Following this tutorial: [URL] I got everything to work, but the running capture framerate is about half of that set (-r 30). I'm stuck on getting something like 14 fps, which is noticeably choppy. I have a relatively decent system (dual core intel), so I don't think I'm hitting the limitations of the machine (it's only using little over the single core during the process).
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I'm aware that this is a common issue - I've spent "some time" googling. But I think my fault is not the common one. I'm trying to set up my old-ish laptop as a Linux astronomy box. I've installed Ubuntu 10.10 on my FujitsuSiemens AmiloPro V2000.
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I downloaded camE and camorama in hope of controling my webcam's framerate for skype because right now it's 24-30 which gives me dark video. But if I lower the frame rate to something like 10, the video magically brightens. camE doesn't do anything..like it turns on my webcam and does some ftp..fails..turns of my webcam ..back to normal. Camorama can't connect to my webcame. Cheese, guvideo are successfully able to conect to my webcam and play around with it.What I want to do is edit my /dev/video0 or some file from where I can fix the framerate for skype.
View 5 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'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
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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
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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-
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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..
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
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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
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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?
View 2 Replies View RelatedI 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..
View 9 Replies View RelatedWhen using libmp3lame with ffmpeg is there a way to use lame options?
Code:
ffmpeg -i foo -acodec libmp3lame foo.mp3
For example:- --vbr-new, --abr <bitrate>, -V (n) and -q <arg>
I think it is possible to pipe ffmpeg to lame then use the options.But is it possible to do it without a pipe.
Code:
ffmpeg -i foo -f wav - | lame --vbr-new -V 4 - foo.mp3
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 RelatedI'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-
[Code].....
I am converting a lot of videos through FFMPEG using X264 and AAC as the codecs. Some videos have mono sound, some have 2 channels, and some have 5.1. I compiled FFMPEG from source so the codecs are installed correctly but I keep getting an error with 5.1 sound. After the conversion the video plays the sound that should come from the center speak through the right. I tried libfaac and the experimental aac codec inside FFMPEG and both give me the same error.
View 9 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
Last januari my son was born, made HD recording with my HD JVC camcorder,This recorder is producing MTS files, which now i want to convert to H264, see the commandline below:
Code:
hemertje@hemertje-laptop:~/video$ ffmpeg -fpre /tmp/ffmpeg.preset -strict experimental -vsync 1 -async 10000 -i "20110419 - lekker spelen.MTS" -vcodec libx264 -crf 24 -acodec aac -ac 2 -ar 48000 -ab 128k -f mp4 "20110419 - lekker spelen.MTS_MP4"
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I am trying to get a listing of video information using ffmpeg or mplayer. I have found a way BUT it is only returning 1 piece of info..
Here is what I'm doing in PHP:
print exec("/usr/local/bin/mplayer -identify file.flv -ao null -vo null -frames 0 2>/dev/null | grep ^ID_");
But this will not provide an entire listing, it will only print out 1 thing such as:
ID_AUDIO_CODEC=mp3
I can specify what exactly I want to know something like:
print exec("/usr/local/bin/mplayer -identify file.flv -ao null -vo null -frames 0 2>/dev/null | grep ^ID_VIDEO_WIDTH");
but I want to just have all the info available at once instead of having to loop through like 20 times..
I creating a mini soft for screencast (audio + video) with ffmpeg.For the video,it's ok.For the sound, the capture of sound of my webcam (/dev/dsp1) it's ok.
Code:
ffmpeg -f oss -ar 44100 -i /dev/dsp1 -acodec mp2 -ab 128k test.mp3
But for my audio desktop :
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I've written a bash script that extracts the audio out of a movie file and saves it out in separate .wav files using ffmpeg. Here is the key ffmpeg command:
Code:
ffmpeg -i $movie_file -vn -ss $start_time -t $duration ${file_name}_${counter}.wav
$start_time and $duration are floating point numbers that contain time information in the form of seconds and milliseconds (ss.xxx). It is important that I am able to control the time down to the millisecond.
My script works exactly as hoped on .avi files. My script needs to work on an existing archive of .mov files. when I try to use it on a .mov, the audio files created always have start times and durations that have been rounded up to the nearest half-second. This breaks up the audio in the wrong spots, and creates files that last too long and have too much extra audio. What can I do to make my ffmpeg command create .wav files from .mov files that properly recognize a specific number of milliseconds? Or, asked another way, how can I eliminate the rounding behavior?
