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Can a .m4a audio file be converted into a .mp3 one losslessly?



 
 
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  #16  
Old April 3rd 19, 04:50 AM posted to microsoft.public.windowsxp.general
Nil[_5_]
external usenet poster
 
Posts: 1,731
Default Can a .m4a audio file be converted into a .mp3 one losslessly?

On 02 Apr 2019, wrote in
microsoft.public.windowsxp.general:

On Tue, 02 Apr 2019 14:51:56 -0400, Nil
wrote:
If the original M4A is high enough quality and you convert to a
high- resolution you might not notice the difference. If you're
going to be editing the files you should consider converting the
M4A to WAV format, doing all your editing in that domain, then
converting them to MP3 as the very last stage.


I am not sure how that changes anything. It is still the
equivalent to an analog "copy of a copy".


It makes a difference if you do multiple edits and save your work in
the interim. If you were working on the original compressed file you'd
be recompressing it each save. Doing all those edits on the WAV file
avoids that.

If you were to do all your edits on the compressed file in one session
with the final format conversion at the end, with no intermediate
saves, it probably wouldn't make any difference.
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  #18  
Old April 3rd 19, 06:19 AM posted to microsoft.public.windowsxp.general
Bill in Co[_3_]
external usenet poster
 
Posts: 303
Default Can a .m4a audio file be converted into a .mp3 one losslessly?

wrote:
On Tue, 02 Apr 2019 23:50:07 -0400, Nil
wrote:

On 02 Apr 2019,
wrote in
microsoft.public.windowsxp.general:

On Tue, 02 Apr 2019 14:51:56 -0400, Nil
wrote:
If the original M4A is high enough quality and you convert to a
high- resolution you might not notice the difference. If you're
going to be editing the files you should consider converting the
M4A to WAV format, doing all your editing in that domain, then
converting them to MP3 as the very last stage.

I am not sure how that changes anything. It is still the
equivalent to an analog "copy of a copy".


It makes a difference if you do multiple edits and save your work in
the interim. If you were working on the original compressed file you'd
be recompressing it each save. Doing all those edits on the WAV file
avoids that.

I think most serious editors do that because it takes a whole lot
longer to "load" than simply to copy the MP3. (Based on Sound Forge)
If you save as WAV it is almost instant. Any other format chugs along.


+1 on that, and on SoundForge, probably the best editor around, especially
if you add some DX plugins to extend its capability. The only thing
missing is spectral editing, but there are other programs for that. I think
one of the best was the old Adobe Audition 1.5, before Adobe ruined it. :-).

I always marvel at just how smooth and truly excellent the editing
experience is when using SoundForge (in comparison to so many other audio
editors). But as for a good cheap alternative, there is always Goldwave.


  #19  
Old April 3rd 19, 02:36 PM posted to microsoft.public.windowsxp.general,alt.windows7.general
J. P. Gilliver (John)[_4_]
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Posts: 2,679
Default Can a .m4a audio file be converted into a .mp3 one losslessly?

In message , Bill in Co
writes:
J. P. Gilliver (John) wrote:

[]
I've just tried. Unfortunately, neither my old Goldwave nor my old
WinAmp did, so I doubt my standalone player would. I just tried with
another .aac file I have, and old GoldWave still didn't like it, but old
WinAmp did, so there are obviously more than one kind of .aac file.


It may have something to do with the licensing, OR you didn't install a
converter pack for Goldwave, if that option exists. With aac we're getting


I suspect it doesn't, and that modern GoldWave versions do take m4a. (I
have 5.58.)

into Apple territory here, and Apple likes to lock things down, as I recall.
But there are programs out there. I don't think it's due to the kind of aac
file, it's due to whether or not the converter has been added to the program
or not, and the licensing stuff


Not entirely. I did try somebody's suggestion of just renaming one of
the .m4a files to .aac, and neither my GoldWave nor my WinAmp 5.61 would
load it; whereas a file I already had, which was called *.aac when I got
it, _did_ load into the WinAmp.
[]
Yes, I was going to bring up the analogy with JPEGs, but you've already done
it. The purists out there would insist on never using a JPEG for anything.
:-) The analogy would be to only use WAV files for everything, with FLAC
being perhaps a close second. I can't imagine living with that limitation,
in terms of disk space, etc. However, if I were an audio engineer working
in the studio, that would of course be different.


Of course. (Might even look at more than 16 bit depth and/or more than
44100 sample rate.)

Since I use a portable
tiny mp3 player, disk space is still important to me. Come to think of it,
even on the HD too, as I have tons of audio files collected and stored there
over all the years, and there wouldn't be enough space for WAV files.

[]
the differences are typically inaudible, if you do it right. Just be
sure to keep the bitrate the same, or preferably higher, in the
conversions.


Hmm, bitrate: I'd always (I think) go via WAV, with no compression (i.
e. raw).


When you convert one format to another, I think the intermediary step is WAV
anyways, so I'm not sure you gain anything by selecting WAV (since I believe
that step happens anyways as it's being converted from one format to the


Well, I think the intermediate step is raw audio data; WAV is a
representation of raw audio data, just with a header saying (among other
things) how it's packed.
[]
That's where we differ - I rarely use as _high_ a rate as 128kbps for
.mp3! 96kbps is about the highest I use normally. For 22050 Hz (which I
use if there's no content above 11 kHz) I don't think I've used over 48k
for mono, 64k for stereo.


Well, I'm talking about good quality music files here on a decent sound
system,, not just stuff off the radio. For the latter, I agree.


I'm perhaps more cloth-eared (-:. Although even on spectrograms (i. e.
showing frequencies beyond my limit - which I think is somewhere between
10 and 15 kHz now), I'm surprised how much even relatively modern
material (up to '70s or even later) has no content above quite a low
frequency. I tend to only change sample rates in binary steps (I feel
that's an extra distortion otherwise), so I don't _often_ use 22050 (as
that _can't_ contain anything above 11025), though it does sometimes
happen. (For a lot of material from 78s, I find I _can_ use 11025 - i.
e. there's nothing much above 5500. And _some_ more modern material.)
That's sample rate rather than bit rate, of course, but the bit rate
needed nearly always halves when the sample rate does, as you'd expect.

A _lot_ of material I have recently extracted from videos (even if
losslessly extracted) has already been brick-wall filtered - a lawnmower
cut on the spectrogram; 15 kHz seems a popular one, with 16 and 13 not
uncommon.
[]
files, or those who insist that anything that has been digitized sounds
inferior to, say, vinyl records,


Oh, purlease (-:.


A system where even high-quality pickups have distortion percentages in
the whole numbers. Not to mention the recent trendies who buy these
cheap record players that have started to appear; I think some of them
are mid-quality, but I gather some have playing weights of several
grammes. Which I wouldn't let at the records which now (a very limited
selection, and virtually all old stuff) seem to cost about 20 pounds.

And I didn't even get into the so called purists who insist that vacuum tube
amplifiers have much less distortion than any transistor amplifiers (but
which was admitedly true back in the 1960's and early 1970s, as I do
recall).


There are two aspects to that. Before limiting, I think most valve (UK
for "toob") amplifiers _do_ have higher distortion, but it's a form that
the human ear finds less objectionable; and at limiting, solid state
circuitry tends to clip limit, which is very audible, whereas valve ones
limit more gracefully. But, when used in mid-range, I'd defy most people
to tell the difference (between well-designed circuitry of both types).

simply because it was digitized, or even that those
expensive gold plated monster cables are required for the purest sound to
the loudspeakers. :-)


Gold plated _connectors_ have some use _if_ you're replugging often -

[]
different; it's basically the corrosion problem again.


Yup, only for the corrosion thing. But some think the gold makes better
sound. :-)


The placebo effect is prevalent in most things, and if they can afford
it and it makes them happy ... (-:

So my advice is let go of the mantra about lossless conversions, and
just do it and enjoy the music. But recode it at a minimum of 128
kbps, using Joint Stereo mode. I also prefer to use a fix bitrate, but
there are those who like to use a variable bitrate, which is another
story.

I've nothing against VBR (as long as the converter uses some sort of
look-ahead), but the old LAME encoder I use (and/or the GoldWave I use
it from) doesn't/don't have VBR.


VBR files can (admitely rarely) be problematic for some audio utilities or
some audio programs, which is why I stick with the simple CBR format, which
works flawlessly on everything, without any hidden surprises along the way.


Well, as we've already discussed, if I was going to do further
processing, I wouldn't use any (lossy) compression at all. (Probably not
lossless [is that what FLAC is?] either, for the extra hassle.)
[]
I have a similar philosophy about making backup images for my HD. I
*always* do a complete image backup - no incrementals. I just don't trust
it. Everything is contained in only one file - there is only one file to go
wrong, if it goes wrong, and there are no other dependencies. :-)

I image my C:-plus-hidden, and sync. my D: (with one of three complete
copies used in rotation), so am similar to you - no incrementals or
differentials.

--
J. P. Gilliver. UMRA: 1960/1985 MB++G()AL-IS-Ch++(p)Ar@T+H+Sh0!:`)DNAf

Beauty is in the eye of the beer holder...
  #21  
Old April 3rd 19, 05:24 PM posted to microsoft.public.windowsxp.general,alt.windows7.general
Bill in Co[_3_]
external usenet poster
 
Posts: 303
Default Can a .m4a audio file be converted into a .mp3 one losslessly?

J. P. Gilliver (John) wrote:
In message , Bill in Co
writes:
J. P. Gilliver (John) wrote:

[]
I've just tried. Unfortunately, neither my old Goldwave nor my old
WinAmp did, so I doubt my standalone player would. I just tried with
another .aac file I have, and old GoldWave still didn't like it, but old
WinAmp did, so there are obviously more than one kind of .aac file.


It may have something to do with the licensing, OR you didn't install a
converter pack for Goldwave, if that option exists. With aac we're
getting


I suspect it doesn't, and that modern GoldWave versions do take m4a. (I
have 5.58.)


Oh, ok. I also recall they still had a version that worked with Windows XP,
but that the later versions wouldn't. (I used it for awhile too, as I have a
collection of audio editors), Another good and affordable audio editor, but
perhaps less well known, is Acon Acoustica, from Germany (was better for me
than Goldwave in terms of audio restoration):

https://acondigital.com/products/aco...-audio-editor/

into Apple territory here, and Apple likes to lock things down, as I
recall. But there are programs out there. I don't think it's due to the
kind of aac file, it's due to whether or not the converter has been
added to the program or not, and the licensing stuff


Not entirely. I did try somebody's suggestion of just renaming one of
the .m4a files to .aac, and neither my GoldWave nor my WinAmp 5.61 would
load it; whereas a file I already had, which was called *.aac when I got
it, _did_ load into the WinAmp.


Oh, ok. I don't know the answer then.

[]
Yes, I was going to bring up the analogy with JPEGs, but you've already
done it. The purists out there would insist on never using a JPEG for
anything. :-) The analogy would be to only use WAV files for
everything, with FLAC being perhaps a close second. I can't imagine
living with that limitation, in terms of disk space, etc. However, if I
were an audio engineer working in the studio, that would of course be
different.


Of course. (Might even look at more than 16 bit depth and/or more than
44100 sample rate.)

Since I use a portable
tiny mp3 player, disk space is still important to me. Come to think of
it, even on the HD too, as I have tons of audio files collected and
stored there over all the years, and there wouldn't be enough space for
WAV files.

[]
the differences are typically inaudible, if you do it right. Just be
sure to keep the bitrate the same, or preferably higher, in the
conversions.

Hmm, bitrate: I'd always (I think) go via WAV, with no compression (i.
e. raw).


When you convert one format to another, I think the intermediary step is
WAV anyways, so I'm not sure you gain anything by selecting WAV (since I
believe that step happens anyways as it's being converted from one
format to the


Well, I think the intermediate step is raw audio data; WAV is a
representation of raw audio data, just with a header saying (among other
things) how it's packed.
[]


I stand corrected. :-)

That's where we differ - I rarely use as _high_ a rate as 128kbps for
.mp3! 96kbps is about the highest I use normally. For 22050 Hz (which I
use if there's no content above 11 kHz) I don't think I've used over 48k
for mono, 64k for stereo.


Well, I'm talking about good quality music files here on a decent sound
system,, not just stuff off the radio. For the latter, I agree.


I'm perhaps more cloth-eared (-:. Although even on spectrograms (i. e.
showing frequencies beyond my limit - which I think is somewhere between
10 and 15 kHz now), I'm surprised how much even relatively modern
material (up to '70s or even later) has no content above quite a low
frequency. I tend to only change sample rates in binary steps (I feel
that's an extra distortion otherwise), so I don't _often_ use 22050 (as
that _can't_ contain anything above 11025), though it does sometimes
happen. (For a lot of material from 78s, I find I _can_ use 11025 - i.
e. there's nothing much above 5500. And _some_ more modern material.)
That's sample rate rather than bit rate, of course, but the bit rate
needed nearly always halves when the sample rate does, as you'd expect.

A _lot_ of material I have recently extracted from videos (even if
losslessly extracted) has already been brick-wall filtered - a lawnmower
cut on the spectrogram; 15 kHz seems a popular one, with 16 and 13 not
uncommon.
[]


I think we're both old enough that presbycusis has set in. When I was
younger I think I could hear up to around 18 kHz or so. But not any more.
So having a brick wall cutoff at 15 kHz on some material seems ok.

files, or those who insist that anything that has been digitized sounds
inferior to, say, vinyl records,

Oh, purlease (-:.


A system where even high-quality pickups have distortion percentages in
the whole numbers. Not to mention the recent trendies who buy these
cheap record players that have started to appear; I think some of them
are mid-quality, but I gather some have playing weights of several
grammes. Which I wouldn't let at the records which now (a very limited
selection, and virtually all old stuff) seem to cost about 20 pounds.


Maybe that's how they get away with it at that price - using several grammes
for the pickup. Not on my records, you don't!!

And I didn't even get into the so called purists who insist that vacuum
tube amplifiers have much less distortion than any transistor amplifiers
(but which was admitedly true back in the 1960's and early 1970s, as I do
recall).


There are two aspects to that. Before limiting, I think most valve (UK
for "toob") amplifiers _do_ have higher distortion, but it's a form that
the human ear finds less objectionable; and at limiting, solid state
circuitry tends to clip limit, which is very audible, whereas valve ones
limit more gracefully. But, when used in mid-range, I'd defy most people
to tell the difference (between well-designed circuitry of both types).


And the FET amps might be closer to the valve amps in terms of that type of
distortion, near clipping But I wasn't thinking of the clipping aspects so
much, but rather the THD and IM distortion at normal audio levels, way
before clipping, which I'm rather sure are much lower for transistor amps
than tube (valve) amps. But maybe all the whooplah of prefering valve amps
was primarily over the clipping aspects, as you just implied. My feeling is
if you're in that situation, you simply need a higher power amp, so you
never drive it into clipping in the first place. :-)

simply because it was digitized, or even that those
expensive gold plated monster cables are required for the purest sound
to the loudspeakers. :-)

Gold plated _connectors_ have some use _if_ you're replugging often -

[]
different; it's basically the corrosion problem again.


Yup, only for the corrosion thing. But some think the gold makes better
sound. :-)


The placebo effect is prevalent in most things, and if they can afford
it and it makes them happy ... (-:


I guess that's one way to look at it. :-)

So my advice is let go of the mantra about lossless conversions, and
just do it and enjoy the music. But recode it at a minimum of 128
kbps, using Joint Stereo mode. I also prefer to use a fix bitrate, but
there are those who like to use a variable bitrate, which is another
story.
I've nothing against VBR (as long as the converter uses some sort of
look-ahead), but the old LAME encoder I use (and/or the GoldWave I use
it from) doesn't/don't have VBR.


VBR files can (admitely rarely) be problematic for some audio utilities
or some audio programs, which is why I stick with the simple CBR format,
which works flawlessly on everything, without any hidden surprises along
the way.


Well, as we've already discussed, if I was going to do further
processing, I wouldn't use any (lossy) compression at all. (Probably not
lossless [is that what FLAC is?] either, for the extra hassle.)
[]


FLAC is indeed a lossless format, but I've never used it. For one thing, it
only reduces the filesize by about half anyways, so why bother. And it's
not universally supported. Wiki has an interesting article on it, and gives
at least some hints as to how it does its thing.

I have a similar philosophy about making backup images for my HD. I
*always* do a complete image backup - no incrementals. I just don't
trust it. Everything is contained in only one file - there is only one
file to go wrong, if it goes wrong, and there are no other dependencies.
:-)

I image my C:-plus-hidden, and sync. my D: (with one of three complete
copies used in rotation), so am similar to you - no incrementals or
differentials.


Yup. I'm just getting more conservative in my approaches as I get older.
:-)


  #24  
Old April 4th 19, 06:19 AM posted to microsoft.public.windowsxp.general
Paul[_32_]
external usenet poster
 
Posts: 11,873
Default Can a .m4a audio file be converted into a .mp3 one losslessly?

wrote:
On Wed, 03 Apr 2019 21:14:23 -0400, Nil
wrote:

On 03 Apr 2019,
wrote in
microsoft.public.windowsxp.general:

I think most serious editors do that because it takes a whole lot
longer to "load" than simply to copy the MP3. (Based on Sound
Forge) If you save as WAV it is almost instant. Any other format
chugs along.

If you're saying what I think you're saying, I don't think that's true.
When you open a compressed audio file, most editors will load it into
RAM until you save it. You can edit the RAM copy as much as you want
with no data loss, but as soon as you save it to a compressed format
file the data loss occurs. If you save multiple times, multiple cycles
of loss occur.

As evidence, note that after a save, if you quit the application, it
quits right away. There is no unexpected cleanup. At least, that's have
every audio editor I've used works.


I am saying I think the work file in RAM is WAV. Then you have the
option of how to save it, knowing another compression will lose more
than the first.
These days with storage as cheap as it is, I would seriously think of
only using WAV if I was an audiophool.


https://manual.audacityteam.org/man/digital_audio.html

"Audio CDs and most computer audio file formats use 16-bit integers.

Audacity uses 32-bit floating-point samples internally and, if
required, converts the sample bit depth when the final mix is exported
"

https://upload.wikimedia.org/wikiped...xample.svg.png

The mantissa is 23 bits, but there is an implicit "1" to the
left of the number. So it's a 24 bit number, in "integer terms".
The mantissa representation ranges between 1.000 and 1.999 kind of thing,
and the .000 and .999 part is the 23 bit mantissa. The exponent is adjusted
(normalized), to keep the rest of the number within that range.

The representation is a function of the kind of math
that the designers think will be applied to the samples.

You could, for example, run an FFT on the samples and make
a spectral plot, without needing to "launder" the samples
in advance. With that internal representation, they're
"ready to go".

Paul
  #25  
Old April 4th 19, 06:21 AM posted to microsoft.public.windowsxp.general,alt.windows7.general
Nil[_5_]
external usenet poster
 
Posts: 1,731
Default Can a .m4a audio file be converted into a .mp3 one losslessly?

On 03 Apr 2019, "Bill in Co" surly_curmudgeon@earthlink wrote in
microsoft.public.windowsxp.general:

FLAC is indeed a lossless format, but I've never used it. For one
thing, it only reduces the filesize by about half anyways, so why
bother. And it's not universally supported.


FLAC natively supports information tags that can be read by any player
that plays FLACS, which is almost every one that's not named Apple. I
see no advantage to using uncompressed WAV files for anything except
editing.

If FLAC isn't "universally" supported, it's within inches of it.
Windows 10 even supports it as an OS feature.
  #26  
Old April 4th 19, 01:39 PM posted to microsoft.public.windowsxp.general,alt.windows7.general
J. P. Gilliver (John)[_4_]
external usenet poster
 
Posts: 2,679
Default Can a .m4a audio file be converted into a .mp3 one losslessly?

In message , Nil
writes:
On 03 Apr 2019, "Bill in Co" surly_curmudgeon@earthlink wrote in
microsoft.public.windowsxp.general:

FLAC is indeed a lossless format, but I've never used it. For one
thing, it only reduces the filesize by about half anyways, so why
bother. And it's not universally supported.


FLAC natively supports information tags that can be read by any player
that plays FLACS


That sounds rather obvious (-:

, which is almost every one that's not named Apple. I


Older software, and in particular older hardware, still exists, and many
people use it (especially in the case of the hardware [i. e. portable
players]).

see no advantage to using uncompressed WAV files for anything except
editing.


If FLAC is indeed truly lossless, then I see no reason why it should
make any difference whether you use it or WAV for editing. Inside the
editor, the audio is going to be in raw format anyway, not either WAV or
FLAC; it's only when it's being saved or loaded that the format has
meaning. I suppose if the time to save/load is noticeably longer
(because of the compression) and you do a lot of saving while you're
editing it might make a difference to you, but it won't make any
difference to the _quality_ if it's lossless.

If FLAC isn't "universally" supported, it's within inches of it.
Windows 10 even supports it as an OS feature.


Remember one of the 'groups this thread is going to is the XP one, and
the other is the 7 one!
--
J. P. Gilliver. UMRA: 1960/1985 MB++G()AL-IS-Ch++(p)Ar@T+H+Sh0!:`)DNAf

Stop calling unmarried people "single" as if they are incomplete. I prefer
spouse-free. It is not a coincidence we are the new majority.
- Scott Adams, 2015
  #27  
Old April 4th 19, 02:01 PM posted to microsoft.public.windowsxp.general
J. P. Gilliver (John)[_4_]
external usenet poster
 
Posts: 2,679
Default Can a .m4a audio file be converted into a .mp3 one losslessly?

In message , Nil
writes:
On 03 Apr 2019, wrote in
microsoft.public.windowsxp.general:

I think most serious editors do that because it takes a whole lot
longer to "load" than simply to copy the MP3. (Based on Sound
Forge) If you save as WAV it is almost instant. Any other format
chugs along.


If you're saying what I think you're saying, I don't think that's true.
When you open a compressed audio file, most editors will load it into
RAM until you save it.


He's referring to the actual loading into RAM. If you have an older
processor, the time to load is noticeable, especially for a long
recording. (I am still _aware_ that it takes longer to load an .mp3
[into GoldWave] on this apparently four-core processor under 7 than it
does a .wav, but it's sufficiently quick that I don't care.)

You can edit the RAM copy as much as you want
with no data loss, but as soon as you save it to a compressed format
file the data loss occurs.


Agreed for the saved file. Whether what remains in RAM changes, depends
on the editor. GoldWave, when I say save a mono .mp3 from a source that
was (nominally) stereo (and came from a .wav), asks me if I want the
"displayed" (and I assume the RAM-held) to change to reflect the format
saved. I would hope most editors either default to keeping the original
or ask, but I don't know.

If you save multiple times, multiple cycles
of loss occur.


Save _and reload from the saved_, yes. Otherwise, see above. If you're
only saving intermediate results during an editing session, _probably_
not - though if you're doing so just in case you make an edit you want
to undo (further than your editor's "undo" can manage), I wouldn't save
in a lossy format anyway.

As evidence, note that after a save, if you quit the application, it
quits right away. There is no unexpected cleanup. At least, that's have
every audio editor I've used works.


GoldWave too; but I assume all that means is that they dump their
buffers. If by "unexpected cleanup" you are referring to the "do you
want to save changes" prompt you get in lots of software not just audio
editors if you quit _after_ a change without a save, I wouldn't have
called that a "cleanup".
--
J. P. Gilliver. UMRA: 1960/1985 MB++G()AL-IS-Ch++(p)Ar@T+H+Sh0!:`)DNAf

The early worm gets the bird.
  #29  
Old April 4th 19, 05:01 PM posted to microsoft.public.windowsxp.general,alt.windows7.general
Bill in Co[_3_]
external usenet poster
 
Posts: 303
Default Can a .m4a audio file be converted into a .mp3 one losslessly?

Nil wrote:
On 03 Apr 2019, "Bill in Co" surly_curmudgeon@earthlink wrote in
microsoft.public.windowsxp.general:

FLAC is indeed a lossless format, but I've never used it. For one
thing, it only reduces the filesize by about half anyways, so why
bother. And it's not universally supported.


FLAC natively supports information tags that can be read by any player
that plays FLACS, which is almost every one that's not named Apple. I
see no advantage to using uncompressed WAV files for anything except
editing.

If FLAC isn't "universally" supported, it's within inches of it.
Windows 10 even supports it as an OS feature.


I simply meant there are some audio utilities or audio players that won't
recognize it (along with some other formats too), like some portable mp3
players, for example. And since it only reduces the file size by half, for
both reasons I don't see much use for it, but that's just my own take on it.
:-). For me, mp3's are the saving grace, with their concurrent 10:1 or
better reduction in file size, whilst yet negligible sound loss. :-)


  #30  
Old April 5th 19, 01:52 AM posted to microsoft.public.windowsxp.general,alt.windows7.general
Nil[_5_]
external usenet poster
 
Posts: 1,731
Default Can a .m4a audio file be converted into a .mp3 one losslessly?

On 04 Apr 2019, "Bill in Co" surly_curmudgeon@earthlink wrote in
microsoft.public.windowsxp.general:

I simply meant there are some audio utilities or audio players
that won't recognize it (along with some other formats too), like
some portable mp3 players, for example. And since it only reduces
the file size by half, for both reasons I don't see much use for
it, but that's just my own take on it.
:-). For me, mp3's are the saving grace, with their concurrent
:10:1 or
better reduction in file size, whilst yet negligible sound loss.
:-)


Unless it's quite old I can't think of a computer audio utility that
can't handle FLAC. Even most hardware players newer than about 10 years
should be able to play them... unless they are named Apple.

I use MP3s for portable players. I use FLAC for archiving.
 




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