I have a bunch of mp3 files that are already tagged. I want to re-compress
them with lame to make the files smaller, but the id3 tags are lost while
encoding. Is there a way to retain id3 tags with lame?
Marty
___
mp3encoder mailing list
mp3encoder
ah-thanks!
On Sat, 10 Nov 2007 09:19:19 -0500
"Rick H" <[EMAIL PROTECTED]> wrote:
Marty,
A Google search shows that Lame won't do this. Something called LAME4 may.
*However*
Google mp3 tag editor command line
One program is here:
http://ostermiller.org/mp3tagger/
You
I have a lot of mp3 files that I want to re-compress with lame into smaller
mp3s. I want the input name to be the output name with _24 appended to
it. How could I use a script to do this in Linux? This is the lame command:
lame -m m -b 24 input.mp3 output_24.mp3
Marty
thanks!
On Sat, 10 Nov 2007 23:09:56 +0100
Robert Hegemann <[EMAIL PROTECTED]> wrote:
Am Samstag, 10. November 2007 21:13 schrieb Marty:
> I have a lot of mp3 files that I want to re-compress with lame into smaller
> mp3s. I want the input name to be the output name with _24 ap
frequency of 16 kHz?
I am able to encode another file with 16 kHz and 24 bits with lame. It's
possible-maybe the problem wav file's header needs to be stripped-how?
Marty
___
mp3encoder mailing list
mp3encoder@minnie.tuhs.org
http://minni
10 MB. I have
the updated speeches in wav files and want to compress them to ~ 10 MB. What
lame or sox commands do I need to use?
On Sat, 15 Apr 2006 09:33:15 -0400
Marty Huntzberry <[EMAIL PROTECTED]> wrote:
> I have a large wav speech file (SPM4854.wav) that I would like to encod
the -resample switch worked well. Thanks for everybody's help!
Marty
On Sat, 15 Apr 2006 17:56:48 +0300
"Dimitar Dimitrov" <[EMAIL PROTECTED]> wrote:
> According to lame's manual, -s sould be used only with raw audio files,
> i.e. wav files with no header.
I'm wondering if by using the following:
lame -b 128 --resample 44.1 low.wav high.mp3
if a 96 kbits stream audiodumped to low.wav by mplayer would actually be a
higher quality and better sounding mp3 (high.mp3) that would have 128
kbits?
I use this for my mp3 studies and audio books that I've first used mplayer to
dump to a wav file:
lame -m m -b 24 --resample 22.05 file_in.wav file_out.mp3
Marty
On Fri, 22 Sep 2006 20:30:52 +0100
Thufir <[EMAIL PROTECTED]> wrote:
> what are some good settings for audiobooks?
44.1 and a bitrate of 192 KHz. The files are 1 speaker with a single
acoustic guitar intro. I'm using the previous command with --mp3input and
getting static.
What am I doing wrong?
Marty
___
mp3encoder mailing list
mp3encoder@minnie.tuh
I do have --decode ability and it makes a nice wav file that contains no
static. When I resample the wav file at b 24 and 22.05 kHz I get static in
the resulting mp3 file. I am using Fedora Core 6 on a Athlon 64 and LAME 3.97
for 64 bit pcs.
Marty
On Thu, 16 Nov 2006 18:46:58 +0100
robert
This resamples without static:
lame -m m -b 24 --resample 22.05 -f --scale-r -2 --mp3input input.mp3 output.mp3
I looked at the audio graph in audacity and realized it was recorded to loud so
I took down the volume a little.
Marty
On Wed, 15 Nov 2006 20:14:31 -0500
Marty Huntzberry <[EM
EMAIL PROTECTED]> wrote:
Marty, unfortunately, there's not much you can do since the recording itself
was low quality. 22KHz should have been OK for voice, but looks like the mic
was placed badly
or the acoustics of your room/hall was not the best.
Your best bet would be to forget about lame
khz speech download that sounds fine:
http://media.ccphilly.org:81/Teaching/Audio/B02_Exodus/WED54924.mp3
The 192 kpbs 44 khz speech I want to compress:
http://www.ccbellmawr.com/radioprogram/Week20070312/SH20070313%20Nehem4_5b.mp3
Marty
On Thu, 15 Mar 2007 23:03:47 -0400
"Ishaan Da
sample 22.05 -f --scale-r -2 --mp3input original.mp3
compressed .mp3
Marty
On Fri, 16 Mar 2
On Fri, 16 Mar 2007 18:48:00 -0400
[EMAIL PROTECTED] wrote:
Having listened to your sample, aside from *alot* of clipping, that
audio should reduce to 22kHz mono very nicely. It's a nice clear
recor
und softer. Other values did increase (r -1,
r-3)maybe it's a glitch in lame. Looking at the output in audacity you
can see it decreaes.
Marty
___
mp3encoder mailing list
mp3encoder@minnie.tuhs.org
https://minnie.tuhs.org/mailman/listinfo/mp3encoder
wow-that worked! I tried that in the past with another speech and didn't like
16 kHz sampling but with this mp3 it works fine. I dropped the
re-sampling spec and the r switch and kept it mono:
lame -m m -b 24 --mp3input original.mp3 compressed.mp3
Thanks for all the help!
Marty
Vis
I ripped a cd that had 10 tracks with grip for Linux...so now I have 10 wav
files. Can I use the --nogap option in LAME to merge these 10 files into 1
mp3? If so, how?
Marty
___
mp3encoder mailing list
mp3encoder@minnie.tuhs.org
https
18 matches
Mail list logo
