Software :: Mp4 To Flv Using Ffmpeg On Ubuntu Server 10.04?
Nov 6, 2010
I am doing a conversion of mp4 to flv on a GUI less server where I have only SSH access.( I tried winff and X forwarding that had hanged while doing this conversion so winff is not possible for me)I do not have any idea of the codecs,bitrate of this mp4 video.Some one had converted from an m4v to mp4 and then uploaded this on the server.When I tried converting on command line as follows
i am trying to install ffmpeg on centos machine. i need a source installation for this. i used FFmpeg 0.6.1 .tar.gz file for this. Also i want to install this with maximum codec support. when i used the configuration option as
i run ubuntu 9.10. i have updated all installed packages. i run a 32bit 3.6 ghz intel p4 dual core with three monitors running on two video cards. all this works correctly. i am able to play video in video player and audacious2 on any monitor.
i'm trying to run a upnp media server for my house. myself and both my tenants have playstation 3s, and i would like to host music, music videos, original content, and the backup copies of legally purchased dvds over the network. i recently converted from windows xp, and all of my media files were created by vlc on windows xp, ripped with winamp, edited and rendered with sony vegas, or downloaded from videos. i have tried both mediatomb (from the repository) and fuppes (from subversion) and had the same result with both.
basically, whenever the media center tries to populate the database and search the files, ffmpeg returns an error that crashes the media center. i do not know enough to implement better error handling in the source. both media servers, when running in the terminal, eventually return the same error message when scanning my sepearate hard drive for media. (~100gb of mp3, avi, mpeg, and mp4 files)
the process then ends.the weird thing is that this does not happen when i disable mp4 files in mediatomb (although i have not figured out how to replicate this feature in fuppes) so i have reason to believe that something in the metadata for mp4 files is throwing ffmpeg off. even if the metadata is bad in the mp4 file, it shouldn't cause a crash.
i tried to compile mediatomb without ffmpeg support, but the ./configure script returned errors whenever i tried to append an option to the configure command. i then reinstalled all the ffmpeg packages in the symantec package manager and still no luck.
CentOS 5, installed ffmpeg and compiled the ffmpeg.so, from fffmpeg-php-0.6.0. Everything works fine. Then upgrade the php to 5.2.13 (using an external repository)recompiled ffmpeg.so but now when trying to load php:PHP Warning: PHP Startup: ffmpeg: Unable to initialize moduleModule compiled with module API=20050922, debug=0, thread-safety=0PHP compiled with module API=20060613, debug=0, thread-safety=0These options need to match in Unknown on line 0
I'm trying to convert a .MTS video file to .mov, and I need to use ffmpeg, because I want it to be scriptable.I managed to convert the file to .avi using this command:Code:ffmpeg -i INPUT.MTS -vcodec libxvid -b 18000k -acodec libmp3lame -deinterlace -ab 192k -s 1280x720 -r 50 OUTPUT.avi
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 have been working on vlc (windows) and now i want to use the same in linux...so ma trying to compile it in linux..... i need ffmpeg functionality and others as such...so please provide me with the complete link and order so as where to fetch the source code and to compile....i have done googling and found lots of tutorials and most of the are confusing and not clear......and i tried doing them and am now confused(whether they are compiled or not) cos when i try to instll them it says already exists and when i try to use the functionality it says canot transcode ffmpeg doesnt exitst.....I am badly need of vlc transcoding options. Am using ubuntu 10.
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..
As I cannot use svn to download the latest ffmpeg from trunk.Can anyone explain to me how to use proxychains properly so I can download the latest ffmpeg and compile it?
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.
