Re: about audacity and sound recording on Linux
On Mon, 19 Feb 2007 23:03:41 +0100 [EMAIL PROTECTED] wrote: 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 FWIW, sox can do inline recording (and coversion to ogg, or mp3, but the ogg is restricted to a speed of 128, I believe, if you do it on the fly), using the -t ossdsp switch, like $ sox -V -c2 -r 44100 -t ossdsp -w -s /dev/dsp $1.ogg Marcus Blumhagen -- David E. Fox Thanks for letting me [EMAIL PROTECTED]change magnetic patterns [EMAIL PROTECTED] on your hard disk. --- -- To UNSUBSCRIBE, email to [EMAIL PROTECTED] with a subject of unsubscribe. Trouble? Contact [EMAIL PROTECTED]
Re: about audacity and sound recording on Linux
Andrew Sackville-West wrote: On Mon, Feb 19, 2007 at 07:14:00PM -0500, H.S. wrote: [EMAIL PROTECTED] wrote: 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. what if you open the wav on the original machine. does it also show the lower levels? or not? A Very interesting point. It is showing the audio wave between +-0.5 there as well (on the machine). I wonder how this came about to be. -HS -- To UNSUBSCRIBE, email to [EMAIL PROTECTED] with a subject of unsubscribe. Trouble? Contact [EMAIL PROTECTED]
Re: about audacity and sound recording on Linux
On Tue, Feb 20, 2007 at 10:30:34AM -0500, H.S. wrote: Andrew Sackville-West wrote: On Mon, Feb 19, 2007 at 07:14:00PM -0500, H.S. wrote: [EMAIL PROTECTED] wrote: 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. what if you open the wav on the original machine. does it also show the lower levels? or not? Very interesting point. It is showing the audio wave between +-0.5 there as well (on the machine). I wonder how this came about to be. some normalization is obviously happening in the export to .wav. This is not necessarily a bad thing, but if you want the un-normalised data, you'll probably have to leave it in aud format until you get aroundt to mixing/editing or whatever. Of course, you need to fix that saturation/clipping situation anyway, so maybe its not really a problem? A signature.asc Description: Digital signature
Re: about audacity and sound recording on Linux
Andrew Sackville-West wrote: some normalization is obviously happening in the export to .wav. This is not necessarily a bad thing, but if you want the un-normalised data, you'll probably have to leave it in aud format until you get aroundt to mixing/editing or whatever. Of course, you need to fix that saturation/clipping situation anyway, so maybe its not really a problem? A Yes, that clipping is the most serious concern at this time. The +-1 to +-0.5 is not much of a concern, only that I wasn't sure what was happening. I am going to get the clipping removed next time I am down there. Just an aside: the new version of Audacity, 1.3.2, is really a great improvement. It is only in Unstable yet, not in testing though. -HS -- To UNSUBSCRIBE, email to [EMAIL PROTECTED] with a subject of unsubscribe. Trouble? Contact [EMAIL PROTECTED]
about audacity and sound recording on Linux
Hello, I just started to use audacity with some live recorded music. I have a few starting questions: 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? 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? 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? Thanks, -HS -- To UNSUBSCRIBE, email to [EMAIL PROTECTED] with a subject of unsubscribe. Trouble? Contact [EMAIL PROTECTED]
Re: about audacity and sound recording on Linux
On Mon, 2007-02-19 at 15:31 -0500, H.S. wrote: 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. I think that's correct. Any waveform that goes above 1.0 or below -1.0 will be clipped and will result in distortion. From http://www.guidesandtutorials.com/audacity-tracks.html I have only dabbled a little bit with Audacity myself, so I could be wrong of course. For these kind of non Debian specific questions, I suggest you try the mailing lists and forums for Audacity itself. 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? I think this is just an arbitrary value. You can zoom in and out to show it in a different scale. 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? You will need the lame package from the unofficial Debian Multimedia repository as Debian does not ship an mp3 encoder. http://debian-multimedia.org/ 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? Yes, with oggenc from the vorbis-tools package. And yes, Ogg support tags. I suggest you use your favourite scripting language (Bash, Perl, Python etc.) and write a small script for calling lame or oggenc and reading the tags from a file. HTH, -- Cheers, Sven Arvidsson http://www.whiz.se PGP Key ID 760BDD22 signature.asc Description: This is a digitally signed message part
Re: about audacity and sound recording on Linux
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 signature.asc Description: Digital signature
Re: about audacity and sound recording on Linux
[EMAIL PROTECTED] wrote: 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 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. 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 like this: 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. Regards thanks, -HS -- To UNSUBSCRIBE, email to [EMAIL PROTECTED] with a subject of unsubscribe. Trouble? Contact [EMAIL PROTECTED]
Re: about audacity and sound recording on Linux
On Mon, Feb 19, 2007 at 07:14:00PM -0500, H.S. wrote: [EMAIL PROTECTED] wrote: 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. what if you open the wav on the original machine. does it also show the lower levels? or not? A signature.asc Description: Digital signature
