I am trying to convert a file using ffmpeg. I have this following code in a terminal.
ffmpeg -i introvideofinal -vcodec libx264 -acodec libfaac introvideofinalfinal.mp4
It returns an error dealing just with the h264 codec saying that I need to use a vpre parameter? I can't find any documentation on using the vpre parameter.
I have some videos in an mkv container that are 1920 pixels wide, but less than 1080 pixels high. This causes problems when playing the videos on a PS3 (after converting to an AVCHD system), because the PS3 won't centre the video, leaving a very large black bar at the bottom but none at the top. Is there a way to use mencoder or ffmpeg to losslessly add padding to the top and bottom to make the video 1920x1080?
If I use
Code:
ffmpeg -i video.mkv -acodec copy -vcodec copy -scodec copy -padtop 132 -padbottom 132 out.mkv
then ffmpeg won't do the padding, and the resulting file is basically exactly the same. If I use
Code:
ffmpeg -i video.mkv -acodec copy -vpre libx264-lossless_fast -scodec copy -padtop 132 -padbottom 132 out.mkv
I get
Code:
Output #0, matroska, to 'anr.mkv':
Metadata:
encoder : Lavf52.64.2
[code]......
I am running 64 Bit Debian, 6.0 install with VLC and libavcodec52, libavformat52, and libavdevice52 installed. I also installed recordmydesktop which creates an ogv file, and I would like to use VLC to convert the file to an MP4. Then I can add it to a project under Open Shot Video Editor.
I get the following errors when I try to convert the file: Streaming / Transcoding failed: It seems your FFMPEG (libavcodec) installation lacks the following encoder: H264 - MPEG-4 AVC (part 10). If you don't know how to fix this, ask for support from your distribution. This is not an error inside VLC media player.Do not contact the VideoLAN project about this issue.
Streaming / Transcoding failed: It seems your FFMPEG (libavcodec) installation lacks the following encoder: MPEG AAC Audio. If you don't know how to fix this, ask for support from your distribution. This is not an error inside VLC media player. Do not contact the VideoLAN project about this issue. I've used Google to search for the error message, searched on the forum.videolan.org, and searched on this forum. So far, I've not found a lead to the solution. What am I overlooking/missing?
How do I convert an ovg to avi file?
Why was ffmpeg removed in Jessie?
I read through more of this thread than I wanted to. [URL] ....
Does libav provide command line utilities to do something like converting ovg to avi, or determining the length of an mp3 file? Why is there absolutely no explanation as to why libav was preferred over ffmeg?
Recently I tried to convert a .flv file to an mpeg file using ffmpeg. Although I changed directory to the directory in which the.flv file resided FFMPEG said the file did not exist. However when I gave the "ls" command the file was present. Where is my mistake?
View 2 Replies View RelatedSo I've been trying to convert .avi files to .mov using FFmpeg. I input something like this:
Code:
ffmpeg -i in.avi -b 1000 out.mov
And it says:
Code:
Unsupported codec for output stream #0.1
So I tried adding -acodec mp3 OR -acodec libmp3lame and it says unknown codec.
I went to install libmp3lame-dev using
Code:
sudo apt-get install libmp3lame-dev
And it comes up with this:
Code:
The following NEW packages will be installed:
libmp3lame-dev
The following packages will be REMOVED:
libdc1394-22{u} libfaac0{u} libxvidcore4{u} linux-headers-2.6.28-11{u}
linux-headers-2.6.28-11-generic{u}
I'm fairly sure I don't want those packages removed? Should mention I'm on Ubuntu 9.04 64-Bit.
OpenCV doesn't work. I wonder why would anyone create a RPM package and not bother to check whether it actually works? Do they get paid per package at Red Hat? The problem seems to be in cap_ffmpeg.cpp. Somehow, OpenCV cannot resolve the ffmpeg CODEC libraries, but the annoying part is that it compiles and installs error free - it just doesn't work. Without any error messages, it is really hard to figure out what is wrong.
View 8 Replies View RelatedI'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,
Code:
kenny_strawn@kenny-AOA110:~$ sqlite3 -html -line /var/www/fluxbb/ProprietarySuckware 'select o_base_url from memos'
Error: unable to open database "/var/www/fluxbb/ProprietarySuckware": unable to open database file
kenny_strawn@kenny-AOA110:~$
This database was automatically generated by FluxBB, so I am wondering just why it isn't opening.
I've been trying to re-mux some mkv files into mp4 files recently and ran into this issue. The mkv files contain an h.264 video stream, two aac audio streams (english and japanese) and one subtitle stream. All I'm trying to do is move all four streams into a mp4 container using the following command:
Code:
ffmpeg -i input.mkv -vcodec copy -acodec copy -acodec copy -scodec copy output.m4v -newaudio
Code:
Input #0, matroska, from 'input.mkv':
Duration: 00:24:42.64, start: 0.000000, bitrate: N/A
Stream #0.0: Video: h264, yuv420p, 640x480, PAR 1:1 DAR 4:3, 23.98 tbr, 1k tbn, 47.95 tbc
Stream #0.1(jpn): Audio: aac, 11025 Hz, stereo, s16
[Code]...
Since the first audio stream doesn't seem to be throwing an error I'm a little confused as to why the second (with the same codec) would give this error. I would appreciate any help with this problem, or alternative solutions to accomplishing the mux.
I ftp into the server and see the files, but when i try to make a directory or upload a file it returns an error saying that permission is denied.I am wanting the users to be able to log on with their user account and then be able to have full control over their directories. What can i do to fix this problem?I have enabled port forwarding for ftp. What else is there i can do? If you need any extra information, let me know! I would like to get this fixed asap!Here is my vsftpd.conf file.
# Example config file /etc/vsftpd.conf
#
# The default compiled in settings are fairly paranoid. This sample file
[code]....
I am running shell command through C program using system() routine.
I am executing "opcontrol --status" an executable using the this routine and I get the following error. access: unix error (2) No such file or directory
But when I give the complete path to the executable it runs perfectly.
The executable is installed in "/usr/local/bin/" And the path variable has this path.
I tried to install ffmpeg on 7.04(Fiesty Fawn). when i ran ./configure from terminal window, i got error regarding c compiler that unable to make executable file.
View 5 Replies View RelatedI'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
View 3 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].....
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 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 could stream the following file at my darwin server it means i am able to stream .mov files with 8KHz sampling rate and 20kb/s bite rate.
Quote:
# ffmpeg -i streamingloadtool.mov
FFmpeg version SVN-r10524, Copyright (c) 2000-2007 Fabrice Bellard, et al.
configuration: --enable-libxvid --enable-gpl --enable-libamr-nb --enable-libmp3lame --enable-libfaac --enable-libfaad
[code]....
The problem is i unable to stream this kuraiamr_nb.mov file at my streaming server. when i play this file in quicktime player its stating that this is a unsupported media type.
I am looking for an option to convert .vob file to mp4 format. Command "ffmpeg -i S.vob x.mp4" is giving error message "[libfaac @ 01cac110] Unable to load libfaac.dll". I tried to convert in other formats like .avi with the same command and got success. Conversion is decreasing the quality like hell. Though size of the file after conversion is as big as original then why this bad quality. Some where I have read if I will not specify any argument it will keep the quality same but here it is not happening. Not sure why....
View 2 Replies View RelatedI 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.
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 have some video files (.avi and .swf) & I need to convert them to the following format:
Stream #0.0(eng): Video: mjpeg, yuvj422p, 320x240, 30 tbr, 30 tbn, 30 tbc
Stream #0.1(eng): Audio: pcm_u8, 8000 Hz, mono, s16, 64 kb/s
how to do this with mencoder or ffmpeg?
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.
View 3 Replies View RelatedToday I was trying to convert a video into Gray-scale.(Black & White) I was successful for avi, mp4 ..
And Now trying for 3gp..
Its giving me error which says like The parameter like bit rate, height , width provided is wrong..
what exactly extra parameter I need to put for 3gp
I want to convert all video file in folder a become audio file in folder b use ffmpeg, how to type it in terminal?
View 7 Replies View Related