Ubuntu Multimedia :: Ffmpeg Crash When Trying To Overwrite File?
Jul 28, 2010
I'm getting some funny behavior from ffmpeg when I'm trying to increase the volume of an mp3 file and have it overwrite itself. For example, if I execute the following command:
Code:
ffmpeg -i name.mp3 -vol 1024 -y name.mp3
it will code the first 5 seconds or so and then quit with:
i try to split mpeg file using ffmpeg. The splitting itself works OK, but the quality is lower. What should I do in order to keep the same video quality?
After conversion : file01.vb750.432x336.mp4 - 432x336, 302Mb.
If I alter the output file dimensions to 352x272 I then get a file size of 301MB, which is just 1MB less than the file of dimensions 432x336 ? I had expected size to be 30-40MB lower. I do not want to compromise quality by reducing the bitrate to below 750 kb/s so how can I alter the ffmpeg command syntax to get a considerably smaller file size with video dimensions 352x272 ? I would like to use the default codecs and avoid x264 for the mean time.
I was trying to run record it now, and switched from recordmydesktop to ffmpeg and it kept crashing. I tried it with the command line
Code: ffmpeg -r 30 -s 1680x1050 -f x11grab -i :0.0 -vcodec msmpg2v2 -qscale 2 filename.avi I have the following installed, but it's wanting libx264 which I don't see at slackbuilds.org:
[Code]....
Where should I start looking for missing things or...?
I also looked for xvidcap and kazam at slackbuilds.org but didn't see them there. Are those available?
We have some large files with sampling data in it. Don't want to delete these files. But want to quickly overwrite the file with 0s and/or 1s and preserve the original file size.
Asuming I have two files, one large file and one small file, I want to write the smaller file to the large file without overwriting the remaining part of the larger file.
Both are binary files, and the large file can become very large, so I want to avoid copying the whole file, as that will take some time. Is there any standard Linux console utility to do this, or do I need to write it myself?
I try to write to a file "date > file" but Linux says 'cannot overwrite existing file'. I tried chmod 755 but still cannot write to the file. What should I do?
I've recently left Windows behind for good and have come to Linux (Ubuntu.) However, I've run into problems after trying to dual boot windows 7 and ubuntu. When I restarted my computer, I got a black screen with white text reading "Minimal BASH-like line editing is supported". I have the Ubuntu live cd and was able to try Ubuntu without having to install it, but after I put in a flash drive with Windows 7 on it, the Ubuntu CD has stopped working entirely.
The weird thing is, the computer that I installed Ubuntu on, already had Ubuntu and Windows 7 on it. And every OS was booting successfully. The reason I deleted the Ubuntu that was already on the computer, is because there were many different versions. I wanted to do a clean sweep and only have Ubuntu and Windows 7 installed on the computer. I know that if you try to dual boot Windows 7 and Ubuntu, Windows will overwrite the GRUB file, making it unable to dual boot.
For example I want a file to be processed by sed, and then overwrite the file with sed's output. I would try this: Code: sed '<regex goes here>' myfile > myfile But it doesn't work as expected, instead it empties the file (I am thinking that as the first byte comes out of sed, it overwrites the whole file and sed has nothing more to do). How can I make this work?
Code: dd bs=8192 if=Image of=/dev/PS0 In the above command, I cant able to understand the option "of=/dev/PS0"
Whether it means, 1) to write the input file "Image" in to "/dev/PS0" of the Floppy device or 2) to overwrite the same input file "Image" in a file format ("/dev/PS0") of Floppy.
I have written a spec file for packaging my application in an rpm, but it reports a conflict when I am trying to update my service script in /etc/init.d/. This is a file that I do need to update/replace so how might I force the rpm to do that?
Code:
Summary: A program that increases WAN performance. Name: packetsqueezer Version: 0.3.05 Release: 1
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?
since arch has started using python3, bleachbit is broken, so i am looking for a system cleaner to replace it. i need something that will overwrite file contents and clear firefox cache + the all the normal functions of a system cleaner.
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
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?
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:
