Software :: Batch-convert A Bunch Of Old Wma Files To Ogg Vorbis?
Jan 15, 2010
I'm not asking for help here, just documenting something I just discovered. Yesterday I wanted to batch-convert a bunch of old wma files to ogg vorbis. Not wanting to go through intermediate wav files, I tried to use ffmpeg to do it in one go. I first tried using the following command (in a loop, which I won't print here).
ffmpeg -i $file -f ogg -acodec vorbis -ab 192k outputdir/$file "vorbis" turns out to be the built-in libavc implementation of the codec. In the process I discovered that the -ab value is always ignored. No matter what value you put, the output is always the default 64k (average, but of course it's vbr). You can however use the poorly-documented -aq option to set the audio quality used. The values don't correspond to the oggenc values though, being a number ranging from 10-100 (or more, I don't know what the maximum is). It's not exactly clear what number corresponds to what average bitrate, so you have to experiment. ~30 seems to give you an average-rate file, while anything above 60 is probably overkill.
Switching to the external libvorbis gave me more flexibility, although at a cost of much longer encoding times (note that ffmpeg must have been compiled with libvorbis support first).
I could use both -ab and -aq (with the numbers corresponding to the oggenc values), with no problems. ffmpeg does display some wrong values in it's output text, however. In addition, there's one more difference. The vorbis (libavc) codec provides an entry in the header of the ogg container reporting the average bitrate, but it doesn't appear to provide a similar bitrate header in the vorbis stream itself. Some programs may not report the bitrate value because of this.
libvorbis provides both headers, avoiding that problem. So to summarize, libvorbis appears to be a better codec choice than vorbis.
I have Ubuntu 9.04 and just installed Sound Converter. I am trying to convert a bunch of .ogg files to mp3 to play on my iPod and it's not working so well. In the Sound Converter options I have is set to convert to high quality mp3. I choose the folder that the files are in and after a moment (slow laptop) Sound Converter populates, I hit 'convert' and it shows that the conversion completes in two seconds. All that it did was create the new folder structure of artist/album but there is nothing in there. Not sure what I am missing. I used Sound Converter before and it worked fine.
I keep trying to convert a bunch of jpg files into pdf, but ImageMagick just seems to keep failing there. Well well, after three thousand fix and reinstall attempts (seriously, I've been trying to fix it for the last month or so), this is what I'm getting today:
So I have a php script that is setup to stream flash video (.flv) and I absolutely love having it. The problem is that any files I want to stream have to be in .flv format for it to work properly as .avi and others obviously don't stream well. Up until now, I've used FFMpeg to change the format from .avi to .flv, however the process takes a lot of time if you have a lot of videos, added to that you have to do one file at a time definitely makes it a pain.Does anyone know of a bash script that can take all the files (i.e. avi, .wmv, .mkv, .mpeg4) in one folder and automatically convert it to .flv? Then possibly delete the old files? Low resolution is fine, so long as it at least viewable. Does anyone have a script or know of a program that can do this (I run Ubuntu 10.04). I think FFMpeg has the best chance of doing this, but I don't know the syntax to actually do so.
I've searched the internet, and while I have found a few scripts, they didn't work for me (still looking into two scripts I found.I am currently messing with them to see if I can get them to work).It would be immensely helpful if someone knew of a way to do this.I also forget to mention that I have used Winff, but I was looking more for a bash script to do this so that I can set a cron job to convert them every hour or so.
a movie is encoded with AC3 in 6 channel audio, what I get out is all of the sounds except for voices, which in 5.1 would be sent to the center channel. What I usually do is fire up avidemux and convert the audio to mp3 stereo, as converting to a 5.1 format usually ends up with a very odd sound (like running everything through an echo chamber). What I'd like to do is run a script to batch-convert these files from AC3 to MP3. The video format may vary, but they are usually XVID. I am comfortable at the command line, but I am not well-versed in audio/video tool terms. I don't need anything extravagant, I just want something that works. Heck, even if it is done one at a time, having a shell script that I can use to simply type:
I am looking for a way to batch convert from DVDs to avi - ideally choosing the resolution of my output device (for use on an Android/iPhone). I really want to do it automatically/via a script from the command line if possible.
Is there an easy way to batch convert CGM images into anything modern (preferably SVG, because they're vector graphics)? The furthest I've gotten is ImageMagick, which tries to open them, but dies saying it can't find "ralcgm", mhich I can't find in any repos.
I have inserted all the filenames individually into the script but when I ran it I got too many errors and i was wondering if someone knew a quicker way to do it e.g. a script that would batch convert all .flv files in that folder to .mp3 format.
I have many video files that I'm trying to convert from *** to .mp4..Currently I'm using Handbrake which does a good job but getting it started is very tedious. In Handbrake I need to confirm and add to queue all of the files. When there are over 200 files at a time it takes way too long. If there is a way to not confirm all of the files please let me know.What program can I use to just add a folder and have it automatically add all of the files to my queue?
I have a bunch of text files that I created with mousepad in xfce. I didn't really think I would need to share them, but I guess I have to. Is there anyway I can batch convert these to rtf so they could be viewed on a windows client?
I have few thousands of icons from my OS/2 PC and I would like to convert them to format acceptable by LINUX GUI (*.png, *.xpm).I attempted to open an OS/2 *.ico files with few LINUx graphical apps (GIMP,Fspot, gThumb,Gwenview,Kolourpaint,Okular) but none can understand the format. It's somewhat problematic for me to convert under OS/2 now so I'm looking for a LINUX app.Are there any LINUX apps that can convert OS/2 *.ico files to a LINUX format in BATCH MODE? If it requires manually "open then save-as", I can't repeat it few thousands of times.
I have a large (~60GB) collection of music in various formats on my hard drive. It is organised in the form Artist/Album/*.ext
The formats include M4A, FLAC, MP3, and OGG. What I would like to do is convert the entire directory, keeping subfolders and ID3 information intact. I would preferably like to be able to do this with a single script.
I am running Ubuntu 10.10 x86_64. I am fairly adept with BASH and the command line, so I foresee no problems there. If I have to write my own script, these are the things I'm not sure about:
(a) maintaining the directory structure. (b) how to tell the script which converter tool to use (LAME, FLAC, etc. (c) keeping ID3 tags.
I have been playing with JWM source and found this cool tutorial at Debian Forums about how its easier to generate a .deb vs installing from source in the traditional manner (./configure, make,etc)[URL] My problem is that when doing the command
dpkg-buildpackage -rfakeroot -us -uc
it starts over, destroys the previous jwm stuff, including my custom files and generates a .deb so, how do I stop it from "cleaning" when i run the above command?
I'm trying to use convert, I have installed the imagemagick. I use this line:convert *.jpg test.pdf but I'm only able to convert to pdf 1 single jpg file, not multiple files at once. When there's more than one file, I get the following error: Segmentation fault
I keep a backup of a bunch of files on a flash drive, so that whenever I change distributions I can just restore all my Android stuff (saves on re-downloading everything). One of these is the Android SDK.
In my ~/.bashrc I add the paths to some executables in the SDK, only if the directory exists, and only if the path is not already in $PATH. For the Android NDK this works fine, but for the SDK I get this:
Code: snfo@snfo:~$ adb devices bash: /home/snfo/Android/sdk/platform-tools/adb: No such file or directory snfo@snfo:~$ ls -F /home/snfo/Android/sdk/platform-tools/adb /home/snfo/Android/sdk/platform-tools/adb*
Everything else is fine though, just that one path is causing trouble.
Now, I've saw something similar to this before whenever you move an executable from one place to another. If you don't re-source your bash config it will continue to keep looking wherever it used to be located. But I've never moved these files.
Basically I need to rename a bunch of .doc files using the for-structure and the mv command (w/ wildcards) in bash. I guess this would be a bit easier if I'd use the rename command, but since this is a school assignment of sorts I need to use for & mv. The .doc files are named "1filename.doc", "2filename.doc" etc. And I've got to rename them to "aaa_1filename.doc", "aaa_2filename.doc", "aaa_3filename.doc" and so on. Tried to dabble quite a bit with the for and mv commands, basically just got a bunch of errors. Every damn time. For 2 hours. The most common error was "mv: missing destination file operand after ..."
I have an rsnapshot backup that I need to move off of a corrupt Linux file system. I need to preserve the internal hardlinks. I've tried rsync -H and using a newer rsync and neither preserve the hardlinks on OS X.
I tried to get rsync -H working and I've isolated it to the file system mounted. I can preserve hard links copying locally (HFS to HFS) but it doesn't preserve when I try to rsync off of a SMB file system mount or AFP file system mount. Is there some mount option solution to getting OS X rsync to obey -H?
is there a simple shell script that would recurse all /home/xxx/public_html directories, and then yank this line (it will always be exactly the same) and better yet, for future, is there any way I can REPACE that line with another..
I recently installed JDK 6 runtime using apt-get install in terminal. I downloaded a .jar file and attempted to run it but I got an error telling me it has blocked the file for some reason.Another thing was, how can I run batch files? I know ubuntu doesn't come with something like MS DOS but is there anything similar that I can run batch files with?
I have a lot of .flac files downloaded from several sites. Most of them come with a .cue file, and the .jpg with the cover, etc. It seems it is the intention of the uploader that one rebuilds the original CDDA. However, if I had a stand-alone CD/DVD player with flac I would hardly see the point of converting the flac to cdda. Furthermore, I could even play the flacs with a software player although, in this case, the audio quality would not be so good due to the noise picked up by the signal from the PC digital circuits.