Discussion:
[Audacity-users] Alternatives to LAME mp3 encoder
jklarl
2017-03-05 18:50:46 UTC
Permalink
Hello Audacity people,



I'm looking for an alternative to the LAME mp3 encoder to use with Audacity if one exists. Which one do you recommend? Thanks in advance...



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stan
2017-03-05 19:55:50 UTC
Permalink
On Sun, 05 Mar 2017 14:50:46 -0400
Post by jklarl
I'm looking for an alternative to the LAME mp3 encoder to use with
Audacity if one exists. Which one do you recommend? Thanks in
advance...
I don't know if it will work with audacity, but fluendo sells codec
packages.

http://www.fluendo.com/en/products/enterprise/fluendo-codec-pack/

The mpg patents, including mp3, are almost all expired. There is
one left that will expire in early 2018. At that point, if mpg is
still relevant, other people will probably develop open source mp3
encoders and decoders.

I think the only open source options at this point are lame and libmad.

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Clyde Lyman
2017-03-05 21:07:00 UTC
Permalink
With the vast amounts of memory we have today, why bother with lossy files at all unless it's to keep continuity with other elements of a collection? I picked up a 400+ GB SDXC Class 10 card about a year ago What I'm more interested in a way to extract audio from a vid without having to go through hoops as with VLC and don't even talk about Freemake, they've become the Amazon of crapware - and worse; addons (pity; they were pretty good)


From: stan <***@q.com>
To: audacity-***@lists.sourceforge.net
Sent: Sunday, March 5, 2017 2:55 PM
Subject: Re: [Audacity-users] Alternatives to LAME mp3 encoder

On Sun, 05 Mar 2017 14:50:46 -0400
Post by jklarl
I'm looking for an alternative to the LAME mp3 encoder to use with
Audacity if one exists.  Which one do you recommend?  Thanks in
advance...
I don't know if it will work with audacity, but fluendo sells codec
packages.

http://www.fluendo.com/en/products/enterprise/fluendo-codec-pack/

The mpg patents, including mp3, are almost all expired.  There is
one left that will expire in early 2018.  At that point, if mpg is
still relevant, other people will probably develop open source mp3
encoders and decoders.

I think the only open source options at this point are lame and libmad.

------------------------------------------------------------------------------
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  Audacity, or Audacity >  About Audacity on a Mac computer) 

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J.B. Nicholson
2017-03-05 22:36:05 UTC
Permalink
Post by Clyde Lyman
With the vast amounts of memory we have today, why bother with lossy
files at all unless it's to keep continuity with other elements of a
collection?
I'm not sure what "continuity with other elements of a collection" means.

Lossy encoding is useful for:

- delivering multimedia over slow links or to low-end computers (mobile
phones, tablets, some laptops, toys). Even YouTube finds it useful to
encode a lot of audio as Opus and a lot of video with VP9 nowadays.

- embedded players that don't handle lossless encoding (some cars and some
portable playing devices don't support FLAC, WAV, or AIFF files, for example)

- maximizing use of (as you point out) inexpensive storage. If I can't tell
the difference between a high-quality Opus encoding over the same audio
ripped from an ordinary Red Book audio CD and encoded as a FLAC (which is
lossless), I might choose to keep the Opus file because it uses
significantly less space and works with every computer program I use for
playback. Thus my casual listening needs are met and I get to store a lot
more audio.


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Clyde Lyman
2017-03-06 16:09:48 UTC
Permalink
By "Continuity with the rest of the collection" I meant something like the following. I have 15\00+ mp3's, having started my collection in the days when an 8GB card was huge. It could make no sense to now put WAV's in that group. There are those who say that the difference between mp3 and loosless files is noticeable. so anything I add to that group will be an mp3
I can think of one more place where mp3 would be useful. The old GEM Genisys keyboard could play mp3's as samples for it synth. Now, an mp3 CD/DVD can hold way more samples than using the WAV/CDA files

Are there actually players that won't play at least FLAC (which is a compressed file if I understand correctly)

From: J.B. Nicholson <***@forestfield.org>
To: audacity-***@lists.sourceforge.net
Sent: Sunday, March 5, 2017 5:36 PM
Subject: [Audacity-users] Why bother with lossy encoding?
Post by Clyde Lyman
With the vast amounts of memory we have today, why bother with lossy
files at all unless it's to keep continuity with other elements of a
collection?
I'm not sure what "continuity with other elements of a collection" means.

Lossy encoding is useful for:

- delivering multimedia over slow links or to low-end computers (mobile
phones, tablets, some laptops, toys). Even YouTube finds it useful to
encode a lot of audio as Opus and a lot of video with VP9 nowadays.

- embedded players that don't handle lossless encoding (some cars and some
portable playing devices don't support FLAC, WAV, or AIFF files, for example)

- maximizing use of (as you point out) inexpensive storage. If I can't tell
the difference between a high-quality Opus encoding over the same audio
ripped from an ordinary Red Book audio CD and encoded as a FLAC (which is
lossless), I might choose to keep the Opus file because it uses
significantly less space and works with every computer program I use for
playback. Thus my casual listening needs are met and I get to store a lot
more audio.


------------------------------------------------------------------------------
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* Exactly what three digit version number of Audacity you are using (Help > About
  Audacity, or Audacity >  About Audacity on a Mac computer) 

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  connected to the computer?

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J.B. Nicholson
2017-03-07 00:24:21 UTC
Permalink
Post by Clyde Lyman
By "Continuity with the rest of the collection" I meant something like
the following. I have 15\00+ mp3's, having started my collection in the
days when an 8GB card was huge. It could make no sense to now put WAV's
in that group. There are those who say that the difference between mp3
and loosless files is noticeable. so anything I add to that group will
be an mp3
There's no need to continue to use MP3 just because you used MP3 in the
past. You can use any format your players can play. Whether one can hear
the artifacts introduced with lossy encoding has nothing to do with what
else is on the same volume, it has to do with the settings chosen at the
time of encoding. So it's entirely possible that one can hear artifacts in
an MP3 file compressed with low-quality settings.

I store my audio collection in formats that favor free software (free as in
freedom, not price) so I have FLAC (primarily), Opus, and older Vorbis
files all in the same collection organized into folders corresponding to my
organizational needs.

I don't re-encode from a lossy source to a lossy destination as that's
generally a good way to introduce audible distortions. Re-encoding either
from lossless to lossy, or from lossless to lossless both have practical
use (and the latter can be done losslessly).

Instead of WAV files I suggest using FLAC. WAV file metadata gets spotty
support but Ogg tags in native FLAC files are widely read correctly.
Post by Clyde Lyman
Are there actually players that won't play at least FLAC (which is a
compressed file if I understand correctly)
FLAC is losslessly compressed at multiple levels of compression -- the more
time one allows the encoder to compress the smaller the output file, but
all FLAC encoding levels are lossless.

There are audio players that won't play FLAC: Apple's portable players
(such as their iThings) won't play FLAC using the proprietary Apple
firmware. One could install Rockbox on some Apple iThings to gain FLAC
playback. But I don't see the point in acquiring an Apple iThing in the
first place as there are so many other devices which play FLAC files (and
have for years) that any device that doesn't play FLAC comes off to me as
uncompetitive and obsolete. And I have other reasons for not wanting to do
business with Apple which aren't related to audio file format support.

This is getting away from any issue dealing with Audacity per se so I'll
stop replying on this thread here.

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* Exactly what three digit version number of Audacity you are using (Help > About
Audacity, or Audacity > About Audacity on a Mac computer)

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Clyde Lyman
2017-03-07 09:57:30 UTC
Permalink
Any way to use Audacity as my defalt music player? I've looked but have not found one and it would be a great one. I'd like Shuffle


From: J.B. Nicholson <***@forestfield.org>
To: audacity-***@lists.sourceforge.net
Sent: Monday, March 6, 2017 7:24 PM
Subject: Re: [Audacity-users] Why bother with lossy encoding?
Post by Clyde Lyman
By "Continuity with the rest of the collection" I meant something like
the following. I have 15\00+ mp3's, having started my collection in the
days when an 8GB card was huge. It could make no sense to now put WAV's
in that group. There are those who say that the difference between mp3
and loosless files is noticeable. so anything I add to that group will
be an mp3
There's no need to continue to use MP3 just because you used MP3 in the
past. You can use any format your players can play. Whether one can hear
the artifacts introduced with lossy encoding has nothing to do with what
else is on the same volume, it has to do with the settings chosen at the
time of encoding. So it's entirely possible that one can hear artifacts in
an MP3 file compressed with low-quality settings.

I store my audio collection in formats that favor free software (free as in
freedom, not price) so I have FLAC (primarily), Opus, and older Vorbis
files all in the same collection organized into folders corresponding to my
organizational needs.

I don't re-encode from a lossy source to a lossy destination as that's
generally a good way to introduce audible distortions. Re-encoding either
from lossless to lossy, or from lossless to lossless both have practical
use (and the latter can be done losslessly).

Instead of WAV files I suggest using FLAC. WAV file metadata gets spotty
support but Ogg tags in native FLAC files are widely read correctly.
Post by Clyde Lyman
Are there actually players that won't play at least FLAC (which is a
compressed file if I understand correctly)
FLAC is losslessly compressed at multiple levels of compression -- the more
time one allows the encoder to compress the smaller the output file, but
all FLAC encoding levels are lossless.

There are audio players that won't play FLAC: Apple's portable players
(such as their iThings) won't play FLAC using the proprietary Apple
firmware. One could install Rockbox on some Apple iThings to gain FLAC
playback. But I don't see the point in acquiring an Apple iThing in the
first place as there are so many other devices which play FLAC files (and
have for years) that any device that doesn't play FLAC comes off to me as
uncompetitive and obsolete. And I have other reasons for not wanting to do
business with Apple which aren't related to audio file format support.

This is getting away from any issue dealing with Audacity per se so I'll
stop replying on this thread here.

------------------------------------------------------------------------------
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*********** ASKING FOR HELP *************

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* What operating system you are using (for example, Windows XP or Mac OS X 10.5.1) 

* Exactly what three digit version number of Audacity you are using (Help > About
  Audacity, or Audacity >  About Audacity on a Mac computer) 

* If this is a recording problem, what equipment you are recording with, and how is it
  connected to the computer?

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Steve the Fiddle
2017-03-07 10:27:37 UTC
Permalink
It would be less confusing if you start a new topic. The topic title
in this email thread is "Why bother with lossy encoding?"

Steve
Post by Clyde Lyman
Any way to use Audacity as my defalt music player? I've looked but have not
found one and it would be a great one. I'd like Shuffle
________________________________
Sent: Monday, March 6, 2017 7:24 PM
Subject: Re: [Audacity-users] Why bother with lossy encoding?
Post by Clyde Lyman
By "Continuity with the rest of the collection" I meant something like
the following. I have 15\00+ mp3's, having started my collection in the
days when an 8GB card was huge. It could make no sense to now put WAV's
in that group. There are those who say that the difference between mp3
and loosless files is noticeable. so anything I add to that group will
be an mp3
There's no need to continue to use MP3 just because you used MP3 in the
past. You can use any format your players can play. Whether one can hear
the artifacts introduced with lossy encoding has nothing to do with what
else is on the same volume, it has to do with the settings chosen at the
time of encoding. So it's entirely possible that one can hear artifacts in
an MP3 file compressed with low-quality settings.
I store my audio collection in formats that favor free software (free as in
freedom, not price) so I have FLAC (primarily), Opus, and older Vorbis
files all in the same collection organized into folders corresponding to my
organizational needs.
I don't re-encode from a lossy source to a lossy destination as that's
generally a good way to introduce audible distortions. Re-encoding either
from lossless to lossy, or from lossless to lossless both have practical
use (and the latter can be done losslessly).
Instead of WAV files I suggest using FLAC. WAV file metadata gets spotty
support but Ogg tags in native FLAC files are widely read correctly.
Post by Clyde Lyman
Are there actually players that won't play at least FLAC (which is a
compressed file if I understand correctly)
FLAC is losslessly compressed at multiple levels of compression -- the more
time one allows the encoder to compress the smaller the output file, but
all FLAC encoding levels are lossless.
There are audio players that won't play FLAC: Apple's portable players
(such as their iThings) won't play FLAC using the proprietary Apple
firmware. One could install Rockbox on some Apple iThings to gain FLAC
playback. But I don't see the point in acquiring an Apple iThing in the
first place as there are so many other devices which play FLAC files (and
have for years) that any device that doesn't play FLAC comes off to me as
uncompetitive and obsolete. And I have other reasons for not wanting to do
business with Apple which aren't related to audio file format support.
This is getting away from any issue dealing with Audacity per se so I'll
stop replying on this thread here.
------------------------------------------------------------------------------
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dictionary content that is easy and intuitive to access. Sign up for an
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http://sdm.link/oxford
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* What operating system you are using (for example, Windows XP or Mac OS X 10.5.1)
* Exactly what three digit version number of Audacity you are using (Help > About
Audacity, or Audacity > About Audacity on a Mac computer)
* If this is a recording problem, what equipment you are recording with, and how is it
connected to the computer?
https://lists.sourceforge.net/lists/listinfo/audacity-users
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Audacity, or Audacity > About Audacity on a Mac computer)
* If this is a recording problem, what equipment you are recording with, and how is it
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Gale Andrews
2017-03-07 17:22:57 UTC
Permalink
Post by Steve the Fiddle
It would be less confusing if you start a new topic. The topic title
in this email thread is "Why bother with lossy encoding?"
Steve
Post by Clyde Lyman
Any way to use Audacity as my defalt music player?
No, short of setting default file associations to Audacity for the audio file
types you are interested in.

Note that you get a "Save Changes?" prompt when you exit Audacity after
importing a file.

If you have questions, please start a new topic about if with a relevant
title (use "Compose" in your e-mail client).



Gale Andrews
Post by Steve the Fiddle
Post by Clyde Lyman
I've looked but have not
found one and it would be a great one. I'd like Shuffle
________________________________
Sent: Monday, March 6, 2017 7:24 PM
Subject: Re: [Audacity-users] Why bother with lossy encoding?
Post by Clyde Lyman
By "Continuity with the rest of the collection" I meant something like
the following. I have 15\00+ mp3's, having started my collection in the
days when an 8GB card was huge. It could make no sense to now put WAV's
in that group. There are those who say that the difference between mp3
and loosless files is noticeable. so anything I add to that group will
be an mp3
There's no need to continue to use MP3 just because you used MP3 in the
past. You can use any format your players can play. Whether one can hear
the artifacts introduced with lossy encoding has nothing to do with what
else is on the same volume, it has to do with the settings chosen at the
time of encoding. So it's entirely possible that one can hear artifacts in
an MP3 file compressed with low-quality settings.
I store my audio collection in formats that favor free software (free as in
freedom, not price) so I have FLAC (primarily), Opus, and older Vorbis
files all in the same collection organized into folders corresponding to my
organizational needs.
I don't re-encode from a lossy source to a lossy destination as that's
generally a good way to introduce audible distortions. Re-encoding either
from lossless to lossy, or from lossless to lossless both have practical
use (and the latter can be done losslessly).
Instead of WAV files I suggest using FLAC. WAV file metadata gets spotty
support but Ogg tags in native FLAC files are widely read correctly.
Post by Clyde Lyman
Are there actually players that won't play at least FLAC (which is a
compressed file if I understand correctly)
FLAC is losslessly compressed at multiple levels of compression -- the more
time one allows the encoder to compress the smaller the output file, but
all FLAC encoding levels are lossless.
There are audio players that won't play FLAC: Apple's portable players
(such as their iThings) won't play FLAC using the proprietary Apple
firmware. One could install Rockbox on some Apple iThings to gain FLAC
playback. But I don't see the point in acquiring an Apple iThing in the
first place as there are so many other devices which play FLAC files (and
have for years) that any device that doesn't play FLAC comes off to me as
uncompetitive and obsolete. And I have other reasons for not wanting to do
business with Apple which aren't related to audio file format support.
This is getting away from any issue dealing with Audacity per se so I'll
stop replying on this thread here.
------------------------------------------------------------------------------
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dictionary content that is easy and intuitive to access. Sign up for an
account today to start using our lexical data to power your apps and
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http://sdm.link/oxford
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* What operating system you are using (for example, Windows XP or Mac OS X 10.5.1)
* Exactly what three digit version number of Audacity you are using (Help > About
Audacity, or Audacity > About Audacity on a Mac computer)
* If this is a recording problem, what equipment you are recording with, and how is it
connected to the computer?
https://lists.sourceforge.net/lists/listinfo/audacity-users
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Audacity, or Audacity > About Audacity on a Mac computer)
* If this is a recording problem, what equipment you are recording with, and how is it
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L A Walsh
2017-03-09 04:07:54 UTC
Permalink
Post by Clyde Lyman
Any way to use Audacity as my defalt music player? I've looked but
have not found one and it would be a great one. I'd like Shuffle
----
Go get 'foobar2000'(free PC player) It's been out for over a decade,
has a bunch of plugins, handles most all audio formats.

Best of all, you can design your own GUI... it's a bit of a chore
to do, but once you have your own look with the panels and plugins you
want, you won't want to go back to generic, but you can use a preset until
you are ready for that.

As for a reason to use FLAC. I know I can go from FLAC to
any other format, losslessly. If I go for MP3, I can't later re-encode
to something else.

Besides, once you've got your home-collection in FLAC on your
hard disk, you won't want to take the time to re-encode into some other
format for each portable you have -- so I just got a good portable player
that can use a 128G memory card (chip?) (smaller than a dime).

If you haven't heard of it, Check out the Fiio line. If you just
want music, the Fiio X3, has a color display to display the album cover
and has up to a 24-bit, 192KHz output ability. You can also use it
as a high-rez output device for your PC if you are into that. I have
less than 50% usage with near 4000 songs -- all full size FLAC -- so
all i need to do is copy them from PC to the player or the chip directly.
Player is 6x10x1.5cm (divide by 2.54 for inches) and weighs about 155 grams
or 5.4oz.

I've left the thing turned on w/screen off (not playing), and grabbed it
a week later, and it was still 'on' with a fair amount of battery left.
They really did a good job -- and I've played it for days str8 w/no recharge
and battery didn't even get low -- really good w/power.

Fiio has other players if you want video and other stuff, they also
have portable amps (same size as the player, but able to power full size
studio headphones). Both charge via USB.



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Billy Geiger
2017-03-09 05:18:42 UTC
Permalink
"Player is 6x10x1.5cm (divide by 2.54 for inches)"....is that MULTIPLY
instead?

***@gmail.com

Billy Geiger
Winder, GA.
(40 miles east
of Atlanta)
Post by L A Walsh
Post by Clyde Lyman
Any way to use Audacity as my defalt music player? I've looked but
have not found one and it would be a great one. I'd like Shuffle
----
Go get 'foobar2000'(free PC player) It's been out for over a decade,
has a bunch of plugins, handles most all audio formats.
Best of all, you can design your own GUI... it's a bit of a chore
to do, but once you have your own look with the panels and plugins you
want, you won't want to go back to generic, but you can use a preset until
you are ready for that.
As for a reason to use FLAC. I know I can go from FLAC to
any other format, losslessly. If I go for MP3, I can't later re-encode
to something else.
Besides, once you've got your home-collection in FLAC on your
hard disk, you won't want to take the time to re-encode into some other
format for each portable you have -- so I just got a good portable player
that can use a 128G memory card (chip?) (smaller than a dime).
If you haven't heard of it, Check out the Fiio line. If you just
want music, the Fiio X3, has a color display to display the album cover
and has up to a 24-bit, 192KHz output ability. You can also use it
as a high-rez output device for your PC if you are into that. I have
less than 50% usage with near 4000 songs -- all full size FLAC -- so
all i need to do is copy them from PC to the player or the chip directly.
Player is 6x10x1.5cm (divide by 2.54 for inches) and weighs about 155 grams
or 5.4oz.
I've left the thing turned on w/screen off (not playing), and grabbed it
a week later, and it was still 'on' with a fair amount of battery left.
They really did a good job -- and I've played it for days str8 w/no recharge
and battery didn't even get low -- really good w/power.
Fiio has other players if you want video and other stuff, they also
have portable amps (same size as the player, but able to power full size
studio headphones). Both charge via USB.
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Post by Clyde Lyman
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L A Walsh
2017-03-09 16:38:57 UTC
Permalink
"Player is 6 x 10 x 1.5 cm (divide by 2.54 for inches)"....is that
MULTIPLY instead?
----
What do you get for the dimensions of the unit, in inches, if
you do that?

? ;-)





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Allistair Bywater
2017-03-09 17:40:59 UTC
Permalink
No, not multiply. You get 2.3" X 3.9" X 0.6" - roughly the size of my
mobile phone, only thicker. More or less the dimensions given on the
Fiio website - http://www.fiio.net/en/products/65

I get confused when I think about it....
Post by L A Walsh
"Player is 6 x 10 x 1.5 cm (divide by 2.54 for inches)"....is that
MULTIPLY instead?
----
What do you get for the dimensions of the unit, in inches, if
you do that?
? ;-)
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Billy Geiger
2017-03-09 20:25:16 UTC
Permalink
Ohhhhhhhhh....okay....had to put my rusty mind in gear....

Sorry....

***@gmail.com

Billy Geiger
Winder, GA.
(40 miles east
of Atlanta)
Post by Allistair Bywater
No, not multiply. You get 2.3" X 3.9" X 0.6" - roughly the size of my
mobile phone, only thicker. More or less the dimensions given on the
Fiio website - http://www.fiio.net/en/products/65
I get confused when I think about it....
Post by L A Walsh
"Player is 6 x 10 x 1.5 cm (divide by 2.54 for inches)"....is that
MULTIPLY instead?
----
What do you get for the dimensions of the unit, in inches, if
you do that?
? ;-)
------------------------------------------------------------
------------------
Post by L A Walsh
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dictionary content that is easy and intuitive to access. Sign up for an
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Post by L A Walsh
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X 10.5.1)
Post by L A Walsh
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(Help > About
Post by L A Walsh
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Post by L A Walsh
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Clyde Lyman
2017-03-10 04:35:55 UTC
Permalink
I use the GOM player, audio and video, they have a great reverb unit. Though I would like a parametric EQ with Q width control, sweepable frequency mids and a high-pass filter. Since I learned to use those things, I don't go near graphics


From: Billy Geiger <***@gmail.com>
To: Discussion list for Audacity users <audacity-***@lists.sourceforge.net>
Sent: Thursday, March 9, 2017 3:25 PM
Subject: Re: [Audacity-users] Why bother with lossy encoding?

Ohhhhhhhhh....okay....had to put my rusty mind in gear....
Sorry....
***@gmail.com

        Billy Geiger
        Winder, GA.
      (40 miles east
         of Atlanta)
On Thu, Mar 9, 2017 at 12:40 PM, Allistair Bywater <***@gmail.com> wrote:

No, not multiply. You get 2.3" X 3.9" X 0.6" - roughly the size of my
mobile phone, only thicker. More or less the dimensions given on the
Fiio website -  http://www.fiio.net/en/ products/65

I get confused when I think about it....
"Player is 6 x 10 x 1.5 cm (divide by 2.54 for inches)"....is that
MULTIPLY instead?
----
      What do you get for the dimensions of the unit, in inches, if
you do that?
? ;-)
------------------------------ ------------------------------ ------------------
Announcing the Oxford Dictionaries API! The API offers world-renowned
dictionary content that is easy and intuitive to access. Sign up for an
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projects. Get started today and enter our developer competition.
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Johnny Rosenberg
2017-03-09 20:59:04 UTC
Permalink
Seems like sound quality is very individual. Some people swallow
everything, some hear defects that don't exist, and I guess most people are
somewhere in between.
I grew up with vinyl and audio cassettes (those very common ones by that
time, 3.175 mm (⅛") wide tape, 47.625 mm/s (1⅞"/s) speed. I hated the
background noise, the ”wow & flutter", defects from those noise reduction
systems and so on. And I hated the vinyls as well, but for different
reasons. I was just annoyed by the sound.
Then I upgraded to a reel to reel tape recorder (a TEAC A-3440 4 channel
thing). Still a lot of tape hizz, but less wow & flutter and so on. Still
annoying, though, even at highest speed 380 mm/s (15"/s).

First time I had something that didn't annoy me was actually when I bought
my first CD player. I also upgraded my mixdown possibilities from a
cassette to a HiFi Video recorder. It sounded a lot better, but had some
annoyances as well. Only about a year later I bought a DAT recorder and was
very happy with it until it broke many years later.

These days I use the FLAC format for documenting my music recordings. I
find it easy to deal with. I can manage tags with my own bash-scripts and I
use that to enter everything I know about the songs I record. Title (all of
them if they are more than one, for instance songs that are translated from
another language, like ”Reflections In A Palace Lake (京郜慕情 – Kyoto Bojo)”
or ”Manchurian Beat (На СПпках МаМчжурОО)”, composers (even when I'm the
composer
), original artists, who plays what instrument, where we recorded
it and exactly when. I also use those tags to automatically upload them to
a link page of mine, for the other musicians to be able to download them
and give me feedback about the mixing and more. I know that most file
formats supports tags, but I find them very easy to work with in FLAC
files. And I like the Open Source idea in this case.

After listening to lossy formats I found that the result depends a lot on
the original sound. If the original sound is a poor noisy old recording
from the 60's or even earlier, not even 256 kb/s (mp3) is enough to avoid
audible artefacts. A good original however, may sound good enough at 160
kb/s (mp3). With Ogg/Vorbis files at least one step lower bitrate may be
ok. Hiss sounds seems to be very tricky to handle for those psycho acoustic
data reducing algorithms.

I guess I could write about this for several days, but I also have a life,
so



Kind regards

Johnny Rosenberg
Post by Allistair Bywater
No, not multiply. You get 2.3" X 3.9" X 0.6" - roughly the size of my
mobile phone, only thicker. More or less the dimensions given on the
Fiio website - http://www.fiio.net/en/products/65
I get confused when I think about it....
Post by L A Walsh
"Player is 6 x 10 x 1.5 cm (divide by 2.54 for inches)"....is that
MULTIPLY instead?
----
What do you get for the dimensions of the unit, in inches, if
you do that?
? ;-)
------------------------------------------------------------
------------------
Post by L A Walsh
Announcing the Oxford Dictionaries API! The API offers world-renowned
dictionary content that is easy and intuitive to access. Sign up for an
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Post by L A Walsh
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X 10.5.1)
Post by L A Walsh
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(Help > About
Post by L A Walsh
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Post by L A Walsh
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Stephane Ascoet
2017-03-14 10:59:31 UTC
Permalink
Post by Johnny Rosenberg
After listening to lossy formats I found that the result depends a lot on
the original sound. If the original sound is a poor noisy old recording
from the 60's or even earlier, not even 256 kb/s (mp3) is enough to avoid
audible artefacts. A good original however, may sound good enough at 160
kb/s (mp3). With Ogg/Vorbis files at least one step lower bitrate may be
ok. Hiss sounds seems to be very tricky to handle for those psycho acoustic
data reducing algorithms.
Yes it seems we are so different because for me the most important is
the "emotion" I get, so I like a good LP or tape but get bored by most
digital sound, except good DSD or PCM played with high-end audiophilic DAC.
--
Bien cordialement, Stephane Ascoet


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Roger
2017-03-14 17:11:50 UTC
Permalink
Post by Stephane Ascoet
Post by Johnny Rosenberg
After listening to lossy formats I found that the result depends a lot on
the original sound. If the original sound is a poor noisy old recording
from the 60's or even earlier, not even 256 kb/s (mp3) is enough to avoid
audible artefacts. A good original however, may sound good enough at 160
kb/s (mp3). With Ogg/Vorbis files at least one step lower bitrate may be
ok. Hiss sounds seems to be very tricky to handle for those psycho acoustic
data reducing algorithms.
Yes it seems we are so different because for me the most important is
the "emotion" I get, so I like a good LP or tape but get bored by most
digital sound, except good DSD or PCM played with high-end audiophilic DAC.
Ditto here, as I consider myself to have good ears for classical music.

Extracted all of my CD's into PCM WAV avoiding the lossy compression codecs,
but still noticing the metallic/tin type sound reminiscent of CD grade audio,
even with a 5.1 home stereo grade DAC. I seem to prefer the 24-bit 48kHz
sampling, but am no where near being rich to upgrade my entire music
collection. Can still recall the days of Tape and LP, where music was less
tinny sounding. But then that's also getting back to the days where tubes were
also more common.

I've also tried upsampling the CD audio to 24-bit 48kHz sampling, while using
an ALSA LP filter, helps but not significantly enough. Shrugs, maybe my ears
are just getting too old and too tired.
Billy Geiger
2017-03-14 17:21:24 UTC
Permalink
hhhmmmmmmmm....

Use Audacity to interject a few DBs of low-end bass....


***@gmail.com

Billy Geiger
Winder, GA.
(40 miles east
of Atlanta)
Post by Johnny Rosenberg
Post by Stephane Ascoet
Post by Johnny Rosenberg
After listening to lossy formats I found that the result depends a lot
on
Post by Stephane Ascoet
Post by Johnny Rosenberg
the original sound. If the original sound is a poor noisy old recording
from the 60's or even earlier, not even 256 kb/s (mp3) is enough to
avoid
Post by Stephane Ascoet
Post by Johnny Rosenberg
audible artefacts. A good original however, may sound good enough at 160
kb/s (mp3). With Ogg/Vorbis files at least one step lower bitrate may be
ok. Hiss sounds seems to be very tricky to handle for those psycho
acoustic
Post by Stephane Ascoet
Post by Johnny Rosenberg
data reducing algorithms.
Yes it seems we are so different because for me the most important is
the "emotion" I get, so I like a good LP or tape but get bored by most
digital sound, except good DSD or PCM played with high-end audiophilic
DAC.
Ditto here, as I consider myself to have good ears for classical music.
Extracted all of my CD's into PCM WAV avoiding the lossy compression codecs,
but still noticing the metallic/tin type sound reminiscent of CD grade audio,
even with a 5.1 home stereo grade DAC. I seem to prefer the 24-bit 48kHz
sampling, but am no where near being rich to upgrade my entire music
collection. Can still recall the days of Tape and LP, where music was less
tinny sounding. But then that's also getting back to the days where tubes were
also more common.
I've also tried upsampling the CD audio to 24-bit 48kHz sampling, while using
an ALSA LP filter, helps but not significantly enough. Shrugs, maybe my ears
are just getting too old and too tired.
------------------------------------------------------------
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Post by Stephane Ascoet
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Stephane Ascoet
2017-03-15 07:16:55 UTC
Permalink
Post by Roger
Ditto here, as I consider myself to have good ears for classical music.
Extracted all of my CD's into PCM WAV avoiding the lossy compression codecs,
but still noticing the metallic/tin type sound reminiscent of CD grade audio,
even with a 5.1 home stereo grade DAC. I seem to prefer the 24-bit 48kHz
sampling, but am no where near being rich to upgrade my entire music
collection. Can still recall the days of Tape and LP, where music was less
tinny sounding. But then that's also getting back to the days where tubes were
also more common.
I've also tried upsampling the CD audio to 24-bit 48kHz sampling, while using
an ALSA LP filter, helps but not significantly enough. Shrugs, maybe my ears
are just getting too old and too tired.
Hi, I've got a Sony 3000 ES expanding them to 96 khz at playing, results
are incredibly impressive with some CDs. With some others, that has been
very poorly engineered, there's no difference, the sound remains tiny.
--
Sincerely, Stephane Ascoet


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Steve the Fiddle
2017-03-15 10:30:58 UTC
Permalink
Just a quick technical point: Increasing the sample rate of
("upsampling") a recording, does not and can not improved the quality
of the encoded sound. Once the sound has been recorded in digital
format, that is "it", that data contains specific audio information,
and any subtle musical nuances that are missing are gone forever. It
is physically impossible to magically restore audio information that
is missing from the original.

The difference that sample rate makes, is that it limits the maximum
frequency that can be represented by the data. Uncompressed PCM data
has an absolute limit to the frequencies that it can represent. The
limit is half the sample rate (known as the "Nyquist frequency"). So
for audio CDs, (sample rate 44100 Hz), audio frequencies must be below
22050 Hz. The only difference that a higher sample rate makes is that
it could theoretically represent frequencies above 22050 Hz, but if
the original recording is on CD, then there are no audio frequencies
above 22050 Hz, and upsampling cannot change that, (other than by
adding distortion).

Steve

On 15 March 2017 at 07:16, Stephane Ascoet
Post by Stephane Ascoet
Post by Roger
Ditto here, as I consider myself to have good ears for classical music.
Extracted all of my CD's into PCM WAV avoiding the lossy compression codecs,
but still noticing the metallic/tin type sound reminiscent of CD grade audio,
even with a 5.1 home stereo grade DAC. I seem to prefer the 24-bit 48kHz
sampling, but am no where near being rich to upgrade my entire music
collection. Can still recall the days of Tape and LP, where music was less
tinny sounding. But then that's also getting back to the days where tubes were
also more common.
I've also tried upsampling the CD audio to 24-bit 48kHz sampling, while using
an ALSA LP filter, helps but not significantly enough. Shrugs, maybe my ears
are just getting too old and too tired.
Hi, I've got a Sony 3000 ES expanding them to 96 khz at playing, results
are incredibly impressive with some CDs. With some others, that has been
very poorly engineered, there's no difference, the sound remains tiny.
--
Sincerely, Stephane Ascoet
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Stephane Ascoet
2017-03-15 10:39:29 UTC
Permalink
Post by Steve the Fiddle
Just a quick technical point: Increasing the sample rate of
("upsampling") a recording, does not and can not improved the quality
of the encoded sound. Once the sound has been recorded in digital
format, that is "it", that data contains specific audio information,
and any subtle musical nuances that are missing are gone forever. It
is physically impossible to magically restore audio information that
is missing from the original.
The difference that sample rate makes, is that it limits the maximum
frequency that can be represented by the data. Uncompressed PCM data
has an absolute limit to the frequencies that it can represent. The
limit is half the sample rate (known as the "Nyquist frequency"). So
for audio CDs, (sample rate 44100 Hz), audio frequencies must be below
22050 Hz. The only difference that a higher sample rate makes is that
it could theoretically represent frequencies above 22050 Hz, but if
the original recording is on CD, then there are no audio frequencies
above 22050 Hz, and upsampling cannot change that, (other than by
adding distortion).
Steve
This is a technical point of view. But, as I already wrote it years ago
on this list, I can really ear a difference. The sound is more warm.
--
Sincerely, Stephane Ascoet


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Steve the Fiddle
2017-03-15 11:20:49 UTC
Permalink
On 15 March 2017 at 10:39, Stephane Ascoet
Post by Stephane Ascoet
Post by Steve the Fiddle
Just a quick technical point: Increasing the sample rate of
("upsampling") a recording, does not and can not improved the quality
of the encoded sound. Once the sound has been recorded in digital
format, that is "it", that data contains specific audio information,
and any subtle musical nuances that are missing are gone forever. It
is physically impossible to magically restore audio information that
is missing from the original.
The difference that sample rate makes, is that it limits the maximum
frequency that can be represented by the data. Uncompressed PCM data
has an absolute limit to the frequencies that it can represent. The
limit is half the sample rate (known as the "Nyquist frequency"). So
for audio CDs, (sample rate 44100 Hz), audio frequencies must be below
22050 Hz. The only difference that a higher sample rate makes is that
it could theoretically represent frequencies above 22050 Hz, but if
the original recording is on CD, then there are no audio frequencies
above 22050 Hz, and upsampling cannot change that, (other than by
adding distortion).
Steve
This is a technical point of view. But, as I already wrote it years ago
on this list, I can really ear a difference. The sound is more warm.
Yes, as I wrote, this is a technical point. I'm only talking about
actual physical facts, not "alternative" facts.

There are many reasons why upsampling may sound "different". The
additional sample rate conversion step could introduce losses for a
number of reasons, and that "difference" in sound may be preferable to
some listeners, just as some listeners prefer the "sizzle" sound of
low quality MP3s
(http://radar.oreilly.com/2009/03/the-sizzling-sound-of-music.html).
This should not, in my opinion, be confused with "sound quality" in
the sense of "accurate reproduction of sound".

There is no right or wrong about personal preference, Some people
prefer vinyl or even compact cassettes. Some prefer CDs and some
prefer MP3s. Some prefer Camembert and some prefer Stilton. Some
prefer Rock and some prefer Jazz. Thank heavens we're not all clones
;-)

Steve
Post by Stephane Ascoet
--
Sincerely, Stephane Ascoet
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stan
2017-03-15 18:12:39 UTC
Permalink
On Wed, 15 Mar 2017 11:20:49 +0000
Post by Steve the Fiddle
There is no right or wrong about personal preference, Some people
prefer vinyl or even compact cassettes. Some prefer CDs and some
prefer MP3s. Some prefer Camembert and some prefer Stilton. Some
prefer Rock and some prefer Jazz. Thank heavens we're not all clones
;-)
Steve
No, that can't be right! Everyone should recognize that *my*
preferences are the *right* preferences. ;-)

Like that old joke. The band comes marching down the street, and a
woman exclaims, "Look, everyone is out of step but my Johnny!"

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Clyde Lyman
2017-03-15 11:22:00 UTC
Permalink
Did analog recording or broadcast go up beyond 20,000 cps? My stereo's HF horns go from 2,000 to 25,000. On the other edn, I'm learning mixing & mastering and have learned that anything below 40 cps is just "rumble" and, musically, should be treated with the HP filter rolling 40 and below off


From: Steve the Fiddle <***@gmail.com>
To: Discussion list for Audacity users <audacity-***@lists.sourceforge.net>
Sent: Wednesday, March 15, 2017 6:30 AM
Subject: Re: [Audacity-users] Why bother with lossy encoding?

Just a quick technical point: Increasing the sample rate of
("upsampling") a recording, does not and can not improved the quality
of the encoded sound. Once the sound has been recorded in digital
format, that is "it", that data contains specific audio information,
and any subtle musical nuances that are missing are gone forever. It
is physically impossible to magically restore audio information that
is missing from the original.

The difference that sample rate makes, is that it limits the maximum
frequency that can be represented by the data. Uncompressed PCM data
has an absolute limit to the frequencies that it can represent. The
limit is half the sample rate (known as the "Nyquist frequency"). So
for audio CDs, (sample rate 44100 Hz), audio frequencies must be below
22050 Hz. The only difference that a higher sample rate makes is that
it could theoretically represent frequencies above 22050 Hz, but if
the original recording is on CD, then there are no audio frequencies
above 22050 Hz, and upsampling cannot change that, (other than by
adding distortion).

Steve

On 15 March 2017 at 07:16, Stephane Ascoet
Post by Stephane Ascoet
Post by Roger
Ditto here, as I consider myself to have good ears for classical music.
Extracted all of my CD's into PCM WAV avoiding the lossy compression codecs,
but still noticing the metallic/tin type sound reminiscent of CD grade audio,
even with a 5.1 home stereo grade DAC.  I seem to prefer the 24-bit 48kHz
sampling, but am no where near being rich to upgrade my entire music
collection.  Can still recall the days of Tape and LP, where music was less
tinny sounding.  But then that's also getting back to the days where tubes were
also more common.
I've also tried upsampling the CD audio to 24-bit 48kHz sampling, while using
an ALSA LP filter, helps but not significantly enough.  Shrugs, maybe my ears
are just getting too old and too tired.
Hi, I've got a Sony 3000 ES expanding them to 96 khz at playing, results
are incredibly impressive with some CDs. With some others, that has been
very poorly engineered, there's no difference, the sound remains tiny.
--
Sincerely, Stephane Ascoet
------------------------------------------------------------------------------
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    Audacity, or Audacity >  About Audacity on a Mac computer)
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  Audacity, or Audacity >  About Audacity on a Mac computer) 

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Steve the Fiddle
2017-03-15 11:44:16 UTC
Permalink
Post by Clyde Lyman
Did analog recording or broadcast go up beyond 20,000 cps? My stereo's HF
horns go from 2,000 to 25,000. On the other edn, I'm learning mixing &
mastering and have learned that anything below 40 cps is just "rumble" and,
musically, should be treated with the HP filter rolling 40 and below off
Usually "just rumble", though in some cases that "rumble" may be part
of the music. For example, the fundamental frequency of the lowest
string of an orchestral (4 string) double bass is 41 Hz (about 33 Hz
for a 5 string bass with a "low C"), and a large organ pipe can go as
staggeringly low as 8 Hz!
(https://en.wikipedia.org/wiki/Boardwalk_Hall_Auditorium_Organ#64-foot_Diaphone-Dulzian)

Steve
Post by Clyde Lyman
________________________________
To: Discussion list for Audacity users
Sent: Wednesday, March 15, 2017 6:30 AM
Subject: Re: [Audacity-users] Why bother with lossy encoding?
Just a quick technical point: Increasing the sample rate of
("upsampling") a recording, does not and can not improved the quality
of the encoded sound. Once the sound has been recorded in digital
format, that is "it", that data contains specific audio information,
and any subtle musical nuances that are missing are gone forever. It
is physically impossible to magically restore audio information that
is missing from the original.
The difference that sample rate makes, is that it limits the maximum
frequency that can be represented by the data. Uncompressed PCM data
has an absolute limit to the frequencies that it can represent. The
limit is half the sample rate (known as the "Nyquist frequency"). So
for audio CDs, (sample rate 44100 Hz), audio frequencies must be below
22050 Hz. The only difference that a higher sample rate makes is that
it could theoretically represent frequencies above 22050 Hz, but if
the original recording is on CD, then there are no audio frequencies
above 22050 Hz, and upsampling cannot change that, (other than by
adding distortion).
Steve
On 15 March 2017 at 07:16, Stephane Ascoet
Post by Stephane Ascoet
Post by Roger
Ditto here, as I consider myself to have good ears for classical music.
Extracted all of my CD's into PCM WAV avoiding the lossy compression codecs,
but still noticing the metallic/tin type sound reminiscent of CD grade audio,
even with a 5.1 home stereo grade DAC. I seem to prefer the 24-bit 48kHz
sampling, but am no where near being rich to upgrade my entire music
collection. Can still recall the days of Tape and LP, where music was less
tinny sounding. But then that's also getting back to the days where tubes were
also more common.
I've also tried upsampling the CD audio to 24-bit 48kHz sampling, while using
an ALSA LP filter, helps but not significantly enough. Shrugs, maybe my ears
are just getting too old and too tired.
Hi, I've got a Sony 3000 ES expanding them to 96 khz at playing, results
are incredibly impressive with some CDs. With some others, that has been
very poorly engineered, there's no difference, the sound remains tiny.
--
Sincerely, Stephane Ascoet
------------------------------------------------------------------------------
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Post by Roger
About
Audacity, or Audacity > About Audacity on a Mac computer)
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Roger
2017-03-15 14:59:24 UTC
Permalink
Post by Steve the Fiddle
Just a quick technical point: Increasing the sample rate of
("upsampling") a recording, does not and can not improved the quality
of the encoded sound. Once the sound has been recorded in digital
Exactly correct.

This is why I tried increasing sampling, while adding noise and augmenting
frequencies. (eg. ALSA LP effect filter.) In essence, in an attempt to trick
the ear. Gives a neat "neato" effect, but really nothing more.

Once the recording is at 16-bits 44100Hz or less, not much more one can do with
it. Some day, I'll have to buy some 24-bit 48000Hz Mozart, and perform a
side-by-side comparison. As well as compare tape/LP medias, but I just figured
at this point, I'll still be stuck with 16-bit 44100Hz recordings due to higher
costs of the 24-bit 48000Hz media and the preference for portable digital media
over non-portability and shortened life span of tape/LP medias.

Hence, just be content with what one has!
--
Roger
http://rogerx.freeshell.org/
Federico Miyara
2017-03-15 19:48:27 UTC
Permalink
Roger,

There is, at least theoretically, a way to increase the bandwidth above
the original Nyquist limit. That is something similar to a series of
psychoacoustic enhancer effects such as the aural exciter, which applies
low level distortion to the high frequency content of the signal, thus
creating spectral components well above the original spectral limit.
This signal is added to the original signal in small doses.

I haven't tested this idea. It works fine when the target frequencies
are in the range 10 kHz - 20 kHz, but I don't know how it would work
above 20 kHz, or even if it would make any audible difference. Most
people cannot hear anything above 18 kHz)

The addition of purely random high frequency noise may be a less
controlled version of this procedure.

Regards,

Federico
Post by Roger
Post by Steve the Fiddle
Just a quick technical point: Increasing the sample rate of
("upsampling") a recording, does not and can not improved the quality
of the encoded sound. Once the sound has been recorded in digital
Exactly correct.
This is why I tried increasing sampling, while adding noise and augmenting
frequencies. (eg. ALSA LP effect filter.) In essence, in an attempt to trick
the ear. Gives a neat "neato" effect, but really nothing more.
Once the recording is at 16-bits 44100Hz or less, not much more one can do with
it. Some day, I'll have to buy some 24-bit 48000Hz Mozart, and perform a
side-by-side comparison. As well as compare tape/LP medias, but I just figured
at this point, I'll still be stuck with 16-bit 44100Hz recordings due to higher
costs of the 24-bit 48000Hz media and the preference for portable digital media
over non-portability and shortened life span of tape/LP medias.
Hence, just be content with what one has!
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Federico Miyara
2017-03-15 18:26:31 UTC
Permalink
Steve,

However, there could be a difference, not by increasing the bandwidth of
the audio signal, but helping to make better profit of it. If we have
only 44100 samples a second to represent a 20 kHz audio bandwidth, the
smoothing filter requirements are stringent, which could lead to several
kinds of artifacts (phase delay, transients). If the same signal is
digitally resampled to 96 kHz with a high-quality codec, then the signal
reconstruction may improve since the filter artifacts can be reduced and
shifted towards higher frequencies.

Regards,

Federico
Post by Steve the Fiddle
Just a quick technical point: Increasing the sample rate of
("upsampling") a recording, does not and can not improved the quality
of the encoded sound. Once the sound has been recorded in digital
format, that is "it", that data contains specific audio information,
and any subtle musical nuances that are missing are gone forever. It
is physically impossible to magically restore audio information that
is missing from the original.
The difference that sample rate makes, is that it limits the maximum
frequency that can be represented by the data. Uncompressed PCM data
has an absolute limit to the frequencies that it can represent. The
limit is half the sample rate (known as the "Nyquist frequency"). So
for audio CDs, (sample rate 44100 Hz), audio frequencies must be below
22050 Hz. The only difference that a higher sample rate makes is that
it could theoretically represent frequencies above 22050 Hz, but if
the original recording is on CD, then there are no audio frequencies
above 22050 Hz, and upsampling cannot change that, (other than by
adding distortion).
Steve
On 15 March 2017 at 07:16, Stephane Ascoet
Post by Stephane Ascoet
Post by Roger
Ditto here, as I consider myself to have good ears for classical music.
Extracted all of my CD's into PCM WAV avoiding the lossy compression codecs,
but still noticing the metallic/tin type sound reminiscent of CD grade audio,
even with a 5.1 home stereo grade DAC. I seem to prefer the 24-bit 48kHz
sampling, but am no where near being rich to upgrade my entire music
collection. Can still recall the days of Tape and LP, where music was less
tinny sounding. But then that's also getting back to the days where tubes were
also more common.
I've also tried upsampling the CD audio to 24-bit 48kHz sampling, while using
an ALSA LP filter, helps but not significantly enough. Shrugs, maybe my ears
are just getting too old and too tired.
Hi, I've got a Sony 3000 ES expanding them to 96 khz at playing, results
are incredibly impressive with some CDs. With some others, that has been
very poorly engineered, there's no difference, the sound remains tiny.
--
Sincerely, Stephane Ascoet
------------------------------------------------------------------------------
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Audacity, or Audacity > About Audacity on a Mac computer)
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Bob Cavanaugh
2017-03-06 19:55:18 UTC
Permalink
From what I understand, wav files sound great on the air, so radio stations
use them. That being said, the processing used by the station, combined with
the nature of fm radio means that by the time I record it, I'm not going to
be able to tell the difference between a wav file and an mp3 file. As has
been pointed out before, why take up so much space on your hard drive with
wav files when the same amount of audio can be used in half the space? My
aircheck collection is about 3 gb, most of which is 128 kbps mp3 files.
Imagine how much that would be in wav files.

-----Original Message-----
From: J.B. Nicholson [mailto:***@forestfield.org]
Sent: Sunday, March 05, 2017 2:36 PM
To: audacity-***@lists.sourceforge.net
Subject: [Audacity-users] Why bother with lossy encoding?
With the vast amounts of memory we have today, why bother with lossy
files at all unless it's to keep continuity with other elements of a
collection?
I'm not sure what "continuity with other elements of a collection" means.

Lossy encoding is useful for:

- delivering multimedia over slow links or to low-end computers (mobile
phones, tablets, some laptops, toys). Even YouTube finds it useful to encode
a lot of audio as Opus and a lot of video with VP9 nowadays.

- embedded players that don't handle lossless encoding (some cars and some
portable playing devices don't support FLAC, WAV, or AIFF files, for
example)

- maximizing use of (as you point out) inexpensive storage. If I can't tell
the difference between a high-quality Opus encoding over the same audio
ripped from an ordinary Red Book audio CD and encoded as a FLAC (which is
lossless), I might choose to keep the Opus file because it uses
significantly less space and works with every computer program I use for
playback. Thus my casual listening needs are met and I get to store a lot
more audio.


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* What operating system you are using (for example, Windows XP or Mac OS X 10.5.1)

* Exactly what three digit version number of Audacity you are using (Help > About
Audacity, or Audacity > About Audacity on a Mac computer)

* If this is a recording problem, what equipment you are recording with, and how is it
connected to the computer?

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Clyde Lyman
2017-03-06 23:29:46 UTC
Permalink
I poked around and found that mp3's at 128kbps sound like radio and 160kbps sound like CD. Now if that's true then even in the 1970's you could get an astonishing amount of fidelity out of radios, especially classical stations. My stereo has a Realistic Accurion sub, some speakers that I picked up at a thrift store and Realistic horns that kick in at 2,000 and go up to 25,000, up until about 27 years ago, certain cash register aisles kicked out a sound that, in freq and volume went righ through me that nobody else coud detect. When I was audio tested in 1973, I was told I was losing some of my upper frequency hearing and when I got a bit nervous the guy said not to worry, ti was in the 48,000cps area and later on in the 1970's a friend of mine was told the same. If those bitrates are ture, I wonder what the 320 kbps is used for
Also, from what I understand, do the 96k and beyond sample rate and 32 bit depth recording settings really accomplish anythinhg? I have the previous incarnation of the M-Audio M-track plus and it maxes at 48k and 24 bits. Even then, the DAW has to step it down to 44k/16 bits which appears to be the same as 160kbps mp3
I mentioned that GEM Genesys as being able to use mp3 samples, also can most sfz players

I am looking for somethng. The full Cakewalk sfz player went on the free market at some time, However I can't find it on any page. Anyone know where I can get it. I have the jr. edition btu the full boat one had a superb GUI and would be worth the $60 if I could find it. it had infinitel layering


From: Bob Cavanaugh <***@comcast.net>
To: 'Discussion list for Audacity users' <audacity-***@lists.sourceforge.net>
Sent: Monday, March 6, 2017 2:55 PM
Subject: Re: [Audacity-users] Why bother with lossy encoding?
From what I understand, wav files sound great on the air, so radio stations
use them. That being said, the processing used by the station, combined with
the nature of fm radio means that by the time I record it, I'm not going to
be able to tell the difference between a wav file and an mp3 file. As has
been pointed out before, why take up so much space on your hard drive with
wav files when the same amount of audio can be used in half the space? My
aircheck collection is about 3 gb, most of which is 128 kbps mp3 files.
Imagine how much that would be in wav files.

-----Original Message-----
From: J.B. Nicholson [mailto:***@forestfield.org]
Sent: Sunday, March 05, 2017 2:36 PM
To: audacity-***@lists.sourceforge.net
Subject: [Audacity-users] Why bother with lossy encoding?
With the vast amounts of memory we have today, why bother with lossy
files at all unless it's to keep continuity with other elements of a
collection?
I'm not sure what "continuity with other elements of a collection" means.

Lossy encoding is useful for:

- delivering multimedia over slow links or to low-end computers (mobile
phones, tablets, some laptops, toys). Even YouTube finds it useful to encode
a lot of audio as Opus and a lot of video with VP9 nowadays.

- embedded players that don't handle lossless encoding (some cars and some
portable playing devices don't support FLAC, WAV, or AIFF files, for
example)

- maximizing use of (as you point out) inexpensive storage. If I can't tell
the difference between a high-quality Opus encoding over the same audio
ripped from an ordinary Red Book audio CD and encoded as a FLAC (which is
lossless), I might choose to keep the Opus file because it uses
significantly less space and works with every computer program I use for
playback. Thus my casual listening needs are met and I get to store a lot
more audio.


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*********** ASKING FOR HELP *************

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so we can help you properly:

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10.5.1) 

* Exactly what three digit version number of Audacity you are using (Help >
About
  Audacity, or Audacity >  About Audacity on a Mac computer) 

* If this is a recording problem, what equipment you are recording with, and
how is it
  connected to the computer?

Mailing list: Audacity-***@lists.sourceforge.net
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  Audacity, or Audacity >  About Audacity on a Mac computer) 

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Roger
2017-03-07 00:54:47 UTC
Permalink
Post by Clyde Lyman
depth recording settings really accomplish anythinhg? I have the previous
incarnation of the M-Audio M-track plus and it maxes at 48k and 24 bits.
Even then, the DAW has to step it down to 44k/16 bits which appears to be
the same as 160kbps mp3
If you're afraid of audio quality loss, typically 24 bit 48000hz sample rate
are considered slightly beyond the human capable range of hearing. (... if not
well beyond depending on who you talk to.) CD quality, 16 bit at 44100hz does
sound "tinny" (or metallic tin) to me. Given that CD quality (eg. 16 bit at
44100hz) standard was set quite some time ago for CD media storage limitations,
was likely adequate for that time period of the 1990's.

If I do any audio recording, I will always use 24 bit at 48000hz sample rate,
and save as PCM WAV.

Either way; the audio streams likely still experience some loss (at 24 bit @
48khz) while the stream is being digitized, as the digitization process is not
absolutely perfect... yet.

For me, I keep everything in PCM WAV format, and use OGG Vorbis for portable
devices. PCM WAV is likely the most compatabile format versus FLAC. On the
flip, file metadata is not really a standard for PCM WAV files, unlike somebody
just mentioned as FLAC having no problems with additional file metadata.
(Similar to PPM/PBM/TIFF versus PNG image formats, etc...) File metadata
embedded into PCM WAV files can be nightmarish or cause significant
incompatabilities, but not also to further mention, UTF-8/UTF-16 can also cause
further anomalies instead of using just ASCII for filenames. (eg. Hardware
vendor FAT16/FAT32)

I'm seeing more and more audio streams either resort to either "MPEG 1.0 L III
cbr96 44100 j-s" if they desire license free audio streaming, or migrating to
AAC/FAAC (MPEG-4) if they can handle paying a few licenses. MP3 from what I
understand has several patents, inhibiting adaptation by hardware companies and
stream providers, etc.
--
Roger
http://rogerx.freeshell.org/
Bob Cavanaugh
2017-03-07 01:22:18 UTC
Permalink
I've seen many streams in AAC or MP3. To me, anything above 128 kbps has no
quality difference. That is, if I didn't know the file format ahead of time,
I don't think I could tell the difference between an mp3 recorded at 44.1
kHZ 128 kbps from a wav recorded at 44.1 kHZ 16 bit. Not being well-versed
in audio processing setups at radio stations, I can't tell you what a
difference wav vs. MP3 makes. I'm told however, that KBPA 103.5 Bob FM in
Austin, TX is one of the best sounding stations in that market, and there
library is all wav files, I was told it was 11 tb! That being said, if I
were to record them off the air, I don't think I'd need to go way overboard
with a huge file format, especially since if I ever find myself in that
area, the radio I'd bring only has 2 gb of memory, not enough to record as
many stations as I'd want with wav format.

-----Original Message-----
From: Roger [mailto:***@gmail.com]
Sent: Monday, March 06, 2017 4:55 PM
To: audacity-***@lists.sourceforge.net
Subject: Re: [Audacity-users] Why bother with lossy encoding?
Post by Clyde Lyman
depth recording settings really accomplish anythinhg? I have the previous
incarnation of the M-Audio M-track plus and it maxes at 48k and 24 bits.
Even then, the DAW has to step it down to 44k/16 bits which appears to be
the same as 160kbps mp3
If you're afraid of audio quality loss, typically 24 bit 48000hz sample rate
are considered slightly beyond the human capable range of hearing. (... if
not
well beyond depending on who you talk to.) CD quality, 16 bit at 44100hz
does
sound "tinny" (or metallic tin) to me. Given that CD quality (eg. 16 bit at
44100hz) standard was set quite some time ago for CD media storage
limitations, was likely adequate for that time period of the 1990's.

If I do any audio recording, I will always use 24 bit at 48000hz sample
rate, and save as PCM WAV.

Either way; the audio streams likely still experience some loss (at 24 bit @
48khz) while the stream is being digitized, as the digitization process is
not absolutely perfect... yet.

For me, I keep everything in PCM WAV format, and use OGG Vorbis for portable
devices. PCM WAV is likely the most compatabile format versus FLAC. On the
flip, file metadata is not really a standard for PCM WAV files, unlike
somebody just mentioned as FLAC having no problems with additional file
metadata.
(Similar to PPM/PBM/TIFF versus PNG image formats, etc...) File metadata
embedded into PCM WAV files can be nightmarish or cause significant
incompatabilities, but not also to further mention, UTF-8/UTF-16 can also
cause further anomalies instead of using just ASCII for filenames. (eg.
Hardware vendor FAT16/FAT32)

I'm seeing more and more audio streams either resort to either "MPEG 1.0 L
III
cbr96 44100 j-s" if they desire license free audio streaming, or migrating
to AAC/FAAC (MPEG-4) if they can handle paying a few licenses. MP3 from
what I understand has several patents, inhibiting adaptation by hardware
companies and stream providers, etc.
--
Roger
http://rogerx.freeshell.org/


------------------------------------------------------------------------------
Announcing the Oxford Dictionaries API! The API offers world-renowned
dictionary content that is easy and intuitive to access. Sign up for an
account today to start using our lexical data to power your apps and
projects. Get started today and enter our developer competition.
http://sdm.link/oxford
*********** ASKING FOR HELP *************

When asking for help on this list, please include the following information, so we can
help you properly:

* What operating system you are using (for example, Windows XP or Mac OS X 10.5.1)

* Exactly what three digit version number of Audacity you are using (Help > About
Audacity, or Audacity > About Audacity on a Mac computer)

* If this is a recording problem, what equipment you are recording with, and how is it
connected to the computer?

Mailing list: Audacity-***@lists.sourceforge.net
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Clyde Lyman
2017-03-07 09:48:58 UTC
Permalink
I looked at the Audacity plugins and I was just blown away, I'd like to use some of those .ny's with other DAWs, like Tracktion 5, REAPER, Kryistal  or LMMS, is that possible


From: Bob Cavanaugh <***@comcast.net>
To: 'Discussion list for Audacity users' <audacity-***@lists.sourceforge.net>
Sent: Monday, March 6, 2017 8:22 PM
Subject: Re: [Audacity-users] Why bother with lossy encoding?

I've seen many streams in AAC or MP3. To me, anything above 128 kbps has no
quality difference. That is, if I didn't know the file format ahead of time,
I don't think I could tell the difference between an mp3 recorded at 44.1
kHZ 128 kbps from a wav recorded at 44.1 kHZ 16 bit. Not being well-versed
in audio processing setups at radio stations, I can't tell you what a
difference wav vs. MP3 makes. I'm told however, that KBPA 103.5 Bob FM in
Austin, TX is one of the best sounding stations in that market, and there
library is all wav files, I was told it was 11 tb! That being said, if I
were to record them off the air, I don't think I'd need to go way overboard
with a huge file format, especially since if I ever find myself in that
area, the radio I'd bring only has 2 gb of memory, not enough to record as
many stations as I'd want with wav format.

-----Original Message-----
From: Roger [mailto:***@gmail.com]
Sent: Monday, March 06, 2017 4:55 PM
To: audacity-***@lists.sourceforge.net
Subject: Re: [Audacity-users] Why bother with lossy encoding?
  depth recording settings really accomplish anythinhg? I have the
previous
  incarnation of the M-Audio M-track plus and it maxes at 48k and 24 bits.
  Even then, the DAW has to step it down to 44k/16 bits which appears to
be
  the same as 160kbps mp3
If you're afraid of audio quality loss, typically 24 bit 48000hz sample rate
are considered slightly beyond the human capable range of hearing. (... if
not
well beyond depending on who you talk to.)  CD quality, 16 bit at 44100hz
does
sound "tinny" (or metallic tin) to me.  Given that CD quality (eg. 16 bit at
44100hz) standard was set quite some time ago for CD media storage
limitations, was likely adequate for that time period of the 1990's.

If I do any audio recording, I will always use 24 bit at 48000hz sample
rate, and save as PCM WAV.

Either way; the audio streams likely still experience some loss (at 24 bit @
48khz) while the stream is being digitized, as the digitization process is
not absolutely perfect...  yet.

For me, I keep everything in PCM WAV format, and use OGG Vorbis for portable
devices.  PCM WAV is likely the most compatabile format versus FLAC.  On the
flip, file metadata is not really a standard for PCM WAV files, unlike
somebody just mentioned as FLAC having no problems with additional file
metadata. 
(Similar to PPM/PBM/TIFF versus PNG image formats, etc...)  File metadata
embedded into PCM WAV files can be nightmarish or cause significant
incompatabilities, but not also to further mention, UTF-8/UTF-16 can also
cause further anomalies instead of using just ASCII for filenames.  (eg.
Hardware vendor FAT16/FAT32)

I'm seeing more and more audio streams either resort to either "MPEG 1.0 L
III
cbr96 44100 j-s" if they desire license free audio streaming, or migrating
to AAC/FAAC (MPEG-4) if they can handle paying a few licenses.  MP3 from
what I understand has several patents, inhibiting adaptation by hardware
companies and stream providers, etc.
--
Roger
http://rogerx.freeshell.org/


------------------------------------------------------------------------------
Announcing the Oxford Dictionaries API! The API offers world-renowned
dictionary content that is easy and intuitive to access. Sign up for an
account today to start using our lexical data to power your apps and
projects. Get started today and enter our developer competition.
http://sdm.link/oxford
*********** ASKING FOR HELP *************

When asking for help on this list, please include the following information, so we can
help you properly:

* What operating system you are using (for example, Windows XP or Mac OS X 10.5.1) 

* Exactly what three digit version number of Audacity you are using (Help > About
  Audacity, or Audacity >  About Audacity on a Mac computer) 

* If this is a recording problem, what equipment you are recording with, and how is it
  connected to the computer?

Mailing list: Audacity-***@lists.sourceforge.net
To UNSUBSCRIBE, use the form at the bottom of this web page:
https://lists.sourceforge.net/lists/listinfo/audacity-users
Steve the Fiddle
2017-03-07 10:03:21 UTC
Permalink
".NY" files are "Nyquist Plug-ins" and are unique to Audacity.

Steve
Post by Clyde Lyman
I looked at the Audacity plugins and I was just blown away, I'd like to use
some of those .ny's with other DAWs, like Tracktion 5, REAPER, Kryistal or
LMMS, is that possible
________________________________
To: 'Discussion list for Audacity users'
Sent: Monday, March 6, 2017 8:22 PM
Subject: Re: [Audacity-users] Why bother with lossy encoding?
I've seen many streams in AAC or MP3. To me, anything above 128 kbps has no
quality difference. That is, if I didn't know the file format ahead of time,
I don't think I could tell the difference between an mp3 recorded at 44.1
kHZ 128 kbps from a wav recorded at 44.1 kHZ 16 bit. Not being well-versed
in audio processing setups at radio stations, I can't tell you what a
difference wav vs. MP3 makes. I'm told however, that KBPA 103.5 Bob FM in
Austin, TX is one of the best sounding stations in that market, and there
library is all wav files, I was told it was 11 tb! That being said, if I
were to record them off the air, I don't think I'd need to go way overboard
with a huge file format, especially since if I ever find myself in that
area, the radio I'd bring only has 2 gb of memory, not enough to record as
many stations as I'd want with wav format.
-----Original Message-----
Sent: Monday, March 06, 2017 4:55 PM
Subject: Re: [Audacity-users] Why bother with lossy encoding?
Post by Clyde Lyman
depth recording settings really accomplish anythinhg? I have the
previous
Post by Clyde Lyman
incarnation of the M-Audio M-track plus and it maxes at 48k and 24 bits.
Even then, the DAW has to step it down to 44k/16 bits which appears to
be
Post by Clyde Lyman
the same as 160kbps mp3
If you're afraid of audio quality loss, typically 24 bit 48000hz sample rate
are considered slightly beyond the human capable range of hearing. (... if not
well beyond depending on who you talk to.) CD quality, 16 bit at 44100hz
does
sound "tinny" (or metallic tin) to me. Given that CD quality (eg. 16 bit at
44100hz) standard was set quite some time ago for CD media storage
limitations, was likely adequate for that time period of the 1990's.
If I do any audio recording, I will always use 24 bit at 48000hz sample
rate, and save as PCM WAV.
48khz) while the stream is being digitized, as the digitization process is
not absolutely perfect... yet.
For me, I keep everything in PCM WAV format, and use OGG Vorbis for portable
devices. PCM WAV is likely the most compatabile format versus FLAC. On the
flip, file metadata is not really a standard for PCM WAV files, unlike
somebody just mentioned as FLAC having no problems with additional file
metadata.
(Similar to PPM/PBM/TIFF versus PNG image formats, etc...) File metadata
embedded into PCM WAV files can be nightmarish or cause significant
incompatabilities, but not also to further mention, UTF-8/UTF-16 can also
cause further anomalies instead of using just ASCII for filenames. (eg.
Hardware vendor FAT16/FAT32)
I'm seeing more and more audio streams either resort to either "MPEG 1.0 L III
cbr96 44100 j-s" if they desire license free audio streaming, or migrating
to AAC/FAAC (MPEG-4) if they can handle paying a few licenses. MP3 from
what I understand has several patents, inhibiting adaptation by hardware
companies and stream providers, etc.
--
Roger
http://rogerx.freeshell.org/
------------------------------------------------------------------------------
Announcing the Oxford Dictionaries API! The API offers world-renowned
dictionary content that is easy and intuitive to access. Sign up for an
account today to start using our lexical data to power your apps and
projects. Get started today and enter our developer competition.
http://sdm.link/oxford
*********** ASKING FOR HELP *************
When asking for help on this list, please include the following information, so we can
* What operating system you are using (for example, Windows XP or Mac OS X 10.5.1)
* Exactly what three digit version number of Audacity you are using (Help > About
Audacity, or Audacity > About Audacity on a Mac computer)
* If this is a recording problem, what equipment you are recording with, and how is it
connected to the computer?
https://lists.sourceforge.net/lists/listinfo/audacity-users
------------------------------------------------------------------------------
Announcing the Oxford Dictionaries API! The API offers world-renowned
dictionary content that is easy and intuitive to access. Sign up for an
account today to start using our lexical data to power your apps and
projects. Get started today and enter our developer competition.
http://sdm.link/oxford
*********** ASKING FOR HELP *************
When asking for help on this list, please include the following information, so we can
* What operating system you are using (for example, Windows XP or Mac OS X 10.5.1)
* Exactly what three digit version number of Audacity you are using (Help > About
Audacity, or Audacity > About Audacity on a Mac computer)
* If this is a recording problem, what equipment you are recording with, and how is it
connected to the computer?
https://lists.sourceforge.net/lists/listinfo/audacity-users
------------------------------------------------------------------------------
Announcing the Oxford Dictionaries API! The API offers world-renowned
dictionary content that is easy and intuitive to access. Sign up for an
account today to start using our lexical data to power your apps and
projects. Get started today and enter our developer competition.
http://sdm.link/oxford
*********** ASKING FOR HELP *************

When asking for help on this list, please include the following information, so we can
help you properly:

* What operating system you are using (for example, Windows XP or Mac OS X 10.5.1)

* Exactly what three digit version number of Audacity you are using (Help > About
Audacity, or Audacity > About Audacity on a Mac computer)

* If this is a recording problem, what equipment you are recording with, and how is it
connected to the computer?

Mailing list: Audacity-***@lists.sourceforge.net
To UNSUBSCRIBE, use the form at the bottom of this web page:
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Clyde Lyman
2017-03-07 12:53:36 UTC
Permalink
Thankx I just started poking around in the effects. the Reverb is awesome; looks better than the one in GOM, and the only problem I have with the EQ is setting th Q if it can be done, some of the Draw images have a tendency to shelf rather than bell curve, unless there's something I'm missing. Drawing the EQ is pretty advanced stuff. I'll also look for things like channel strip (EQ, Compression, saturation and reverb) or maybe you can set things up so that you can automate one. I have the distinct impression there is much more to Audacity than most people know


From: Steve the Fiddle <***@gmail.com>
To: Discussion list for Audacity users <audacity-***@lists.sourceforge.net>
Sent: Tuesday, March 7, 2017 5:03 AM
Subject: Re: [Audacity-users] Why bother with lossy encoding?

".NY" files are "Nyquist Plug-ins" and are unique to Audacity.

Steve
Post by Clyde Lyman
I looked at the Audacity plugins and I was just blown away, I'd like to use
some of those .ny's with other DAWs, like Tracktion 5, REAPER, Kryistal  or
LMMS, is that possible
________________________________
To: 'Discussion list for Audacity users'
Sent: Monday, March 6, 2017 8:22 PM
Subject: Re: [Audacity-users] Why bother with lossy encoding?
I've seen many streams in AAC or MP3. To me, anything above 128 kbps has no
quality difference. That is, if I didn't know the file format ahead of time,
I don't think I could tell the difference between an mp3 recorded at 44.1
kHZ 128 kbps from a wav recorded at 44.1 kHZ 16 bit. Not being well-versed
in audio processing setups at radio stations, I can't tell you what a
difference wav vs. MP3 makes. I'm told however, that KBPA 103.5 Bob FM in
Austin, TX is one of the best sounding stations in that market, and there
library is all wav files, I was told it was 11 tb! That being said, if I
were to record them off the air, I don't think I'd need to go way overboard
with a huge file format, especially since if I ever find myself in that
area, the radio I'd bring only has 2 gb of memory, not enough to record as
many stations as I'd want with wav format.
-----Original Message-----
Sent: Monday, March 06, 2017 4:55 PM
Subject: Re: [Audacity-users] Why bother with lossy encoding?
  depth recording settings really accomplish anythinhg? I have the
previous
  incarnation of the M-Audio M-track plus and it maxes at 48k and 24 bits.
  Even then, the DAW has to step it down to 44k/16 bits which appears to
be
  the same as 160kbps mp3
If you're afraid of audio quality loss, typically 24 bit 48000hz sample rate
are considered slightly beyond the human capable range of hearing. (... if not
well beyond depending on who you talk to.)  CD quality, 16 bit at 44100hz
does
sound "tinny" (or metallic tin) to me.  Given that CD quality (eg. 16 bit at
44100hz) standard was set quite some time ago for CD media storage
limitations, was likely adequate for that time period of the 1990's.
If I do any audio recording, I will always use 24 bit at 48000hz sample
rate, and save as PCM WAV.
48khz) while the stream is being digitized, as the digitization process is
not absolutely perfect...  yet.
For me, I keep everything in PCM WAV format, and use OGG Vorbis for portable
devices.  PCM WAV is likely the most compatabile format versus FLAC.  On the
flip, file metadata is not really a standard for PCM WAV files, unlike
somebody just mentioned as FLAC having no problems with additional file
metadata.
(Similar to PPM/PBM/TIFF versus PNG image formats, etc...)  File metadata
embedded into PCM WAV files can be nightmarish or cause significant
incompatabilities, but not also to further mention, UTF-8/UTF-16 can also
cause further anomalies instead of using just ASCII for filenames.  (eg.
Hardware vendor FAT16/FAT32)
I'm seeing more and more audio streams either resort to either "MPEG 1.0 L III
cbr96 44100 j-s" if they desire license free audio streaming, or migrating
to AAC/FAAC (MPEG-4) if they can handle paying a few licenses.  MP3 from
what I understand has several patents, inhibiting adaptation by hardware
companies and stream providers, etc.
--
Roger
http://rogerx.freeshell.org/
------------------------------------------------------------------------------
Announcing the Oxford Dictionaries API! The API offers world-renowned
dictionary content that is easy and intuitive to access. Sign up for an
account today to start using our lexical data to power your apps and
projects. Get started today and enter our developer competition.
http://sdm.link/oxford
*********** ASKING FOR HELP *************
When asking for help on this list, please include the following information, so we can
* What operating system you are using (for example, Windows XP or Mac OS X 10.5.1)
* Exactly what three digit version number of Audacity you are using (Help > About
  Audacity, or Audacity >  About Audacity on a Mac computer)
* If this is a recording problem, what equipment you are recording with, and how is it
  connected to the computer?
https://lists.sourceforge.net/lists/listinfo/audacity-users
------------------------------------------------------------------------------
Announcing the Oxford Dictionaries API! The API offers world-renowned
dictionary content that is easy and intuitive to access. Sign up for an
account today to start using our lexical data to power your apps and
projects. Get started today and enter our developer competition.
http://sdm.link/oxford
*********** ASKING FOR HELP *************
When asking for help on this list, please include the following information, so we can
* What operating system you are using (for example, Windows XP or Mac OS X 10.5.1)
* Exactly what three digit version number of Audacity you are using (Help > About
    Audacity, or Audacity >  About Audacity on a Mac computer)
* If this is a recording problem, what equipment you are recording with, and how is it
    connected to the computer?
https://lists.sourceforge.net/lists/listinfo/audacity-users
------------------------------------------------------------------------------
Announcing the Oxford Dictionaries API! The API offers world-renowned
dictionary content that is easy and intuitive to access. Sign up for an
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Bob Cavanaugh
2017-03-07 01:34:26 UTC
Permalink
I’m not sure, though I’m not sure I would have used the wording you used, the 320 kbps bitrate could be used for anything you want. As far as comparison though, I’ve heard fm sounds like 112 kbps, or 96 depending on who you ask. The CC Witness I just got this last Christmas has its default bitrate set at 96 kbps for fm, I had to bump that up to 128 because I wasn’t getting the dynamic range I wanted out of fm recordings. As I said in the previous message though, I wouldn’t go above 128.



From: Clyde Lyman [mailto:***@verizon.net]
Sent: Monday, March 06, 2017 3:30 PM
To: Discussion list for Audacity users <audacity-***@lists.sourceforge.net>
Subject: Re: [Audacity-users] Why bother with lossy encoding?



I poked around and found that mp3's at 128kbps sound like radio and 160kbps sound like CD. Now if that's true then even in the 1970's you could get an astonishing amount of fidelity out of radios, especially classical stations. My stereo has a Realistic Accurion sub, some speakers that I picked up at a thrift store and Realistic horns that kick in at 2,000 and go up to 25,000, up until about 27 years ago, certain cash register aisles kicked out a sound that, in freq and volume went righ through me that nobody else coud detect. When I was audio tested in 1973, I was told I was losing some of my upper frequency hearing and when I got a bit nervous the guy said not to worry, ti was in the 48,000cps area and later on in the 1970's a friend of mine was told the same. If those bitrates are ture, I wonder what the 320 kbps is used for



Also, from what I understand, do the 96k and beyond sample rate and 32 bit depth recording settings really accomplish anythinhg? I have the previous incarnation of the M-Audio M-track plus and it maxes at 48k and 24 bits. Even then, the DAW has to step it down to 44k/16 bits which appears to be the same as 160kbps mp3



I mentioned that GEM Genesys as being able to use mp3 samples, also can most sfz players

I am looking for somethng. The full Cakewalk sfz player went on the free market at some time, However I can't find it on any page. Anyone know where I can get it. I have the jr. edition btu the full boat one had a superb GUI and would be worth the $60 if I could find it. it had infinitel layering



_____

From: Bob Cavanaugh <***@comcast.net <mailto:***@comcast.net> >
To: 'Discussion list for Audacity users' <audacity-***@lists.sourceforge.net <mailto:audacity-***@lists.sourceforge.net> >
Sent: Monday, March 6, 2017 2:55 PM
Subject: Re: [Audacity-users] Why bother with lossy encoding?
From what I understand, wav files sound great on the air, so radio stations
use them. That being said, the processing used by the station, combined with
the nature of fm radio means that by the time I record it, I'm not going to
be able to tell the difference between a wav file and an mp3 file. As has
been pointed out before, why take up so much space on your hard drive with
wav files when the same amount of audio can be used in half the space? My
aircheck collection is about 3 gb, most of which is 128 kbps mp3 files.
Imagine how much that would be in wav files.

-----Original Message-----
From: J.B. Nicholson [mailto:***@forestfield.org <mailto:***@forestfield.org> ]
Sent: Sunday, March 05, 2017 2:36 PM
To: audacity-***@lists.sourceforge.net <mailto:audacity-***@lists.sourceforge.net>
Subject: [Audacity-users] Why bother with lossy encoding?
With the vast amounts of memory we have today, why bother with lossy
files at all unless it's to keep continuity with other elements of a
collection?
I'm not sure what "continuity with other elements of a collection" means.

Lossy encoding is useful for:

- delivering multimedia over slow links or to low-end computers (mobile
phones, tablets, some laptops, toys). Even YouTube finds it useful to encode
a lot of audio as Opus and a lot of video with VP9 nowadays.

- embedded players that don't handle lossless encoding (some cars and some
portable playing devices don't support FLAC, WAV, or AIFF files, for
example)

- maximizing use of (as you point out) inexpensive storage. If I can't tell
the difference between a high-quality Opus encoding over the same audio
ripped from an ordinary Red Book audio CD and encoded as a FLAC (which is
lossless), I might choose to keep the Opus file because it uses
significantly less space and works with every computer program I use for
playback. Thus my casual listening needs are met and I get to store a lot
more audio.


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Stephane Ascoet
2017-03-08 09:53:14 UTC
Permalink
Post by Clyde Lyman
I poked around and found that mp3's at 128kbps sound like radio and 160kbps sound like CD. Now if that's true then even in the 1970's you could get an astonishing amount of fidelity out
Hi, I've always been bothered by such comparaisons since they are not
accurate, only very approximative. Sound annoyances between each of the
above cases are very different. In a general thought, analog and digital
can't be compared easily because they are so much different
--
Bien cordialement, Stephane Ascoet


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Roger
2017-03-08 19:13:14 UTC
Permalink
Post by Stephane Ascoet
Post by Clyde Lyman
I poked around and found that mp3's at 128kbps sound like radio and 160kbps sound like CD. Now if that's true then even in the 1970's you could get an astonishing amount of fidelity out
Hi, I've always been bothered by such comparaisons since they are not
accurate, only very approximative. Sound annoyances between each of the
above cases are very different. In a general thought, analog and digital
can't be compared easily because they are so much different
Sound is subjective to each individual's ear's abilities and each individual's
mental state. Testing or benchmarking is usually limited to such tasks as,
either one can ear the sound or they cannot hear the sound.

In other words, since most everybody does drugs, CD quality sound or music
maybe extremely adequate or any anomalies are not noticeable at all. Those
sober will likely or may hear a difference. (eg. Some noises bother some
people more than others.)

The best method I've found when comparing audio quality, is usually a utilizing
a side-by-side comparison, using a small 1-2 second clipping and repeatedly
replaying until your brain memorizes each audio stream's anomalies, if there
are any anomalies at all.

Not only this, but the two streams you mentioned are using two different base
formats, not easily replicated over and over as per the previous mentioned
testing method as you already mentioned.

--
Roger
http://rogerx.freeshell.org/

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Gale Andrews
2017-03-06 16:40:36 UTC
Permalink
Libmad is a decoder, not encoder.

Audacity can only use LAME, unless you export using (external program),
which means typing out commands.


Gale
Post by stan
On Sun, 05 Mar 2017 14:50:46 -0400
Post by jklarl
I'm looking for an alternative to the LAME mp3 encoder to use with
Audacity if one exists. Which one do you recommend? Thanks in
advance...
I don't know if it will work with audacity, but fluendo sells codec
packages.
http://www.fluendo.com/en/products/enterprise/fluendo-codec-pack/
The mpg patents, including mp3, are almost all expired. There is
one left that will expire in early 2018. At that point, if mpg is
still relevant, other people will probably develop open source mp3
encoders and decoders.
I think the only open source options at this point are lame and libmad.
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