[Industrial One]

How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates? The clips in question
are 48 KHz 128 kbps CBR/VBR and a couple at 96. I don't understand why
those assholes can't downsample to 32 KHz if they're gonna use such
low bitrates.

[Randy Yates]

Industrial One writes:

How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.
--
% Randy Yates % "Remember the good old 1980's, when
%% Fuquay-Varina, NC % things were so uncomplicated?"
%%% 919-577-9882 % 'Ticket To The Moon'
%%%% % *Time*, Electric Light Orchestra
http://www.digitalsignallabs.com

[Industrial One]

On 15 Okt., 02:41, Randy Yates wrote:
Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

No ****. I said "improve quality," not "replace."

Btw, you should know that machines are capable of whatever we can
already do. If I can imagine in my mind a high quality version of the
audio without the ringing from a low quality source, a computer can do
the same.

[isw]

In article
,
Industrial One wrote:

On 15 Okt., 02:41, Randy Yates wrote:
Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

No ****. I said "improve quality," not "replace."

Btw, you should know that machines are capable of whatever we can
already do. If I can imagine in my mind a high quality version of the
audio without the ringing from a low quality source, a computer can do
the same.

It's not enough that you can *imagine* it; you must know *precisely* how
to *do* it. Then you can teach a computer how to do it, only a lot
faster than you can.

Or alternately, the computer can indeed "imagine" it, but can no more
deliver that as a usable output than you can "output" what you imagined.

Isaac

[Industrial One]

On Oct 15, 4:02 am, isw wrote:
In article
,
Industrial One wrote:

On 15 Okt., 02:41, Randy Yates wrote:
Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

No ****. I said "improve quality," not "replace."

Btw, you should know that machines are capable of whatever we can
already do. If I can imagine in my mind a high quality version of the
audio without the ringing from a low quality source, a computer can do
the same.

It's not enough that you can *imagine* it; you must know *precisely* how
to *do* it. Then you can teach a computer how to do it, only a lot
faster than you can.

Or alternately, the computer can indeed "imagine" it, but can no more
deliver that as a usable output than you can "output" what you imagined.

Isaac

I know, which is why I'm asking this group for suggestions. There must
be a way, just like there's a way to improve the quality of low-
bitrate DivX clips by applying a deblocking algorithm -- the most
advanced out there cannot completely restore the original quality but
still looks WAY better than if you left it alone. So what's my best
option? To leave my song as it is?

[jamesgangnc]

"Industrial One" wrote in message
...
On Oct 15, 4:02 am, isw wrote:
In article
,
Industrial One wrote:

On 15 Okt., 02:41, Randy Yates wrote:
Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

No ****. I said "improve quality," not "replace."

Btw, you should know that machines are capable of whatever we can
already do. If I can imagine in my mind a high quality version of the
audio without the ringing from a low quality source, a computer can do
the same.

It's not enough that you can *imagine* it; you must know *precisely* how
to *do* it. Then you can teach a computer how to do it, only a lot
faster than you can.

Or alternately, the computer can indeed "imagine" it, but can no more
deliver that as a usable output than you can "output" what you imagined.

Isaac

I know, which is why I'm asking this group for suggestions. There must
be a way, just like there's a way to improve the quality of low-
bitrate DivX clips by applying a deblocking algorithm -- the most
advanced out there cannot completely restore the original quality but
still looks WAY better than if you left it alone. So what's my best
option? To leave my song as it is?

Your best option is to go find a better original.

[Richard Crowley]

"Randy Yates" wrote ...
Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

Hence the term "lossy compression".
Does I1 have a list of troll questions that he posts regularly?

[Eeyore]

Industrial One wrote:

On 15 Okt., 02:41, Randy Yates wrote:
Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

No ****. I said "improve quality," not "replace."

There are different software decoders I think. Or is it just encoders Try
some anyway..

Graham

[Randy Yates]

Industrial One writes:

On 15 Okt., 02:41, Randy Yates wrote:
Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

No ****. I said "improve quality," not "replace."

Which is a ridiculous request. Sorta like "improving the quality" of the
output of an 8-bit A/D. The noise is (or in your case, artifacts are)
there to stay.
--
% Randy Yates % "So now it's getting late,
%% Fuquay-Varina, NC % and those who hesitate
%%% 919-577-9882 % got no one..."
%%%% % 'Waterfall', *Face The Music*, ELO
http://www.digitalsignallabs.com

[Mr.T]

"Randy Yates" wrote in message
...
Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

No ****. I said "improve quality," not "replace."

Which is a ridiculous request. Sorta like "improving the quality" of the
output of an 8-bit A/D. The noise is (or in your case, artifacts are)
there to stay.

Sure, but since he doesn't define what HE means by "improve", maybe he *can*
do it.
IF silence is an "improvement" (sure is in many cases IMO) then it's
actually very EASY! :-)

MrT.

[Chronic Philharmonic]

"Industrial One" wrote in message
...
On 15 Okt., 02:41, Randy Yates wrote:
Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

No ****. I said "improve quality," not "replace."

It depends on what you mean by quality and what Mr. Yates means by quality.
Information in the original waveform was discarded in the encoding process,
so fidelity to the original sound is irretrievably lost. All you can do now
is fiddle with it to see if you can find some further distortion that is
more to your liking.

Btw, you should know that machines are capable of whatever we can
already do. If I can imagine in my mind a high quality version of the
audio without the ringing from a low quality source, a computer can do
the same.

I can imagine all sorts of things that no computer is or ever will be
capable of.

[Richard Crowley]

"Industrial One" wrote from Goooooooooogle Groups...
Btw, you should know that machines are capable of whatever we can
already do.

Nominated for silliest remark of the year.
But then it is only mid-October.

If "I-1" keeps up the good work, he will make 1st
Class Troll and give Troll Emeritus "Radium" a run
for his position.

[Industrial One]

On Oct 15, 3:39 pm, "Richard Crowley" wrote:
"Randy Yates" wrote ...

Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

Hence the term "lossy compression".
Does I1 have a list of troll questions that he posts regularly?

"I1..." I kinda like it, despite how robotic it sounds.

On Oct 15, 9:56 pm, Randy Yates wrote:
Which is a ridiculous request. Sorta like "improving the quality" of the
output of an 8-bit A/D. The noise is (or in your case, artifacts are)
there to stay.

Bull****, the noise can be removed via noise-removal techniques and re-
saved as 16-bit.

On Oct 16, 12:46 am, "Mr.T" MrT@home wrote:
"Randy Yates" wrote in message

...

Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

No ****. I said "improve quality," not "replace."

Which is a ridiculous request. Sorta like "improving the quality" of the
output of an 8-bit A/D. The noise is (or in your case, artifacts are)
there to stay.

Sure, but since he doesn't define what HE means by "improve", maybe he *can*
do it.
IF silence is an "improvement" (sure is in many cases IMO) then it's
actually very EASY! :-)

MrT.

**** you Mr.T. Didn't I tell you to stay outta my threads?

On Oct 16, 5:22 am, "Chronic Philharmonic"
wrote:
"Industrial One" wrote in message

...

On 15 Okt., 02:41, Randy Yates wrote:
Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

No ****. I said "improve quality," not "replace."

It depends on what you mean by quality and what Mr. Yates means by quality.
Information in the original waveform was discarded in the encoding process,
so fidelity to the original sound is irretrievably lost. All you can do now
is fiddle with it to see if you can find some further distortion that is
more to your liking.

By quality I mean presentability. By running a smart deblocking algo
on a low-bitrate DivX clip, do I "restore information?" Not exactly,
but I interpolate/extrapolate the information I already have to make
the video much more presentable and perceivably higher quality. How do
you think your own mind can simulate a higher quality image of the one
you seen on your grainy TV? Some information is "gone" but the
information already present makes it obvious what would be there if it
wasn't gone. Neural networks just aren't at the stage yet where it can
restore images automatically without heavy human guidance.

I'm asking if the same can be done for sound. Can I "de-smear" and "de-
ring" it? If you insist I can't, then ok.

Btw, you should know that machines are capable of whatever we can
already do. If I can imagine in my mind a high quality version of the
audio without the ringing from a low quality source, a computer can do
the same.

I can imagine all sorts of things that no computer is or ever will be
capable of.

Like how the members of the Fraunhofer committee back in 1984 thought
consumer CPUs will never reach the stage to decode MP3s in real-time?

[Arny Krueger]

"Industrial One" wrote in
message


How can I improve the quality of a clip that suffers from
smearing/ ringing artifacts due to low-medium bitrates?

You might be able do obtain a perceived improvement with filtering and noise
gating.

The clips in question are 48 KHz 128 kbps CBR/VBR and a
couple at 96. I don't understand why those assholes can't
downsample to 32 KHz if they're gonna use such low
bitrates.

Downsampling to 32 KHz can actually improve the results when you use such
low bitrates.

32 KHz isn't that ugly of a sample rate - it allows some kind of frequency
response up to about 16 KHz. Please remember that FM stereo pretty well tops
out at 15 KHz.

Some general rules for coding to low bitrates are to forget stereo and go to
mono, and decrease the bandwidth as much as you can without losing too much
intelligibility. Spoken word in mono with a 5 to 8 Hz bandwidth isn't
usually all that bad, and classical music with 11 KHz bandwidth can often be
quite satisfying.

[Industrial One]

Arny Krueger wrote:
"Industrial One" wrote in
message


How can I improve the quality of a clip that suffers from
smearing/ ringing artifacts due to low-medium bitrates?

You might be able do obtain a perceived improvement with filtering and noise
gating.

Such as?

The clips in question are 48 KHz 128 kbps CBR/VBR and a
couple at 96. I don't understand why those assholes can't
downsample to 32 KHz if they're gonna use such low
bitrates.

Downsampling to 32 KHz can actually improve the results when you use such
low bitrates.

Duh, all of my MP3s in the past were 32 KHz 96-128 kbps. No artifacts
at all. Unfortunetaly, I can't control how other retards encode their
material, and the **** I downloaded was some old anime ripped from a
Laserdisc. I really doubt there is a higher quality copy available on
the net beside the one I snatched which already took forever to
download.

32 KHz isn't that ugly of a sample rate - it allows some kind of frequency
response up to about 16 KHz. Please remember that FM stereo pretty well tops
out at 15 KHz.

I doubt there is any significant difference at all, as most can't hear
over 16 khz anyway. I'm 18 and can hear up to 17, which is probably
why I sometimes notice a difference if I concentrate really hard. For
all intents and purposes, even 22 KHz is allright -- you lose some
cymbals but meh.

Some general rules for coding to low bitrates are to forget stereo and go to
mono, and decrease the bandwidth as much as you can without losing too much
intelligibility. Spoken word in mono with a 5 to 8 Hz bandwidth isn't
usually all that bad, and classical music with 11 KHz bandwidth can often be
quite satisfying.

**** on that ****! With the advent of spectral band replication and
parametric stereo there's no need for downsampling or downmixing
anymore.

[[email protected]]

On Oct 16, 10:39 am, Industrial One
wrote:
I'm 18

And that explains everything.

and can hear up to 17,

have the technical skills of 15, social skills of 12, and
most of the time act like a 2 year old.

How can I improve the quality of a clip that suffers
from smearing/ ringing artifacts due to low-medium
bitrates?

Perhaps your best choice is to not use the computer without
your Mom and Dad's permission.

[Arny Krueger]

"Industrial One" wrote in
message

Arny Krueger wrote:
"Industrial One" wrote in
message


How can I improve the quality of a clip that suffers
from smearing/ ringing artifacts due to low-medium
bitrates?

You might be able do obtain a perceived improvement with
filtering and noise gating.

Such as?

At really low bit rates there is often background noise and echos. Low pass
filtering can mitigate some of the irritation due to the noise, and a noise
gate can help with the some of the background noise and some of the echoes.

[Industrial One]

On Oct 16, 3:12 pm, wrote:
On Oct 16, 10:39 am, Industrial One
wrote:

I'm 18

And that explains everything.

and can hear up to 17,

have the technical skills of 15, social skills of 12, and
most of the time act like a 2 year old.

How can I improve the quality of a clip that suffers
from smearing/ ringing artifacts due to low-medium
bitrates?

Perhaps your best choice is to not use the computer without
your Mom and Dad's permission.

You got a problem?

P.S. I own this computer and apartment. My mom probably OD'ed and my
dad is in a nuthouse.

On Oct 16, 3:21 pm, "Arny Krueger" wrote:
"Industrial One" wrote in


Arny Krueger wrote:
"Industrial One" wrote in
message


How can I improve the quality of a clip that suffers
from smearing/ ringing artifacts due to low-medium
bitrates?

You might be able do obtain a perceived improvement with
filtering and noise gating.
Such as?

At really low bit rates there is often background noise and echos. Low pass
filtering can mitigate some of the irritation due to the noise, and a noise
gate can help with the some of the background noise and some of the echoes.

Oh **** it, the audio stays. The problem is not the noise and removing
any echo would probably remove legitimate reverb effects in the audio.
I don't even know why I'm bitching. It doesn't sound bad, it's just
not not up to par to the quality it could've had. Oh well, I doubt the
mental lonely ****s on eBay would care after they buy my "remastered"
copies. The picture is fine and that's all I care about.

[[email protected]]

On Oct 16, 11:45 am, Industrial One
wrote:
Perhaps your best choice is to not use the computer without
your Mom and Dad's permission.

You got a problem?

Nope, but it seems you're willing to share yours with
the world.

P.S. I own this computer and apartment. My mom
probably OD'ed and my dad is in a nuthouse.

No, they're probably hiding under a rock, regretting the
day they didn't pay attention to the "birth control" chapter
in sex ed.

I don't even know why I'm bitching

Because you're an unsocialized annoying little ass
with poor impulse control whose best skill is attention
seeking behavior. Once in a great while, you're a
source of mild entertainment.

[Randy Yates]

Industrial One writes:

Arny Krueger wrote:
"Industrial One" wrote in
message


How can I improve the quality of a clip that suffers from
smearing/ ringing artifacts due to low-medium bitrates?

You might be able do obtain a perceived improvement with filtering and noise
gating.

Such as?

The clips in question are 48 KHz 128 kbps CBR/VBR and a
couple at 96. I don't understand why those assholes can't
downsample to 32 KHz if they're gonna use such low
bitrates.

Downsampling to 32 KHz can actually improve the results when you use such
low bitrates.

Duh, all of my MP3s in the past were 32 KHz 96-128 kbps. No artifacts
at all. Unfortunetaly, I can't control how other retards encode their
material, and the **** I downloaded was some old anime ripped from a
Laserdisc. I really doubt there is a higher quality copy available on
the net beside the one I snatched which already took forever to
download.

32 KHz isn't that ugly of a sample rate - it allows some kind of frequency
response up to about 16 KHz. Please remember that FM stereo pretty well tops
out at 15 KHz.

I doubt there is any significant difference at all, as most can't hear
over 16 khz anyway. I'm 18 and can hear up to 17, which is probably
why I sometimes notice a difference if I concentrate really hard. For
all intents and purposes, even 22 KHz is allright -- you lose some
cymbals but meh.

I agree.

Some general rules for coding to low bitrates are to forget stereo and go to
mono, and decrease the bandwidth as much as you can without losing too much
intelligibility. Spoken word in mono with a 5 to 8 Hz bandwidth isn't
usually all that bad, and classical music with 11 KHz bandwidth can often be
quite satisfying.

**** on that ****! With the advent of spectral band replication and
parametric stereo there's no need for downsampling or downmixing
anymore.

So you've read http://en.wikipedia.org/wiki/Parametric_Stereo?

These are some impressive new developments in audio ENCODING - won't
help you too much with DECODING your files.
--
% Randy Yates % "She's sweet on Wagner-I think she'd die for Beethoven.
%% Fuquay-Varina, NC % She love the way Puccini lays down a tune, and
%%% 919-577-9882 % Verdi's always creepin' from her room."
%%%% % "Rockaria", *A New World Record*, ELO
http://www.digitalsignallabs.com

[Industrial One]

On Oct 16, 4:39 pm, wrote:
On Oct 16, 11:45 am, Industrial One
[SNIP]
**** on that ****! With the advent of spectral band replication and
parametric stereo there's no need for downsampling or downmixing
anymore.

So you've readhttp://en.wikipedia.org/wiki/Parametric_Stereo?

Yeah, wrote part of it too.

These are some impressive new developments in audio ENCODING - won't
help you too much with DECODING your files.

No ****... I was replying to Arny who said audio should be downmixed
and downsampled if one aims to save space. So I explained why it ain't
necessary. I really wish them dickbrains would ditch MP3 and start
using MP4. I recently downloaded this 100-meg 9-hour trance
collection track. I didn't even realize it was so long until I noticed
the duration tab displayed an extra digit for this song. It's at
24kbps mono, 22 KHz. A ****LOAD of quality could've been preserved if
the retard ripper used AAC-HEv2.

[Industrial One]

On Oct 16, 4:39 pm, wrote:
On Oct 16, 11:45 am, Industrial One
wrote:

Perhaps your best choice is to not use the computer without
your Mom and Dad's permission.

You got a problem?

Nope, but it seems you're willing to share yours with
the world.

P.S. I own this computer and apartment. My mom
probably OD'ed and my dad is in a nuthouse.

No, they're probably hiding under a rock, regretting the
day they didn't pay attention to the "birth control" chapter
in sex ed.

I don't even know why I'm bitching

Because you're an unsocialized annoying little ass
with poor impulse control whose best skill is attention
seeking behavior. Once in a great while, you're a
source of mild entertainment.

Aww, poor Dickpierce. I got someone to cheer you up: www.goatse.cz

FEEL THE STRETCH!

[Chronic Philharmonic]

"Industrial One" wrote in message
...
On Oct 15, 3:39 pm, "Richard Crowley" wrote:
"Randy Yates" wrote ...

Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

Hence the term "lossy compression".
Does I1 have a list of troll questions that he posts regularly?

"I1..." I kinda like it, despite how robotic it sounds.

On Oct 15, 9:56 pm, Randy Yates wrote:
Which is a ridiculous request. Sorta like "improving the quality" of the
output of an 8-bit A/D. The noise is (or in your case, artifacts are)
there to stay.

Bull****, the noise can be removed via noise-removal techniques and re-
saved as 16-bit.

If that were true, we'd just save everything as 8-bits, and do the noise
removal. Noise removal techniques are iffy at best, and obnoxious at worst,
even when meticulously tuned and applied by hand.

[Chronic Philharmonic]

"Randy Yates" wrote in message
...
Industrial One writes:


[...]

So you've read http://en.wikipedia.org/wiki/Parametric_Stereo?

These are some impressive new developments in audio ENCODING - won't
help you too much with DECODING your files.

Interesting article. I thought it was quite telling that the effect doesn't
work particularly well at higher bitrates. Of course, other encoding schemes
use sum and difference, taking advantage of the fact that the difference
between the two channels channels is usually much smaller than the mono sum.
That goes all the way back to stereo encoding on vinyl as well as FM and TV
stereo, and later, FLAC, et. al.

[Arny Krueger]

"Chronic Philharmonic" wrote in
message
"Randy Yates" wrote in message
...
Industrial One writes:


[...]

So you've read
http://en.wikipedia.org/wiki/Parametric_Stereo?

These are some impressive new developments in audio
ENCODING - won't help you too much with DECODING your
(existing) files.

Interesting article. I thought it was quite telling that
the effect doesn't work particularly well at higher
bitrates.

I suspect that it works no better or worse at higher bitrates in an absolute
sense, but it is not as acceptable because listener expectations are so much
higher at higher bitrates.

The mention of Satellite radio in one of the Wiki articles is telling,
because the audio quality standards for the best known satellite radio
network in the U.S. are abysmal. They might be good enough for Howard Stern
or a NASCAR race, but they are not for what most people here would call
quality audio.

Of course, other encoding schemes use sum and
difference, taking advantage of the fact that the
difference between the two channels channels is usually
much smaller than the mono sum.

IOW, you don't need a high quality, full-bandpass difference channel to
create the perception of space and directionality.

That goes all the way back to stereo encoding on vinyl as well as FM and
TV
stereo, and later,

For most of the life of FM stereo, real world FM stereo receivers
characteristically lost lots of separation at high frequencies.

FLAC, et. al.

AFAIK FLAC is lossless, and makes no compromises at all.

[Chronic Philharmonic]

"Arny Krueger" wrote in message
...
"Chronic Philharmonic" wrote in
message
"Randy Yates" wrote in message
...
Industrial One writes:


[...]


[...]

FLAC, et. al.

AFAIK FLAC is lossless, and makes no compromises at all.

Right, but they store the sum and difference, rather than essentially
duplicating the majority of both channels. It is lossless, but not wasteful.

[isw]

In article ,
"Arny Krueger" wrote:

-- snip --

For most of the life of FM stereo, real world FM stereo receivers
characteristically lost lots of separation at high frequencies.

Having worked on some of the earliest FM stereo encoders which actually
*met* all the FCC specifications, I would say that a lot of the problem
was with *encoders*, not decoders. The degree of matching (both
amplitude and phase) required between the L and R low-pass filters
necessary for good HF separation was not widely understood -- and even
less often realized.

With the advent of digital filtering techniques, things got a *lot*
easier.

Isaac

[Randy Yates]

isw writes:

In article ,
"Arny Krueger" wrote:

-- snip --

For most of the life of FM stereo, real world FM stereo receivers
characteristically lost lots of separation at high frequencies.

Having worked on some of the earliest FM stereo encoders which actually
*met* all the FCC specifications, I would say that a lot of the problem
was with *encoders*, not decoders. The degree of matching (both
amplitude and phase) required between the L and R low-pass filters
necessary for good HF separation was not widely understood -- and even
less often realized.

With the advent of digital filtering techniques, things got a *lot*
easier.

Having implemented a full BTSC decoder in the digital domain about a
year and a half ago, I can say from personal experience that it's not
all that easy. The time I spent on the various filters involved - trying
to get them designed to the required accuracy - was very painful.

Not to detract, however, from your correct point, isw, that implementing
a good encoder/decoder (for FM or analog TV broadcast) is no mean feat.
--
% Randy Yates % "Watching all the days go by...
%% Fuquay-Varina, NC % Who are you and who am I?"
%%% 919-577-9882 % 'Mission (A World Record)',
%%%% % *A New World Record*, ELO
http://www.digitalsignallabs.com

[Industrial One]

On Oct 17, 4:02 am, "Chronic Philharmonic"
wrote:
"Industrial One" wrote in message

...



On Oct 15, 3:39 pm, "Richard Crowley" wrote:
"Randy Yates" wrote ...

Industrial One writes:
How can I improve the quality of a clip that suffers from smearing/
ringing artifacts due to low-medium bitrates?

You cannot. You cannot replace information that has been lost.

Hence the term "lossy compression".
Does I1 have a list of troll questions that he posts regularly?

"I1..." I kinda like it, despite how robotic it sounds.

On Oct 15, 9:56 pm, Randy Yates wrote:
Which is a ridiculous request. Sorta like "improving the quality" of the
output of an 8-bit A/D. The noise is (or in your case, artifacts are)
there to stay.

Bull****, the noise can be removed via noise-removal techniques and re-
saved as 16-bit.

If that were true, we'd just save everything as 8-bits, and do the noise
removal. Noise removal techniques are iffy at best, and obnoxious at worst,
even when meticulously tuned and applied by hand.

Because it's useless if I'm gonna compress to MP3 since it'll smear
and **** up the noise, making it harder to detect and remove. But as
long as the noise dB are significantly lower than the signal, it can
be easily removed, especially by hand.

[Earl Kiosterud]

"Chronic Philharmonic" wrote in message
...


"Randy Yates" wrote in message ...
Industrial One writes:


[...]

So you've read http://en.wikipedia.org/wiki/Parametric_Stereo?

These are some impressive new developments in audio ENCODING - won't
help you too much with DECODING your files.

Interesting article. I thought it was quite telling that the effect doesn't work
particularly well at higher bitrates. Of course, other encoding schemes use sum and
difference, taking advantage of the fact that the difference between the two channels
channels is usually much smaller than the mono sum. That goes all the way back to stereo
encoding on vinyl as well as FM and TV stereo, and later, FLAC, et. al.


I know this is getting off-topic, but I thought it might be interesting to point out the
there wasn't really any "encoding" of stereo as such on vinyl. The two channels
independently moved the stylus, each at 45° (thus at 90° to each other). Today it's called
"discrete" channels. The result was that if there was no LR difference, then the stylus
moved only laterally, which means that a mono record would play properly on a stereo system.
That's also why stereo records would not play properly on a mono cartridge, because it
probably wasn't designed to allow much vertical movement, and would cause damage to the
extent that there was LR difference. In the worst case of LR difference, such as where one
channel was the same stuff as the other, but of inverse polarity, the stylus moved only
vertically.
--
Earl

[Chronic Philharmonic]

"Earl Kiosterud" wrote in message
news

"Chronic Philharmonic" wrote in message
...


"Randy Yates" wrote in message
...
Industrial One writes:


[...]

So you've read http://en.wikipedia.org/wiki/Parametric_Stereo?

These are some impressive new developments in audio ENCODING - won't
help you too much with DECODING your files.

Interesting article. I thought it was quite telling that the effect
doesn't work particularly well at higher bitrates. Of course, other
encoding schemes use sum and difference, taking advantage of the fact
that the difference between the two channels channels is usually much
smaller than the mono sum. That goes all the way back to stereo encoding
on vinyl as well as FM and TV stereo, and later, FLAC, et. al.


I know this is getting off-topic, but I thought it might be interesting to
point out the there wasn't really any "encoding" of stereo as such on
vinyl. The two channels independently moved the stylus, each at 45° (thus
at 90° to each other). Today it's called "discrete" channels. The result
was that if there was no LR difference, then the stylus moved only
laterally, which means that a mono record would play properly on a stereo
system. That's also why stereo records would not play properly on a mono
cartridge, because it probably wasn't designed to allow much vertical
movement, and would cause damage to the extent that there was LR
difference. In the worst case of LR difference, such as where one channel
was the same stuff as the other, but of inverse polarity, the stylus moved
only vertically.

I would respectfully argue that "encoding" is whatever you do to get audio
onto the disc -- mono or stereo. The implementation with some (perhaps all)
cutting heads and playback pickups might have been 45/45, but it is
mathematically identical to L+R (lateral) and L-R (vertical). A 45/45 pickup
can play back a record made with a L+R (lateral)/L-R (vertical) cutter,
without modification.

Statistically, L+R is more closely correlated than L-R, so there is less
vertical activity on average. Not only that, but this encoding allows the
L+R amplitude to be limited separately from L-R (L-R limiting would reduce
channel separation momentarily). Not only that, but the signals could be
equalized separately, so less bass is sent to the L-R channel. This reduces
the risk of the cutter losing contact with the surface, avoiding excessive
distortion and skips on playback.

I suppose we could start a new topic if there is any further interest in
this.

[Serge Auckland[_2_]]

"Chronic Philharmonic" wrote in message
...


"Earl Kiosterud" wrote in message
news

"Chronic Philharmonic" wrote in message
...


"Randy Yates" wrote in message
...
Industrial One writes:


[...]

So you've read http://en.wikipedia.org/wiki/Parametric_Stereo?

These are some impressive new developments in audio ENCODING - won't
help you too much with DECODING your files.

Interesting article. I thought it was quite telling that the effect
doesn't work particularly well at higher bitrates. Of course, other
encoding schemes use sum and difference, taking advantage of the fact
that the difference between the two channels channels is usually much
smaller than the mono sum. That goes all the way back to stereo encoding
on vinyl as well as FM and TV stereo, and later, FLAC, et. al.


I know this is getting off-topic, but I thought it might be interesting
to point out the there wasn't really any "encoding" of stereo as such on
vinyl. The two channels independently moved the stylus, each at 45°
(thus at 90° to each other). Today it's called "discrete" channels. The
result was that if there was no LR difference, then the stylus moved only
laterally, which means that a mono record would play properly on a stereo
system. That's also why stereo records would not play properly on a mono
cartridge, because it probably wasn't designed to allow much vertical
movement, and would cause damage to the extent that there was LR
difference. In the worst case of LR difference, such as where one
channel was the same stuff as the other, but of inverse polarity, the
stylus moved only vertically.

I would respectfully argue that "encoding" is whatever you do to get audio
onto the disc -- mono or stereo. The implementation with some (perhaps
all) cutting heads and playback pickups might have been 45/45, but it is
mathematically identical to L+R (lateral) and L-R (vertical). A 45/45
pickup can play back a record made with a L+R (lateral)/L-R (vertical)
cutter, without modification.

Statistically, L+R is more closely correlated than L-R, so there is less
vertical activity on average. Not only that, but this encoding allows the
L+R amplitude to be limited separately from L-R (L-R limiting would reduce
channel separation momentarily). Not only that, but the signals could be
equalized separately, so less bass is sent to the L-R channel. This
reduces the risk of the cutter losing contact with the surface, avoiding
excessive distortion and skips on playback.

I suppose we could start a new topic if there is any further interest in
this.

But as far as I am aware, there never has been any separate processing of
the L+R and L-R, only L and R separately, but using the same EQ and
compressor settings. On rock recordings, the L-R was necessarily minimised
by mixing kick drums and sometimes bass to centre, with the vocalist almost
always dead centre. Classical and Jazz tended to have more L-R, but as the
music wasn't so heavily compressed, the levels were lower anyway. It is
essential that any L,R processing be done with identical settings as
otherwise the central image will wander depending on frequency content and
level. There were some mono/stereo compatible records (Synchro Stereo was
one I recall) which I understand mixed low frequencies to mono and thus kept
the L-R signal small whilst still offering a noticeable stereo effect. I
have several such records of classical music, and they sound adequate in
stereo, but have sufficiently small L-R levels that they can be played with
a mono pickup without damage.

S.
--
http://audiopages.googlepages.com

[Arny Krueger]

"Serge Auckland" wrote in
message
"Chronic Philharmonic" wrote
in message
...


"Earl Kiosterud" wrote in message
news

"Chronic Philharmonic" wrote
in message
...


"Randy Yates" wrote in message
...

I know this is getting off-topic, but I thought it
might be interesting to point out the there wasn't
really any "encoding" of stereo as such on vinyl. The
two channels independently moved the stylus, each at
45° (thus at 90° to each other). Today it's called
"discrete" channels. The result was that if there was
no LR difference, then the stylus moved only laterally,
which means that a mono record would play properly on a
stereo system. That's also why stereo records would not
play properly on a mono cartridge, because it probably
wasn't designed to allow much vertical movement, and
would cause damage to the extent that there was LR
difference. In the worst case of LR difference, such
as where one channel was the same stuff as the other,
but of inverse polarity, the stylus moved only
vertically.

I would respectfully argue that "encoding" is whatever
you do to get audio onto the disc -- mono or stereo.

Agreed - the LP was an example of encoding an electrical signal into a
mechanical signal. It was descended from an earlier process that encoded an
acoustical signal as a mechanical signal.

The implementation with some (perhaps all) cutting heads and
playback pickups might have been 45/45, but it is
mathematically identical to L+R (lateral) and L-R
(vertical). A 45/45 pickup can play back a record made
with a L+R (lateral)/L-R (vertical) cutter, without
modification.

At some point in the processing of audio recorded on LPs, the signal was
turned every which way but lose, and sum/difference processing was very
common because of its impact on trackability on very modest playback
equipment.

Statistically, L+R is more closely correlated than L-R,
so there is less vertical activity on average.

Not only that, but vertical (L-R) dynamic range is far more limited than
horizontal (L+R) dynamic range. You run out of vertical dynamic range when
the cutting stylus digs a hole or becomes airborne. Both can happen and did
happen in the real world.

You run out of horizontal dynamic range when the cutting stylus loops
through an adjacent groove or creates a radius that can't be tracked by the
probable playback stylus. The adjacent groove problem can be managed by
increasing the pitch (space between adjacent tracks) of the grooves.
Increasing pitch cuts the amount of time that you can record.

The problem of creating radii that can't be tracked can be managed by using
smaller radii, which was really what elliptical styli were all about. It's
also possible within limits to modify the trajectory of the stylus so that
the intended stylus has the desired mechanical trajectory despite obvious
geometric limits. The real problem with mainstream vinyl was that it had to
be cut for the lowest common denominator playback system or else the
recording will sound extraordinarily crappy to way too many people, and have
a short life.

Not only
that, but this encoding allows the L+R amplitude to be
limited separately from L-R (L-R limiting would reduce
channel separation momentarily). Not only that, but the
signals could be equalized separately, so less bass is
sent to the L-R channel. This reduces the risk of the
cutter losing contact with the surface, avoiding
excessive distortion and skips on playback.

This was all done routinely, particularly in the latter days of vinyl, just
before the CD came out.

But as far as I am aware, there never has been any
separate processing of the L+R and L-R, only L and R
separately,

Then with all due respect, you weren't aware of the LP SOTA in the latter
days.

but using the same EQ and compressor
settings.

Ditto.

On rock recordings, the L-R was necessarily
minimised by mixing kick drums and sometimes bass to
centre, with the vocalist almost always dead centre.

Well that too. The advance of doing this is that the best people make better
artistic choices than electronics, particularly the limited electronics of
the late 1970s and early 80s.


Classical and Jazz tended to have more L-R, but as the
music wasn't so heavily compressed, the levels were lower
anyway.

Except that it isn't allowable to dig a hole or send the stylus into the air
or loop an adjacent track ever, even during crescendos. Just for fun they
adopted the convention of recording LPs from the outside edge in, so the
crescendos always ended up in the inner grooves where available dynamic
range was minimized.

It is essential that any L,R processing be done
with identical settings as otherwise the central image
will wander depending on frequency content and level.

As that doesn't happen anyway.

There were some mono/stereo compatible records (Synchro
Stereo was one I recall) which I understand mixed low
frequencies to mono and thus kept the L-R signal small
whilst still offering a noticeable stereo effect.

In fact that was happening to a certain degree very often on mainstream
releases with no special labeling.

I have
several such records of classical music, and they sound
adequate in stereo, but have sufficiently small L-R
levels that they can be played with a mono pickup without
damage.

Eventually a lot of pop came to be that way for any number of reasons, and
persists to this day even though most digital media has as much power
bandwidth as anything.

[Serge Auckland[_2_]]

"Arny Krueger" wrote in message
. ..
"Serge Auckland" wrote in
message
"Chronic Philharmonic" wrote
in message
...


"Earl Kiosterud" wrote in message
news
"Chronic Philharmonic" wrote
in message
...


"Randy Yates" wrote in message
...


snipped


Not only
that, but this encoding allows the L+R amplitude to be
limited separately from L-R (L-R limiting would reduce
channel separation momentarily). Not only that, but the
signals could be equalized separately, so less bass is
sent to the L-R channel. This reduces the risk of the
cutter losing contact with the surface, avoiding
excessive distortion and skips on playback.

This was all done routinely, particularly in the latter days of vinyl,
just before the CD came out.

But as far as I am aware, there never has been any
separate processing of the L+R and L-R, only L and R
separately,

Then with all due respect, you weren't aware of the LP SOTA in the latter
days.

Possibly not, so thanks for the update.



but using the same EQ and compressor
settings.

Ditto.

I don't understand this part: If the L&R have different compressor and EQ
settings, then the image will wander depending on level and frequencies. In
Broadcast at least, it's normal that the L&R settings are linked in a stereo
compressor/limiter and/or equaliser to avoid any image drift. Is this not
also done on LP mastering? If not, how is image drift avoided?



On rock recordings, the L-R was necessarily
minimised by mixing kick drums and sometimes bass to
centre, with the vocalist almost always dead centre.

Well that too. The advance of doing this is that the best people make
better artistic choices than electronics, particularly the limited
electronics of the late 1970s and early 80s.


Classical and Jazz tended to have more L-R, but as the
music wasn't so heavily compressed, the levels were lower
anyway.

Except that it isn't allowable to dig a hole or send the stylus into the
air or loop an adjacent track ever, even during crescendos. Just for fun
they adopted the convention of recording LPs from the outside edge in, so
the crescendos always ended up in the inner grooves where available
dynamic range was minimized.

Agreed. It always seemed odd to me that records played outside-in, when it
would be more logical to play inside-out.


It is essential that any L,R processing be done
with identical settings as otherwise the central image
will wander depending on frequency content and level.

As that doesn't happen anyway.

Why not? If a stereo signal has L&R processed independantly, then the image
will drift with level and frequency. That's why most stereo
compressor/limiters and EQs have a "link" button that provides the same
control signal to both channels.

S.


--
http://audiopages.googlepages.com

[Arny Krueger]

"Serge Auckland" wrote in
message
"Arny Krueger" wrote in message
. ..
"Serge Auckland" wrote in
message
"Chronic Philharmonic" wrote
in message
...


"Earl Kiosterud" wrote in message
news
"Chronic Philharmonic"
wrote in message
...


"Randy Yates" wrote in message
...


snipped


Not only
that, but this encoding allows the L+R amplitude to be
limited separately from L-R (L-R limiting would reduce
channel separation momentarily). Not only that, but the
signals could be equalized separately, so less bass is
sent to the L-R channel. This reduces the risk of the
cutter losing contact with the surface, avoiding
excessive distortion and skips on playback.

This was all done routinely, particularly in the latter
days of vinyl, just before the CD came out.

But as far as I am aware, there never has been any
separate processing of the L+R and L-R, only L and R
separately,

Then with all due respect, you weren't aware of the LP
SOTA in the latter days.

Possibly not, so thanks for the update.

but using the same EQ and compressor
settings.

Ditto.

I don't understand this part: If the L&R have different
compressor and EQ settings, then the image will wander
depending on level and frequencies.

Even if you the compressors are identical, there will still be wandering
channels.

For example, I compress both channels 2:1 above -10 dB.

One channel is 10 dB below the other, and they both steadily increase their
volume.

The channel that hits 10 dB first starts increasing more slowly and thus
starts sliding towards the center.

That's why they link control signals.

In Broadcast at
least, it's normal that the L&R settings are linked in a
stereo compressor/limiter and/or equaliser to avoid any
image drift.

Yes, using the same control signal on both compressors helps.

Is this not also done on LP mastering?

Yes, if L & R are compressed then the control signals are tied together and
any effects on imaging are second order.

If L-R is compressed, then both channels slide to the center.

If L+R is compressed, then diffuse sound becomes more diffuse.

Synchronizing their compresson without affecting imaging would be a neat
trick. Maybe that is one reason why highly-compressed recordings tend to
sound like mud.

[Richard Crowley]

"Serge Auckland" wrote ...
Agreed. It always seemed odd to me that records played outside-in, when it
would be more logical to play inside-out.

OTOH, note that for professional use (i.e. transcription, etc.) the
practice was often to record from the center out. Note further that
optical discs (CD, DVD play from the center to the outside edge.

[Mr.T]

"Serge Auckland" wrote in message
...
Agreed. It always seemed odd to me that records played outside-in, when it
would be more logical to play inside-out.

But the consequences of the stylus jumping the run out groove and falling
off the record onto the platter would be devastating. More likely, but less
expensive with the cheap players used by many of course, but still not
something the public would be happy with, especially since the limitations
(and therefore possible benefits) were unknown to the masses in any case.

MrT.

[Mr.T]

"Richard Crowley" wrote in message
...
"Serge Auckland" wrote ...
Agreed. It always seemed odd to me that records played outside-in, when
it
would be more logical to play inside-out.

OTOH, note that for professional use (i.e. transcription, etc.) the
practice was often to record from the center out. Note further that
optical discs (CD, DVD play from the center to the outside edge.

Which of course provides no audible benefit though. The reason is simply
that any size disk can be used without special size detection, since the TOC
always starts in the same place.

MrT.

[Dave Platt]

Agreed. It always seemed odd to me that records played outside-in, when it
would be more logical to play inside-out.

But the consequences of the stylus jumping the run out groove and falling
off the record onto the platter would be devastating.

My recollection is that records suffer from a higher level of
high-frequency tracing distortion when playing the inner grooves, due
in part to the fact that records are cut using constant angular
velocity, The wavelength of the signals (in the vinyl) becomes
smaller in the inner grooves, making it more difficult for the diamond
stylus to track the groove accurately. The fact that the stylus isn't
exactly tangent to the groove (at most points), makes life even more
complicated.

Mastering an LP involves a set of tradeoffs involving recording time,
level, and distortion. As the side length becomes greater, you need
correspondingly more spirals in the groove. You can go further
in towards the center, and suffer increasing levels of distortion in
the inner grooves.

Or, you can decrease the pitch (the distance between the grooves) so
that you don't go so far in. If you do this, you end up having to
reduce the audio level (turn down the volume) - otherwise, crosstalk
between adjacent grooves becomes more obvious (pre- and post-echo) and
in severe cases you end up accidentally cutting from one groove to the
next and ruining the master. Reducing the cutting amplitude will tend
to reduce playback tracing distortion, but it can result in the
record's surface noise being more obvious.

This all gets *really* complicated if the LP is being recorded "direct to
disk" rather than via a master tape - the mastering engineer has to
set the lathe's pitch adjustment "on the fly" based on his/her
knowledge of what the musicians are going to be playing in the next
few seconds.

--
Dave Platt AE6EO
Friends of Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!

[Richard Crowley]

"Dave Platt" wrote ...
This all gets *really* complicated if the LP is being recorded "direct to
disk" rather than via a master tape - the mastering engineer has to
set the lathe's pitch adjustment "on the fly" based on his/her
knowledge of what the musicians are going to be playing in the next
few seconds.

That's why we have rehearsals and run-throughs.