xiphmont/demo/neil-young.html


Well written article and there's a lot of truth to it. However in his zeal
to
make his point, the author has made a few assumptions that I don't think
are
really in evidence.

1) In spite of the author's premise, I really don't think that ANYONE
expects
a digital system frequency response beyond 22.5KHz to actually have any
(or
at least not very much) actual program content.

I just answered a post from a fairly widely respected member of a pro audio
forum who was bragging about how much 20 KHz content he was seeing in
recordings.

(a) Few microphones have any response much above 20 KHz and most large
capsule mikes of the type generally used for recording have a huge
resonance
peak somewhere between about 8 and 16 KHz and drop off rapidly above that.

He was rebutting this exact issue. FWIW I agree with you.

(b) Those in the know realize that sampling rates of 88.2, 96, 176.4, 192,
and 384 KHz have merit simply because they move the Nyquist frequency WAY
out
of the audio passband and that 88.2KHz is probably quite far enough with
96
and above merely being overkill.

Let's face it, digital filtering has improved to the point where such
humungeous (2 cotave) guard bands serve no purpose at all.

2) His characterization of "golden ears" shows a basic lack of
understanding
of the actual meaning of the term. He's right that nobody has the kind of
super hearing that he characterizes as being the definition of the
golden-eared audiophile. But it's a strawman argument because "super
hearing"
is not what being "golden-eared" is all about. Most audiophiles these days
are over fifty. They certainly do not have super hearing. Many can't hear
much over 12 KHz, and if they have been exposed (either in their careers
or,
more likely, being exposed to loud electronic rock-n-roll in their youth)
they may not hear that wide a frequency response, and yet they still
posses
"golden ears". How is that possible, I hear some of you ask?

My answer is: denial. Case in point. A few weeks ago I was sitting next to a
reasonably well-known *name* in high end audio at a listening session. I was
complaining vigorously about the audible hum and noise. A number of people
around us shared my concern but Mr. name said that he heard nothing wrong.

Because all the
term "golden ears" means is that some sound enthusiasts care enough about
the
sound of music to have trained themselves to listen for artifacts in
reproduced music and the equipment used to reproduce it and to identify
and
quantify those artifacts. Things like ragged frequency response, different
types of distortion and their origin, problems in recordings and the
ability
to tell a real stereo recording from a multi-channeled mono one. That's
all
it is.

I'm on both sides of this argument. To some degree effectiveness at audio is
about both raw ability but it is also, and perhaps more preeminently as you
seem to be saying, about identifying the various sounds in what you hear.
However, through the magic of sighted evaluations there is also a large
subset of audiophiles and professionals whose hearing seems to be mostly in
their wallets. Any time you want to you can expose them with a blind test,
whether stealthy or in the open. Many have become too crafty to trap that
way. It's not a homogenious world out there.

When my friend J. Gordon Holt was in his late 70's, he could still
listen to a stereo system and tell you exactly what was wrong with it!
Things
like: the phono cartridge is mis-tracking, the speakers have boomy bass,
the
room has standing waves, The amplifier has high-frequency distortion, the
speakers are out of phase (a common one), etc., etc., etc.

Of course, and us blind testing advocates agree that a lot of problems like
those don't need blind tests to identify or prove. These can all be faults
and artifacts that are well above the well-known thresholds of audibility,
or not.

3) He is right about 24-bit. It is a much better CAPTURE format than is
16-bit simply because it allows the recordist more headroom. 32-bit
floating-point recording is even better. The ideal LPCM capture format
would
be 32-bit/88.2KHz. It moves the Nyquist frequency well outside the
passband,
and it gives the engineer lots of headroom.

On balance, if you know what you are doing you can make great-sounding
recordings with 16/44. 24 bits gets you 144 dB dynamic range, but in fact
audio gear that performs at the even *just* 20 bit level is still not sold
on every street corner. There are few live venues and recording studios that
have even 13 bits of acosutical dynamic range.

4) What he really should be attacking is the high-resolution audio content
provider's' insistence on using FLAC to deliver the files. FLAC is so
flawed
that the way that these sites use it, it's anything but lossless!

I don't know about that. In my tests and usage FLAC has proven itself to be
bit-perfect and sonically ideal. However, I have to admit that its
relatively gentle approximately 2:1 compression is not why I like it or use
it. What I like about FLAC is its support for tagging that thoroughly
eclipses what legacy .wav files support.

Otherwise, the article is good reading and should be de riguer for anyone
contemplating purchasing high-res downloads.

Especially when combined what is known about 50% of extant so-called hi rez
recordings having been through low-rez (typically analog recording)
production steps that limit their as-delivered performance to something like
12 bit resolution and CD format bandpass, but with more response variations.

[Audio Empire]

On Wed, 7 Mar 2012 16:12:03 -0800, Arny Krueger wrote
(in article ):


BTW, Arny. Welcome back. We've missed you!


(b) Those in the know realize that sampling rates of 88.2, 96, 176.4, 192,
and 384 KHz have merit simply because they move the Nyquist frequency WAY
out
of the audio passband and that 88.2KHz is probably quite far enough with
96
and above merely being overkill.

Let's face it, digital filtering has improved to the point where such
humungeous (2 cotave) guard bands serve no purpose at all.

While digital filtering is , as you say, very good, the precognitive nature
of the digital filter makes "pre-ringing" a condition that doesn't exist in
analog filtering. If one were to use a high sample rate (say, 88.2 KHz), then
gentle, analog filtering could be done therefore eliminating the pre-ringing.


2) His characterization of "golden ears" shows a basic lack of
understanding
of the actual meaning of the term. He's right that nobody has the kind of
super hearing that he characterizes as being the definition of the
golden-eared audiophile. But it's a strawman argument because "super
hearing"
is not what being "golden-eared" is all about. Most audiophiles these days
are over fifty. They certainly do not have super hearing. Many can't hear
much over 12 KHz, and if they have been exposed (either in their careers
or,
more likely, being exposed to loud electronic rock-n-roll in their youth)
they may not hear that wide a frequency response, and yet they still
posses
"golden ears". How is that possible, I hear some of you ask?

My answer is: denial. Case in point. A few weeks ago I was sitting next to a
reasonably well-known *name* in high end audio at a listening session. I was
complaining vigorously about the audible hum and noise. A number of people
around us shared my concern but Mr. name said that he heard nothing wrong.

OK, this, of course can happen. The title "Golden-eared Audiophile", is,
after all, self-annointed, and as such often gets applied to people who are,
to quote Clint Eastwood as 'Dirty Harry' , "...legends in their own minds."
OTOH, I have known more than a few audiophiles who have developed a fine
sense of listening acuity and can hear many things that the average listener
doesn't hear. There is a big difference between between "can't hear" and
"doesn't hear" Real Golden Eared types understand this difference and many of
them realize that being golden-eared does NOT in any way disqualify one from
being susceptible to sighted and other kinds of expectational bias. Golden
Ears are useful tools, to be sure, but when comparing things, they are no
substitute for the well executed bias-controlled test.

Because all the
term "golden ears" means is that some sound enthusiasts care enough about
the
sound of music to have trained themselves to listen for artifacts in
reproduced music and the equipment used to reproduce it and to identify
and
quantify those artifacts. Things like ragged frequency response, different
types of distortion and their origin, problems in recordings and the
ability
to tell a real stereo recording from a multi-channeled mono one. That's
all
it is.

I'm on both sides of this argument. To some degree effectiveness at audio is
about both raw ability but it is also, and perhaps more preeminently as you
seem to be saying, about identifying the various sounds in what you hear.
However, through the magic of sighted evaluations there is also a large
subset of audiophiles and professionals whose hearing seems to be mostly in
their wallets. Any time you want to you can expose them with a blind test,
whether stealthy or in the open. Many have become too crafty to trap that
way. It's not a homogenious world out there.

Amen there. The old "this costs 10-times what that costs, so this must be
better" syndrome is hard to fight. That's why DBTs are so important. OTOH,
some people just want nice things and would buy the 10X component even if a
DBT showed both units to perform identically.

When my friend J. Gordon Holt was in his late 70's, he could still
listen to a stereo system and tell you exactly what was wrong with it!
Things
like: the phono cartridge is mis-tracking, the speakers have boomy bass,
the
room has standing waves, The amplifier has high-frequency distortion, the
speakers are out of phase (a common one), etc., etc., etc.

Of course, and us blind testing advocates agree that a lot of problems like
those don't need blind tests to identify or prove. These can all be faults
and artifacts that are well above the well-known thresholds of audibility,
or not.

Yep. Listening and Objective or bias-controlled testing are not mutually
exclusive concepts. They need to be used together in order to make truly
informed music system selections and to stay away from the snake-oil.

3) He is right about 24-bit. It is a much better CAPTURE format than is
16-bit simply because it allows the recordist more headroom. 32-bit
floating-point recording is even better. The ideal LPCM capture format
would
be 32-bit/88.2KHz. It moves the Nyquist frequency well outside the
passband,
and it gives the engineer lots of headroom.

On balance, if you know what you are doing you can make great-sounding
recordings with 16/44. 24 bits gets you 144 dB dynamic range, but in fact
audio gear that performs at the even *just* 20 bit level is still not sold
on every street corner. There are few live venues and recording studios that
have even 13 bits of acosutical dynamic range.

In a studio environment, this is absolutely true. However it is less true in
location recording. The ability to have plenty of headroom is more than just
a luxury. Sometimes it is the difference between success and failure -
especially if you are recording a group cold.

4) What he really should be attacking is the high-resolution audio content
provider's' insistence on using FLAC to deliver the files. FLAC is so
flawed
that the way that these sites use it, it's anything but lossless!

I don't know about that. In my tests and usage FLAC has proven itself to be
bit-perfect and sonically ideal. However, I have to admit that its
relatively gentle approximately 2:1 compression is not why I like it or use
it. What I like about FLAC is its support for tagging that thoroughly
eclipses what legacy .wav files support.

Well a lot of research has been done by a Dr. Charles Zeilig whose tests seem
to show that FLAC algorithms vary all over the place and different settings
yield different quality results. Apparently FLAC-Zero and FLAC-8 sound
significantly poorer than does the WAV file from which it was compressed.

Otherwise, the article is good reading and should be de riguer for anyone
contemplating purchasing high-res downloads.

Especially when combined what is known about 50% of extant so-called hi rez
recordings having been through low-rez (typically analog recording)
production steps that limit their as-delivered performance to something like
12 bit resolution and CD format bandpass, but with more response variations.

I don't think that the fact that many of the sources for so-called Hi-Rez
downloads are analog tapes is particularly important, but I do think that the
fact that many recordings being sold as 24-bit 96 KHz and above are really
red book masters that have up-sampled to 24/96 is a rip-off and those guilty
of selling up-sampled standard resolution digital files as true high-rez
should be punished. To put that another way:

Digitizing an analog master tape with a 24-bit/88.2 KHz (or higher )ADC to
yield a "high-rez" copy is OK, but up-sampling a standard resolution digital
master to 24/88.2 KHz or higher is cheating. Of course, that's just my
opinion, you understand.

[Arny Krueger[_4_]]

"Audio Empire" wrote in message
...
On Wed, 7 Mar 2012 16:12:03 -0800, Arny Krueger wrote

Let's face it, digital filtering has improved to the point where such
humungeous (2 cotave) guard bands serve no purpose at all.

While digital filtering is , as you say, very good, the precognitive
nature
of the digital filter makes "pre-ringing" a condition that doesn't exist
in
analog filtering. If one were to use a high sample rate (say, 88.2 KHz),
then

It turns out that sharp-cutoff digital filtering can be tuned so that it has
pre-ringing, post-ringing, or anything in-between. These days most digital
filter parameters are calculated using sophisticated mathematical tools such
as Matlab. Worked out examples can be found he
http://www.mathworks.com/products/ds...lpfirdemo.html.

Digital Filters can be designed to be minimum phase, in which case there is
no pre-ringing at all, just like conventional analog filters. Commonly they
are designed to be linear phase or zero-phase which implies pre-ringing.
Digital filtering of many kinds is easily implemented with software and
applied to high sample rate digital audio files.DBTs have been done
comparing various approaches. The general outcome has been that the 22.05
KHz Nyquist frequency of Red Book audio is high enough that there is
already considerable margin, and the details of the filter design are fairly
non-critical.

[Wrettyirraple]

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[Audio Empire]

On Thu, 8 Mar 2012 16:05:07 -0800, Arny Krueger wrote
(in article ):

"Audio Empire" wrote in message
...
On Wed, 7 Mar 2012 16:12:03 -0800, Arny Krueger wrote

Let's face it, digital filtering has improved to the point where such
humungeous (2 cotave) guard bands serve no purpose at all.

While digital filtering is , as you say, very good, the precognitive
nature
of the digital filter makes "pre-ringing" a condition that doesn't exist
in
analog filtering. If one were to use a high sample rate (say, 88.2 KHz),
then

It turns out that sharp-cutoff digital filtering can be tuned so that it has
pre-ringing, post-ringing, or anything in-between. These days most digital
filter parameters are calculated using sophisticated mathematical tools such
as Matlab. Worked out examples can be found he

http://www.mathworks.com/products/dsp-
system/demos.html?file=/products/demos/s
hipping/dsp/lpfirdemo.html.

Digital Filters can be designed to be minimum phase, in which case there is
no pre-ringing at all, just like conventional analog filters. Commonly they
are designed to be linear phase or zero-phase which implies pre-ringing.
Digital filtering of many kinds is easily implemented with software and
applied to high sample rate digital audio files.DBTs have been done
comparing various approaches. The general outcome has been that the 22.05
KHz Nyquist frequency of Red Book audio is high enough that there is
already considerable margin, and the details of the filter design are fairly
non-critical.

Thanks for the links.