I'm installing ffmpeg with libavfilter using this guide but i can not get the movie filter to compile when passing --enable-avfilter-lavf to ffmpeg i'm using libavfilter r5935 and the test copy of ffmpeg.
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how to make videos in Linux and have found some interesting stuff. I subscribe to Linux Journal and ran across a nice video showing how to make a movie with ffmpeg. If you are interested check out this video. [URL]
View 1 Replies View RelatedI am trying to compile OpenCV 2.0 with ffmpeg (with x264) support on my Ubuntu 11.04 64-bit machine. (Since I want to use the binaries provided by other developers, I have to use opencv 2.0 version)
I followed the guide from: [URL] to compile x264 and ffmpeg manually, and succeeded.
Then I followed the guide in the INSTALL file provided by the OpenCV 2.0 package. I use CMake to configure and generate them, and use "make" command to try compiling. However, I got the following error report, which haunted me for almost half a week.
Code:
Linking CXX static library ../../lib/libhighgui_pch_dephelp.a
[ 71%] Built target highgui_pch_dephelp
Scanning dependencies of target pch_Generate_highgui
[ 71%] Generating _highgui.h
[ 71%] [COLOR="rgb(65, 105, 225)"]Generating
[Code].....
I'm running to 11.04 and I can't seem to get the file to play sound. I've tried others and none seem to work. I've tried downloading all sorts of codecs but either not the right ones or just not doing something right...
View 3 Replies View RelatedWhat is an easy way detect corrupt movie file during or before using ffmpeg to encode them? is there a command I can use to check if the file is not corrupt and then pipe it ffmpeg or standard input?
View 1 Replies View RelatedSince ffmpeg 6 is out for some time, when will Ubuntu Lucid afford such a new ffmpeg package from its repository?
View 2 Replies View RelatedI want pass a variable at compile time.
for example in x.c
int global_var = POSITION;
How can I send this POSITION at compile time. ie like # make POSITION=10 Is it possible?
I am having trouble getting ffmpeg to compile.
I have removed x264 and reinstalled making sure shared was enabled.
Then ran ldconfig.
Here is my compile variables
Code:
And here is the error.
Code:
As I understand (and remember) debian does not package x264 with its ffmpeg. [I say remember because I have a debian-multimedia line in my sources.list that I believe I added when I wanted a x264 enabled ffmpeg]
Is this status still the same? Do we still need to compile ffmpeg by hand if we want x264?
I have trouble compiling FFMPEG with VDPAU support. I have clean typical installation Fedora 14 + all updates and after installation I install Nvidia driver with VDPAU support - kmod (VPDAU works well, tested with mplayer - I play MKV 1080p on Atom D525 + chipset Nvidia ION with 10-15% CPU ussage). I try compilie FFMPEG with this procedure:
[Code]...
My system is CentOS 5.5 x86_32, and I am currently trying to get the x264 video encoder to work and to read libav* input, like elementary H.264 streams. All the prebuilt x264 binaries seem to come without lavf and/or ffms support, and I was unable to build ffmpeg/ffmpegsource on my system. So these x264 versions only support RAW YUV 4:2:0 input, which is of no use to me. I would like to use the x264 binary itself, as it seems that ffmpeg+libx264 does not have option mappings for all the options I could give to x264 directly.
Is there any pre-built x264, that supports lavf/libavcodec for reading H.264 input? Or is there a good, working guide on how to build that on CentOS 5? When trying to build latest ffmpeg (trying to get ffms/lavf dependencies on my system), it tells me, that my "pkg-config" is too old, and that I should set FFMPEG_CFLAGS and FFMPEG_LIBS to work around the issue. No idea what I should do here.
Then, when trying latest ffmpegsource (again, to get the ffms/lavf deps satisfied), it tells me I do not have libx264. Maybe here my version is too old or something. I installed libx264_98, maybe wrong version.
Ah, the output of x264's configure script looks like this:
Platform: X86
System: LINUX
asm: yes
avs: no
[Code].....
Is there a way to filter static or noise out of an mp3? I have avidemux and audacity but I'm not sure how to go about it, or if its actually possible.
View 5 Replies View RelatedI was just wondering whether there is some way to permanently turn on the Extrastereo filter in SMPlayer. I have to turn it on every time manually
View 3 Replies View RelatedI mostly need it to filter away background noise from interviews. So, the editor should allow me to tweak the recordings in the frequency domain. I'm a Kubuntu user, so, if possible, propose me KDE apps!
View 1 Replies View RelatedI'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):
Code:
mkvextract tracks file.mkv 1:temp_video.avi 2:temp_audio.ogg 3:temp_sub.srt
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy file.avi
However, I keep getting this annoying error:
Code:
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:
Code:
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy -acodec libmp3lame file.avi
I'm still getting the same error, but when I use
Code:
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy -acodec libmp3lame -ac 2 file.avi
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
Code:
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy -acodec copy file.avi
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.
i want to convert a song from ogg format to mp3 using ffmpeg the command i use to do this is
ffmpeg -i /home/john/Music/input.ogg to mp3 output.mp3
i then get then get a message back saying
"unknown format"
how do i correct this ?
The command line to convert to ogg theora
ffmpeg -y -i input.flv -sameq OUTPUT.ogv
runs perfectly on 9.04 but on 10.04 it gives errors:
Code:
ffmpeg -y -i input.flv -sameq OUTPUT.ogv
FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1, Copyright (c) 2000-2009 Fabrice Bellard, et al.
configuration: --extra-version=4:0.5.1-1ubuntu1 --prefix=/usr --enable-avfilter
[Code].....
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:
Code:
ffmpeg -i inputfile.mkv -sameq -ac 2 outputfile.mpeg
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:
Quote:
ffmpeg -i in.dts/mkv -acodec ac3 -ac 5 -ab 448k out.ac3
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.
View 6 Replies View Relatedi 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.
View 9 Replies View RelatedRecently 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?
View 3 Replies View RelatedI'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
[Code]....
i have a mp4 file and i will convert it with ffmpeg and add new subtitle i use this command
Code:
ffmpeg -i Taylor-Swift-Mine.mp4 -b 768000 -r 24 -s 640x360 -aspect 16:9 -newsubtitle Mine.srt -ab 128000 -ac 2 -ar 44100 out.mp4
and output command
Code:
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
when i try to watch a movie on movie player from the internet i get the requires a Advanced Streaming Format (ASF) demuxer plugin which is not installed. how do i install it?
View 3 Replies View Relatedhave 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..
View 2 Replies View RelatedI 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:
/usr/lib/libavformat.so.52
/usr/lib/libavformat.so.52.36.0
/usr/lib/i686/cmov/libavformat.so.52
/usr/lib/i686/cmov/libavformat.so.52.36.0
/usr/share/lintian/overrides/libavformat52
[Code]....
Did I install the wrong version of FFmpeg or libavformat? Any pointers?
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.
Code:
ffmpeg -i filename.flv -sameq -acodec libmp3lame -f asf filename.mp3
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.
how can i merge two files with ffmpeg?
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
[Code].....
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?
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