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nautilus script audio conversion

HI Im running Ubuntu 7.10 and Gnome. I was trying to batch convert a bunch of wave file to high quality mp3's(320kbs). I figured it's best to use those nautilus scripts below:
http://g-scripts.sourceforge.net/cat-multimedia.php
I copied them to the scripts directory, and made them executable.
When I right click in nautilus and choose audio-converter or Naudilus, nothing happens. When I try Wom-audio converter a window comes up where only the mp3's option does anything at all, and it only puts out an empty file (0kb)!?
What am I missing here?

thank you!

update: I found this:
https://bugs.launchpad.net/ubuntu/+s...rt/+bug/130055

and it added a new menu item under scripts, which seems to work, but it also only puts out a file with 0kb!?
why is that?

doing the conversion with lame only, also puts out an emapty file, So here must be where the problem is.

Quote:

lame test.wav test.mp3
LAME 3.97 32bits (http://www.mp3dev.org/)
CPU features: MMX (ASM used), SSE, SSE2
Using polyphase lowpass filter, transition band: 8269 Hz - 8535 Hz
Encoding test.wav to test.mp3
Encoding as 22.05 kHz 32 kbps single-ch MPEG-2 Layer III (11x) qval=3
Only 8, 16, 24 and 32 bit input files supported

Here is the script that I use to convert oggs to mp3s. I first convert the ogg to a wave file and then use lame to convert the temporary wave file to an mp3.

Code:

#!/usr/bin/env  bash

# convert ogg file to wav file using mplayer



# ${1} is the ogg filename

oggfile="${1}"

echo "$oggfile"  # debug

mplayer -vo null -vc null -ao pcm:waveheader:file="${oggfile%.ogg}.wav" "${oggfile}"



# Extract the tag variables TITLE, YEAR, ARTIST, ALBUM

# define get_tags() function

. ~/bin/get_tags "${oggfile}"

get_tags ${oggfile}



if [ -d $DATE ]; then YEAR=$DATE; fi



# Convert .wav file to .mp3 file.  Set the tags as well.

lame "${oggfile%.ogg}.wav"      \

            --tt "${TITLE}"  \

            --ty "${DATE}"  \

            --ta "${ARTIST}" \

            --tl "${ALBUM}"  \

            --tg "${GENRE}"  \

            --tn "${TRACKNUMBER}" \

    "${oggfile%.ogg}.mp3"



# remove the temporary file

rm "${oggfile%.ogg}.wav"

Here is where the get_tags() function is defined:

Code:

cat bin/get_tags

get_tags()

{

    # clear out old tags

    ALBUM='';

    ARTIST='';

    DATE='';

    GENRE='';

    TITLE='';

    # the tags returned by ogginfo are of the form NAME=VALUE.

    # put single quotes around the values to encapsulate white space.

    # eval will set the tag variables ALBUM, ARTIST, DATE, GENRE, TITLE

    eval $(ogginfo ${1} | sed '/[A-Z][A-Z]*=/!d;s/=\(.*\)$/='"'"'\1'"'"'/;s/^\t//')

}

nice, thank you! How would I use this script just for wav to mp3 (high quality, 320kbs) conversion?

This is the part that performs wav to ogg conversion:

Code:

# Convert .wav file to .mp3 file.  Set the tags as well.

lame "${oggfile%.ogg}.wav"      \

            --tt "${TITLE}"  \

            --ty "${DATE}"  \

            --ta "${ARTIST}" \

            --tl "${ALBUM}"  \

            --tg "${GENRE}"  \

            --tn "${TRACKNUMBER}" \

    "${oggfile%.ogg}.mp3"



# remove the temporary file

rm "${oggfile%.ogg}.wav"

It is lame that does the work of conversion, so you can do it with just a lame command.
Look at the manpage of lame for more options such as setting the cbr bitrate or vbr bitrate ranges.

Code:

      CBR (constant bitrate, the default) options:



      -b n  For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)

              n = 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320



              For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)

              n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160



              Default is 128 for MPEG1 and 64 for MPEG2.



      --cbr  enforce use of constant bitrate





      ABR (average bitrate) options:



      --abr n

              Turns on encoding with a targeted average bitrate of n kbits, allowing to use frames of different sizes.  The allowed range of n is 8 -  310,  you  can  use  any

              integer value within that range.



              It  can be combined with the -b and -B switches like: lame --abr 123 -b 64 -B 192 a.wav a.mp3 which would limit the allowed frame sizes between 64 and 192 kbits.

great, thanks you! that's what I was looking for, I also found this:
http://www.experts-exchange.com/Digi..._23692279.html
for i in *.wav; do lame -h -b 192 "$i" "${i%.wav}.mp3"; done
but, I get errors using this line. Sorry Im not well versed in the command line!
I am in a folder with a bunch of .wav files and want to convert them, how do I need to change that line?

thanks in advance for any insight!

I guess my problem is that lame isn't really working from the command line, it always put out an empty file, even when trying to do a simple conversion such as:

Quote:

$ lame -b 128 test.wav test.mp3
LAME 3.97 32bits (http://www.mp3dev.org/)
CPU features: MMX (ASM used), SSE, SSE2
polyphase lowpass filter disabled
Encoding test.wav to test.mp3
Encoding as 22.05 kHz 128 kbps single-ch MPEG-2 Layer III (2.8x) qval=3
Only 8, 16, 24 and 32 bit input files supported
phi@Betelgeuse:~/Music/NEW$ lame -b 32 test.wav test.mp3
LAME 3.97 32bits (http://www.mp3dev.org/)
CPU features: MMX (ASM used), SSE, SSE2
Using polyphase lowpass filter, transition band: 8269 Hz - 8535 Hz
Encoding test.wav to test.mp3
Encoding as 22.05 kHz 32 kbps single-ch MPEG-2 Layer III (11x) qval=3
Only 8, 16, 24 and 32 bit input files supported

This is frustrating, I have no idea why it's doing that!

You might also try using sox to convert the audio.

One thing to check is that you have the needed libraries.

Ah! I have it, but I get:
'SoX was compiled without MP3 encoding support'
same thing probably happened on all the other command line encoders I got installed here.
For some reason ripping off CD's, encodes just fine, but not from my HD.

I guess I have to recompile and reinstall, what a PITA!

Thanks anyhow!~