Ubuntu Multimedia :: How To Batch Convert Files To .flv?
Apr 16, 2011
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'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).
Code: 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 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 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 recently been tasked to extract the subtitles from a lot of mkv files. Hundreds of them, maybe even more than a thousand. To do this, I modified a script I found online:
#!/bin/bash IFS="|" if test -z $1; then
So in the above example the subtitle is actually in track number one and my script would be borked for that particular file. Is there a way to integrate mkvinfo into the script and parse it to see what track should be extracted? Like, read it line-by-line and change the value of some #TRACKNO variable everytime a string like "| + Track number:" appears, and stop when a string like "| + Track type: subtitles" appears? Maybe even skip doing anything if there aren't any subtitles.
PS: I actually prefer SRT subtitles to ***. If there was some command line tool I could use to convert the resulting *** file to SRT I would be much obliged.
System - openSUSE 11.2 "Emerald" KDE (with gnome base) Player - vlc I'm hoping to find a batch of codecs for my newly installed openSUSE OS. I have a very troublesome collection of .mkv files that took several codec packs to make them work. For a brief explanation, I had tried haali and matroska both together and they still didn't work on certain mkvs. I ended up using CCCP, but that's win only as far as I can tell. It took the latest update of CCCP to work on all of my mkvs.
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 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've got a few videos (MKV) which have AC3 audio, I copied them to my Xoom tablet but AC3 is not supported so there is no sound. Hence I need to convert the audio to MP3, can anyone recommend a good program for doing this?
My sister has a Nintendo DSi that can only play music files if converted to aac format. I could do it on my desktop that runs Windows XP, but unfortunately my tower doesn't have an SD slot to do so. The laptop I'm using does have a slot and is running Ubuntu. The thing is, is there any way I can convert mp3 files to aac so they can play on my sister's DSi?
I'm running ClipBucket on my Ubuntu 10.04 LTS server x64.Conversion of AVI to FLV works just fine, however when I try to upload an MPG file, it fails. Apparently I should recompile ffmpeg to allow for MPG > FLV conversions.
I have a collection of *.mkv files which are 720 HD resolution (1920x720). I'd like to watch them on my netbook, but it can't handle HD content. The videos play fine on any decently-powerful desktop, under either Windows XP/7 or Ubuntu, but they are simply too much for the Atom CPU of the netbook.Is there a way to downconvert the .mkv files to something the netbook can handle? It doesn't have any problem with standard DVD content, which is of course lower resolution. Cutting HD content down to the netbook's native resolution(1024x600) would probably be sufficient.
A free software package available in the repositories would be great, but I have no problem manually installing something if necessary. I have not done any work with converting videos to different formats, so I don't know what's out there for Ubuntu.