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.
When ever i open vim, i get the error that the following error: E484: Can't open file/abcd/configFiles/vim/syntax/syntax.vim There was a .vimrc file in my home folder that i have removed.
Still i keep getting the same error. Presently in my home folder there is no .gvimrc or .vimrc file.
But still i keep getting the same error. I am not too sure where this file is mentioned.
Background info: The SHELL has been changed from tcsh to bash Earlier i had created a .vimrc file in tcsh, i have removed the .vimrc in bash SHELL.
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?
Does Recordmydesktop have a file size limit? I'm considering using the Zero compression setting to keep CPU usage down, but I don't want to run up against a 2GB or 4GB file size limit. While I know some filesystems impose this limit, most screen recorders I've used have a 2GB or 4GB limit when recording, regardless of the filesystem.Is this an issue with Recordmydesktop
I have rendered out less than two minutes of High Definition animation to Avi-Jpeg Format in Blender. Now, I have this 2 GB file. Going at this rate, it's going to take a whole DVD set to play a 15 min movie.
Surely, this file doesn't need 2 GB just for 1:45 min/sec, right? Even at high def? How do I reduce this file size down to like a manageable number, say, 20 miB?
I changed the case of one letter ('N' to 'n') in a track name using Rhythmbox and the result is a file which is 7727 bytes smaller than the original (I saved a copy of the original file before editing the track name). The file is in ogg/vorbis format (160 Kb/s). I changed the track name back and the file size didn't change. However, the contents of the file are still drastically different (vastly more changed than merely the letter in the track name).I'm worried that Rhythmbox is doing something dreadful like decoding the file, changing the track name, and then completely re-encoding the file from scratch (meaning it runs the decoded audio through the lossy encoder to regenerate the file). Does anyone know whether or not Rhythmbox is doing this when the user edits the meta data (like the track name, artist, etc.)?
I haven't used WinFF before yesterday. Looks simple so I tried to convert an AVI file of 600MB to DV and ended up with an 8GB conversion.For device preset I used Raw DV for Pal Fullscreen as I'm in Australia.Does WinFF always produce such large files?
An arbitrary matrix can be solved using Gauss-Jordan elimination with O(n^3) complexity. A tridiagonal matrix (i.e. a special type of matrix) can be solved using Gauss-Jordan with O(n) complexity. That is, a for an arbitrary matrix, Gauss-Jordan is a cumbersome algorithm, but for a tridiagonal matrix, the algorithm can be expressed as a loop and a simple formula. If I were writing, for example, a compiler or some optimization program, is there a way to test the problem to see if it has become less complex, so I can express the algorithm more simply?
Has anyone else noticed an increase in file size of ripped CD tracks to MP3 between previous releases of Ubuntu and Karmic? Specifically using the same Gstreamer pipeline settings the file size in Karmic is now considerably larger than in previous releases like Jaunty or Intrepid. Is there a fix for this bug. Seems like Sound Juicer/Rhythmbox have some significant bugs in Karmic!
is lvresize with --resizefs options re-size the Logical Volume and then re-size the file system? i mean we don't need to use resize2fs?I looked at man pages but it doesn't explain this option.
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?
today I upgraded via official testing repository Gnome to version 3.18. After this, icons on desktop and nautilus are bigger, than before. Next thing, gaps between icons are smaller than before. I tried change theme to default (Adwaita), then run gtk-update-icon-cache, but without result.
Normal view - icons are big for this view. URL....
Small view - icons are still big for this view. URL...
How can I change icons size and gaps size? Or is it bug for this version?
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've read and read & am still no smarter. I've tried to make a persistent partition(one that mounts whenever I boot/login). Either I don't have the right file or syntax, I've given up and need help. Please tell me exactly which file to edit and the proper syntax to put there. The partition on my machine I want to do this for is "sdb2" uuid "02f5852a-c3e2-47e8-b1dd-93592f1f87ee" label "archives"
