On Mon, Feb 19, 2007 at 03:31:07PM -0500, H.S. wrote:
> [...]
> Audacity:
> 1. If the input waveform seems to go beyond the +1 and -1 scale, what
> does that signify? I assume that shows recording circuit is being
> saturated and that the output from mixer should be reduced.
> 2. If the input waveform is being shown saturated, how would that
> manifest itself in the playback?
> [...]
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
(everyone knows this squeaking sound, if he/she has ever been to a live
concert where the output from the speakers was fed back into the
micro). It is also called clipping, for more info look here:
http://en.wikipedia.org/wiki/Clipping_%28music%29
> [...]
> 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?
> [...]
> Exporting to mp3
> 1. I would like to export a number of wav files to mp3 files. Instead of
> doing it one by one from audacity, how can I export them using a shell
> script? I want to be able to set some basic tag info in a file and call
> that file to fill in the mp3 tags automatically. In essence, I want to
> call a script that converts all wav files in a directory to mp3 files.
> And of course, I would like to be able to set the bitrate in the script.
> Suggestions on which tool to use for this?
> 2. I can export to ogg format from audacity. Can I do the same thing as
> (1) for this as well? Does ogg format support tags?
> [...]
$ 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
this:
for file in *.wav
do
oggenc --artist `cat "$file".tag | grep "^Artist:" | sed -e 's/^.*:[ ]*//'` "$file"
done
Of course this is just a quick and dirty example to get the idea.
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.
Regards
--
Marcus Blumhagen
"Any intelligent fool can make things bigger, more complex, and more
violent. It takes a touch of genius -- and a lot of courage -- to move
in the opposite direction."
-- Albert Einstein
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