OpenSUSE Multimedia :: FFmpeg Was Removed From The Repository?
Apr 2, 2011
I checked for updates today and noticed that the Packman repository have switched to the new forked FFmpeg library: LibAV. This include gstreamer and other dependent libraries.
VLC runs without a problem but i wonder what other applications depending on FFmpeg will not run...
It looks like FFmpeg was removed from the repository.
Just wondering if anyone else has this partial upgrade being offered to them: ffmpeg, libavdevice52, libpostproc51, libswscale0 are held back (grayed out) in Update Manager.
I only use Update Manager manager to see if there are any updates and then use sudo apt-get update, sudo apt-get-upgrade and sudo apt-get dist-upgrade to actually examine the updates as it shows if anything will be removed without anything to replace it.
And I know to never do a partial upgrade.
Update Manager displays "Partial Upgrade" with the files listed above grayed out.
I did use sudo apt-get-upgrade to install a few packages earlier that were safe to install and now -
sudo apt-get-upgrade currently gives this output:
Code: The following packages have been kept back: ffmpeg libavdevice52 libpostproc51 libswscale0 0 upgraded, 0 newly installed, 0 to remove and 4 not upgraded
I've been having a somewhat disagreeable problem updating ffmpeg and all the other resources that need updating (libavcodec52 et al) for some time now. I'm asked to delete most of the software/libs that are needed for certain software (2man2dvd, etc...). At times it looks like I have some serious contortions to go through in order to update these rpm's. I'm using 11.0.
I'm writing a wrapper for FFMPEG and need help of a multimedia guru. The more I read about containers, formats and codecs the more confused I become, it all seems to be a pretty big mess. To convert a video file with ffmpeg you need to specify a few arguments for ffmpeg:
- input file - some video/audio options (bit rates, sizes and so on) - format (to force format) - codec (to force codec) - output file
My confusion is all about the format and output file extension. If I want to encode my file in MPEG-4, I'll use either 'libx264' or 'libxvid' codec. Then the encoded stream needs to be stored in a file. If I use '.mp4' as the output file, it'll be stored in that file. If I use '.mkv', the video will still be MPEG-4 but stored in a Matroska container. But what do I do with the format option? When I force the format to 'mp4', does it mean I don't have to enter output file, it'll be stored as '.mp4' anyway? Or if I use '-matroska' will I still have to put it into a container '.mkv'? Can I use a '-matroska' format but '.mp4' output file for example?
What's the difference between the format and output file in this context? Or what does forcing the format really do?
I've probably exhausted all possible ffmpeg argument combinations for encoding with a libx264 codec - none worked, the codec always either segfaults or tells me incorrect parameters. I've installed, uninstalled and re-installed all available versions of the codec - no difference. Did anyone have any luck with it? Are there any tricks or conflicts in different arguments? Does the input file have to be in any particular container/format/codec for the x264 to work on?
My examples: >ffmpeg -i video.mpg -vcodec libx264 video.mkv >... Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height >ffmpeg -i video.mpg -vcodec libx264 -b 4000 -s hd1080 -f h264 video.mkv Same error If I use any other codec, they all work fine. give a working ffmpeg arguments example,
I am having problems with ffmpeg. My goal is to capture a video stream from my webcam and feed that into a webcam-capturing program. But to get that to work, I will need ffmpeg to work. I need the following command to work, but I get an error:
Code: $ ffmpeg -b 100K -an -f video4linux2 -s 320x240 -r 10 -i $device -b 100K -f image2pipe -vcodec mjpeg - | perl -pi -e 's/\xFF\xD8/KIRSLESEP\xFF\xD8/ig' ffmpeg: relocation error: /usr/lib/libavfilter.so.2: symbol avformat_find_stream_info, version LIBAVFORMAT_53 not defined in file libavformat.so.53 with link time reference
I'm trying to record a video with sound through FFmpeg, and while it records the system sounds just fine, it doesn't record the microphone (in fact, it crashes on doing so!). The strange thing is that if I use arecord, it works just perfectly, but through ffmpeg it crashes.This is the command that I use to launch FFmpeg:
I've been having trouble with CUPS.I uninstalled it and I wanted to re-install it. When I removed CUPS some other files were removed too. I don't have an option of installing the software or updating a system, in the Control Center.
I've got a bunch of HD files in .mkv format that I want to convert to a format that the 360 will read (currently just trying .avi files) so I can stream the media to my xbox360. I'd rather avoid transcoding the entire file, so I tried unpacking the .mkv file into audio and video (and subtitle if necessary) and repacking said file into avi using ffmpeg. The repacking is where I am having the problem. Here's the commands I tried at first (it might look familiar):
Error while opening codec for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height Where stream #0.1 is the output audio stream to the .avi file. I think I have narrowed it down to the fact that the input streams are all 6 channel (5.1 surround sound) which mp2 doesn't support (the ffmpeg audio default) so I did this instead:
to change the sound to stereo it works. Problem is I'd like to keep my media in surround sound. Any thoughts on how to get all 6 channels in the new mp3 file? I'm fairly certain that mp3s support 5.1. And after some messing around, I haven't found that my ffmpeg can transcode audio format from 5.1 back to 5.1. So since (I think) the 360 can play .ac3 files I decided to just copy both the video and audio file to the .avi file sans transcoding using
however now about halfway through, I run into this error:
Code:
NULL @ 0x22f3a00]error, non monotone timestamps 62562 >= 62562
av_interleaved_write_frame(): Error while opening file I can't seem to find any solution to this problem anywhere online, does anyone know how to get around this? I'd prefer to use ffmpeg for this if possible. (so I can write a script and just attack an entire directory at once) For what it's worth, I'm using the latest ffmpeg in the ubuntu (karmic) default repositories.
It seems more and more people are encoding with the MKV container on bit torrent these days, and a lot of the shows I'm watching are starting to release almost exclusively with .mkv formatted videos. This is not a problem if I want to watch the shows on my computer but I've become accustomed to watching them on my PlayStation3 using my thumb drive. It seems the offical documentation for the PS3 includes a list of supported codecs [URL], but when I use FFMPEG to convert with libxvid video and aac audio in the MP4 container my PS3 says the output video is not supported. I've also tried most combinations of libxvid, libx264, mpeg4 for -vcodec and aac, libmp3lame for -acodec in several different container formats but nothing seems to work. I have found one option that always works:
I don't like doing it this way, however, as the output file is twice the size and the audio quality is terrible. If I don't reduce the audio channels to only two using -ac 2 FFMPEG throws an error (apparently MKV audio supports 6 channels). And preserving the video quality in MPEG video using -sameq produces too a large file (and I prefer to keep my files as lossless as possible). Ideally I want to save the files on an external HD I have but if a single episode of a show is 1.5 GB it's not very pratical.
Anyway, the PS3 docs say it supports h264 and xvid with aac audio, but apparently I'm doing something wrong. Has anyone sucessfully used FFMPEG to convert an MKV to MP4 for use on a PS3?
I have a .mkv container with proper video (h264/~6000kbps) and audio (DTS/1500kbps). For some reason I want to have similar quality but with another video and audio codecs. For video it goes well and ffmpeg is very helpful here, no problems. Also, I want DTS to become AC3/320-448kbps/5-6 channels. Commands like this:
gives me not proper reordered sound in channels, so I hear center on the right, music on the left, etc.
Is there a way to make that properly with ffmpeg?
Also, I have a question about multi core encoding cause I have a quad processor. Ffmpeg seems not using them in all the way, I get something like 25% of CPU usage while encoding. Command -threads, even using the h264, does not make any extra load of cpu.
I'm making little animated bits in GIMP with the gap package. I can only get GIMP to encode the files as .avi. Problem here is that when I try to use any converter, it compresses 80MB files down to 1MB-4MB files and, unsurprisingly, the video quality is terrible. WinFF and ffmpeg (via terminal) output at a drastically reduced size. Is there some way to keep the files big/big-ish? I don't want to lose any detail.
i want to convert DVD movie to mp4 using x264 and aac. I'm having some issues with GUI apps.I use ffmpeg in terminal for all my single file converts and prefer to use it but don't know how to use it in terminal for a DVD movie to mp4.
Recently I installed gtk-recordMyDesktop, so that I could record my movements and send the Media File to my dad, so he could learn how to use Ubuntu too [plug]My problem is converting the .ogv file to .avi - I have tried ffmpeg but it loses quality dramatically.Can anyone tell me how to get better quality video out of ffmpeg, or let me know another way to convert the .ogv file to .avi or something that is widely used?
I've been recording one of my classes with my android phone, and the file it gives me is an .amr file. I'd like to convert that to .mp3, and I was hoping I could use ffmpeg to do it via the command line. Here's what I did, and the output that I got:
Quote:
roger@KarmaLaptop:~/Desktop$ ffmpeg -i Net202-week6.amr Net202-week6.mp3 FFmpeg version 0.6-4:0.6-2ubuntu6, Copyright (c) 2000-2010 the FFmpeg developers built on Oct 5 2010 22:36:53 with gcc 4.4.5 configuration: --extra-version=4:0.6-2ubuntu6 --prefix=/usr --enable-avfilter
ffmpeg version N-30884-g54dd50d, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 20 2011 19:09:46 with gcc 4.4.3 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-
[code]...
At least one output file must be specified Do you see i can't convert my file now how i add subtitle to my video in ffmpeg
have downloaded from internet to a mp4 format, so I can play them in my Nokia 5800.This is command line that works perfectly:ffmpeg -i "input.avi" -f mp4 -vcodec libxvid -s 640x360 -b 768kb -r 25 -aspect 16:9 -acodec libfaac -ab 96kb -ar 44100 -ac "output.mp4"n:Is there a way to make it really quiet so I can run it from cron..
I installed Audacity to convert an mp4 to an mp3. Now, I don't necessarily need Audacity to do this, but for the time being I'm more concerned with getting Audacity to work properly as opposed to getting an mp3 onto my iPod.
In Audacity, I went to Edit>Preferences>Libraries. For the MP3 Export Library, Audacity recognizes LAME, but under the FFmpeg Import/Export Library it says "FFmpeg library not found." I hit the Download button and read this page on how to install FFmpeg, noticing the warning about needing FFmpeg 0.5 or later on Linux.
I fired up Synaptic and searched for "ffmpeg," chose the vanilla version and installed it along with two dependencies which I don't exactly remember but I bet they were libavformat52 and libavdevice52.
Back in the Audacity Preferences window I chose the "Locate..." button to point to the newly installed libraries, but Audacity is not recognizing/installing them successfully. I've tried pointing to the following files:
I have a file with about 6 .flv files and I wish to batch convert them to libmp3lame. I have tried making a #!bin/bash script with all the files in e.g.
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.
one is : /home/pt/t/pa1.flv the other is : /home/pt/t/pa2.flv 1 o merge with ffmpeg ffmpeg -i /home/pt/t/pa1.flv -i /home/pt/t/pa2.flv -vcodec copy -acodec copy /home/pt/t/dd.flv the problem is: the merged file ( /home/pt/t/dd.flv ) just contain one file--the first one /home/pt/t/pa1.flv,there is no the second file--/home/pt/t/pa2.flv in the /home/pt/t/dd.flv
I just installed 10.04 today and i love it I'm trying to install some software tried devede but i didn't like it only text menu's and nothing more.I also found tovid software looks really cool and i was wanting to try it i tried to install the debs but no go the tovid website say's to install from subversion it also says something about installing ffmpeg from subversion.So what is subversion and how do i install from it?
I recently developed a taste for the Alac format and ffmpeg will oblige with this line of code Code: ffmpeg -i <input> -acodec alac <output>.m4a and this worked beautifully one file at a time and How does one do all the files in a given folder? Is there an asterisk one adds as in shntool.
I've seen posts with similar titles on these forums, but I know nothing about the plethora of codes out there and all those thread seem to be way over my head. I've installed ffmpeg (an unrestricted version) but I can't convert m4a audio files to mp3 audio files. I installed a package called 'libavcodec52' from synaptic because it came up in the search results for 'm4a' and its description said something about m4a and ffmpeg but still no luck..
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'm having trouble to find the right ffmpeg options to encode a video that can be read on a htc G1 cell phone. I have used several codecs and formats but none is working.
I have followed these instruction to install ffmpeg and x264 [URL]
Here is my ffmpeg config :
Code: FFmpeg version SVN-r24953, Copyright (c) 2000-2010 the FFmpeg developers built on Aug 27 2010 22:44:01 with gcc 4.4.1 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-