I wanted to share an .mp4 file on my webserver but Firefox didn’t play it. It turned out Firefox prefers the .webm and .ogv formats, while Chrome can play .mp4.
Convert mp4 then:
.mp4 to .webm
ffmpeg -i input.mp4 -vcodec libvpx -acodec libvorbis -b:v 600k -cpu-used 4 -threads 8 output.webm
To change the quality, play with the “600k” value (bitrate).
.mp4 to .ogv
ffmpeg -i final.mp4 -vcodec libtheora -acodec libvorbis -b:v 600k -cpu-used 4 -threads 8 final.ogv
Here the difference is the codec (libtheora instead of libvpx).
Maybe I missed something, but the .webm file seemed to me to have better quality than .ogv. So I think it’s enough to support two formats only: .mp4 for Chrome and .webm for Firefox. By the way, in my test the input file (.mp4) had the best quality and smallest filesize, but again, I may miss something.
And here is an HTML5 code that can play your video:
<video controls preload="metadata" width="1024" height="768" poster="images/front.jpg"> <source src="video.mp4" type="video/mp4"> <source src="video.webm" type="video/webm"> <p>Please use a modern browser to view this video.</p> </video>
I wanted to play a 720p .mkv file on an old laptop but it was painfully slow. How to reduce the quality of an .mkv file?
I found the solution here. The following worked for me:
$ ffmpeg -i Movie.mkv -vf scale=-1:360 -c:v libx264 -crf 18 -preset veryslow -c:a copy MyMovie_360p.mkv
Visit the link above for an explanation of the parameters.
ffmpeg -i input.wmv -c:v libx264 -crf 23 -profile:v high -r 30 -c:a libfaac -q:a 100 -ar 48000 output.mp4
More info here.
You have a video with an audio track (call it
video.mp4). You have an audio file (e.g.
audio.mp3) that you want to put on the video, replacing the original audio of the video file.
With Audacity, you can open both
audio.mp3 and you can synchronize the
mp3 to match the original audio. Important points: speech/music should start at the same time, and its length should be the same or a little bit less than the original audio.
Once you have a good
audio.mp3 file, do the replacement with the following command:
ffmpeg -i audio.mp3 -i video.mp4 -c copy final_video.mp4
Before the previous solution, I used Avidemux, but it tends to crash sometimes :) When I have such videos where Avidemux crashes, I use ffmpeg from the command-line.
There are some great audio codecs for FFmpeg which are open source but not completely compatible with GPL, thus they are not included in the FFmpeg package that you can install from the official repositories.
If you want these codecs, you must compile your own FFmpeg.
Here you can find a nice description of the available audio codecs. Which one should you use? In short, they come in this order:
libfaac > Native FFmpeg AAC ≥
libfdk_aac provides the best quality.
You can download static builds from here but they don’t include
libfaac either :(
Fortunately, compiling FFmpeg under Ubuntu is quite easy. You just need to follow this guide.
I have an installer script for Ubuntu called jabbatron. The new version of this script includes (1) compiling FFmpeg, and (2) updating FFmpeg. They are available under the menu point “(170) install from source (mc, tesseract3, ffmpeg)…”. This script automates all the steps that are described in the aforementioned guide.
Update (20130227): I made a resizer script. See the end of the post.
You have an Android/iOS phone and you want to watch movies on it. However, if you transfer a movie to the phone, the media player may stop playing it or it may even freeze. Apparently, the movie is too big for your phone. What to do?
If the media player has problems playing the movie, then it’s too big, thus you should resize it. Then the phone will be able to play it nicely.
Under Windows there is a nice video converter called AVS Video Converter. But if we are under Linux, what can we do?
Well, ffmpeg can do the job for us. But the question is: how to parameterize ffmpeg? :) Here it is:
ffmpeg -i INPUT -codec:v libx264 -quality good -cpu-used 0 -b:v BITRATE -profile:v baseline -level 30 -y -maxrate 2000k -bufsize 2000k -vf scale=WIDTH:HEIGHT -threads THREADS -codec:a libvo_aacenc -b:a 128k OUTPUT.mp4
Let’s see a concrete example. I have a Huawei Ascend Y200 whose resolution is 480×320 pixels. As for the bitrate, 600 kb/s is enough IMO. I want ffmpeg to use 2 cores of the CPU. So:
time ffmpeg -i input.avi -codec:v libx264 -quality good -cpu-used 0 -b:v 600k -profile:v baseline -level 30 -y -maxrate 2000k -bufsize 2000k -vf scale=480:320 -threads 2 -codec:a libvo_aacenc -b:a 128k output.mp4
I like to see how much time the conversion takes, that’s why I added the “
Another advantage of this approach is that you can launch the conversion in batch mode. Say you want to convert all the episodes of your favorite TV show. No problem, just write a little script.
I use Ubuntu and it comes with an old ffmpeg that fails for instance with the parameters above. So I downloaded a static ffmpeg build from http://ffmpeg.gusari.org/static/. FFmpeg exists under Windows too, so the method presented above should work on Windows (though I didn’t try it).
If you prefer GUI applications, you can take a look at HandBrake.
I made a script that can process movies in batch mode. Available here.
ffmpeg -i input.avi -f mp3 output.mp3
This thing stopped working for me under Ubuntu 12.04. However, I had luck with “soundconverter” to extract mp3 from flv files.
If you have problems with ffmpeg, try to compile it yourself.