I tried to convert an AVI-video to a MP4-video for my ipod. Therefore I used ffmpeg as I can read in the FAQ
Code:
ffmpeg -i in.avi -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +mv4 -trellis 2 -cmp 2 -subcmp 2 -s 320x180 -metadata title=X out.mp4
The problem is, that the given encoder libfaac is not installed and I can't get it through the pack-adminstration
[Code]....
I am trying to convert some mp3's into m4r format.I got to step 4 on this page:
Code:
ffmpeg.exe -i ringtone.mp3 -ac 1 -ab 128000 -f mp4 -acodec libfaac -y ringtone.m4r
minus the .exe part. here's what i use:
Code:
ffmpeg -i test.mp3 -ac 1 -ab 128000 -f mp4 -acodec libfaac -y test.m4r
but ffmpeg throws me this error:
Code:
Unknown encoder 'libfaac'
even though i have libfaac0 and libfaac-dev installed
Is there an easy way to enable libfaac in ffmpeg? I'm hoping to not have to compile it myself, but it appears that it was compiled without libfaac support. Not sure if what I'm using is from rpmfusion, but it's enabled.
View 9 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 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'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 Relatedcommand line option in ffmpeg to do this
I am told I need the output file to comply with this
Video Resolution: 480*320
Video Bitrate: 768 kbps
Audio Bitrate: 128 kbps
Video Format: MPEG-4 (be sure not to use H264, as it�s not supported in the current firmware)
[Code]....
i am using Ubuntu 11.04 on my computer system. I urgently need a good video converter for converting videos.I have already installed FFmpeg and men-coder,Winff etc. The problem is each has its own drawback.For instance ffmpeg cannot convert a .avi to .3gp with audio working. My preferences are the converter should be user friendly, should support all popular video formats.
View 2 Replies View RelatedIs there a difference between faac and libfaac? I have an iPod Shuffle.
One of these:-
It plays m4a tracks created using faac encoder.
Like this:-
Code:
ffmpeg -i foo.flac -f wav - | faac -o test1.m4a -b 152 -
But it won't play tracks created using libfaac.
Like this:-
Code:
ffmpeg -i foo.flac -acodec libfaac -ab 152k test2.m4a
I've compared the MediaInfo for the two files.
They look very similar, faac does a more thorough job of tagging. Does ffmpeg do something to the file when it's converting or muxing with libfaac to make it unplayable by the Shuffle?
I've uploaded the two files.
test1.m4a is created using faac
test2.m4a is created using libfaac
They are here:- [URL]. What's the difference?
Using Ubuntu Karmic 9.10
and
FAAC 1.26.1
and
FFmpeg version git-N-28456-g1c31b26
Since 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'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 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 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?
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?
Is there a way I can convert video with ffmpeg to be in mp4 format and play on either iphone or android?
I can't seem to get anything that I convert to play on a phone.
I can get other mp4 files I found to play on the phone but any file I convert into that format won't actually play.
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 am trying to use 'convert' in command prompt to convert image file format.I get the following error.
convert: UnableToOpenConfigureFile `delegates.xml' @ configure.c/GetConfigureOptions/589.
convert: NoDecodeDelegateForThisImageFormat `airplane.jpg' @ constitute.c/ReadImage/530.
convert: MissingAnImageFilename `airplane.ppm' @ convert.c/ConvertImageCommand/2838.
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 RelatedIt 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 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 Related