Re: about audacity and sound recording on Linux
I am not sure about audacity as I am not familiar with it, but
according to the basics of signal processing going beyond the input
gain is never a good idea since this produces additional harmonics
resulting in harmonic distortion. So depending on the amount of over
gain the resulting sound can be very ugly, since in the extreme case
you turn a harmonic wave into a rectangular wave (since everything
above the limit simply is cut out), which hurts the ears
I see your point. In my case, it appears that the problem is manifesting
in the form of short lived 'static' noise. I am going to try to reduce
the mixer output (which feeds the line-in in the computer) to bring the
levels down. I have been researching this issue (related to practical
situations) and notice that the meter levels in audacity should be more
or less kept near 2/3 of the max level.
3. On a machine, I exported a portion of the captured audio to a wav
file (basically, saved a portion of the input). I then transfered it to
my home computer running Debian. While that sound wave file was shown
between +1 and -1 in the original machine, on my home machine is was
being shown between +0.5 and -0.5 in audacity. What gives?
How did you transfer the WAV? Did you do any more processing to it?
Was it burnt to CD and maybe normalized on the fly?
I exported as wav from aucacity, transfered it to my home computer (via
scp) and opened that wav file in audacity. I don't think there any kind
of processing going on during the exporting the au file to wav.
$ apt-cache search "convert.*wav.*(ogg|mp3)"
soundconverter - convert sound files to other formats
dir2ogg - converts mp3, m4a, and wav files into ogg-vorbis format
Maybe this is what you want, but of course you can convert using plain
oggenc or lame from a shell script using a for loop. Kind of like
for file in *.wav
oggenc --artist `cat "$file".tag | grep "^Artist:" | sed -e 's/^.*:[ ]*//'` "$file"
Of course this is just a quick and dirty example to get the idea.
Looks good. I have to work in this script thing a bit. Ultimately, my
idea is to have a basic set of steps so that anybody can do the
recording and I don't have to be physically present there. Something
1. Instructions on how to open audacity and start and stop the
recording. This is quite simple really.
2. Instructions on selecting and exporting a portion of recorded track
to ogg or wav format. I will probably have a few screen captures and
write up the steps in openoffice.org.
3. If the above export is wav format, I plan to make a script that can
be called once to convert all wav files in the current directory to ogg.
Will probably place it in /usr/local/bin.
BTW, if you want to do all recording and converting on the fly, you
can use arecord to record from alsa and pipe the output directly to
oggenc or lame (but at least for lame with VBR this might brake the
time information, so you get wrong display about play length in some
players). And it does not need to stop here, one can also use sox to
do some additional processing between arecord and oggenc/lame.
Just some ideas of mine. Hopefully this is of help for you.