I have a large wav speech file (SPM4854.wav) that I would like to encode to an
mp3 with 16 kHz 24 kbps single-ch MPEG-2 Layer III (10.7x) qval=3. Typing:
lame -s 16 -b 24 SPM4854.wav SPM48524.mp3
encodes an mp3 at 8 kHz and 24 bits. I have sox-can I use that to help me make
a file with a fr
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
will automatically resample the input
> when using high compression ratios.
>
>
> On 4/15/06, Marty Huntzberry <[EMAIL PROTECTED]> wrote:
> > I have a large wav speech file (SPM4854.wav) that I would like to encode to
> > an mp3 with 16 kHz 24 k
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?
Marty
___
m
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? In the case of
On other speeches (1 person speaking for an hour) I use LAME to compress like
this:
lame -m m -b 24 --resample 22.05 -f input.wav output.mp3
This works fine and produces no static. I am now trying to take huge 35 MB mp3
files and compress them. They were previously encoded at a frequency
of 4
<[EMAIL PROTECTED]> wrote:
Am 16.11.2006, 02:14 Uhr, schrieb Marty Huntzberry
<[EMAIL PROTECTED]>:
Hello Marty.
> On other speeches (1 person speaking for an hour) I use LAME to compress
> like this:
>
> lame -m m -b 24 --resample 22.05 -f input.wav output.mp3
>
>
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
The only effect in audacity and lame that seems to work is reducing the audio.
Why does it sound good in it's original mp3 format of 192 kbps and 44
khz but sounds bad at 24 kbps and 22 khz? It's mainly a voice lecture.
On Thu, 15 Mar 2007 16:12:53 +0530
"tech list" <[EMAIL PROTECTED]> wrote:
ch codecs - they will get you much better quality
at the kind of bitrates you want. I'd suggest Speex.
Are you transcoding? (i.e. you don't have the original 22 KHz/16-bit PCM
(WAV) audio, but are using the 192 kbps MP3 as the "source")?
Cheers,
-Ishaan
On 3/15/07, Marty Huntzber
Would you post the commands that you used with postfish, sox, and lame?
I cannot get postfish to compile on my Athlon 64 running Fedora Core Linux 64
bit edition. I'll try compiling in Slackware 11 (32 bit i86 edition)
and see if it works there.
What I'm trying to do is reduce it to a 22 kHz
The version of postfish I found is a pre-release that's pretty old from here:
http://svn.xiph.org/trunk/postfish/
It won't compile in Slackware either.where are you getting postfish?
I tried the -h switch and thought the speaker sounded bad at times.
That r -2 switch makes it sound softer.
it my media server: http://linuxhippy.servemp3.com:8001
On Mon, 19 Mar 2007 20:10:47 +0100
Gabriel Bouvigne <[EMAIL PROTECTED]> wrote:
Marty Huntzberry a écrit :
> That r -2 switch makes it sound softer. Other values did increase (r -1,
> r-3)maybe it's a glitch in lame. Lo
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
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