Minimal example: transcode from MP3 to WMA:
ffmpeg -i input.mp3 output.wma
You can get the list of supported formats with:
ffmpeg -formats
Convert WAV to MP3, mix down to mono (use 1 audio channel), set bit rate to 64 kbps and sample rate to 22050 Hz:
ffmpeg -i input.wav -ac 1 -ab 64000 -ar 22050 output.mp3
Convert any MP3 file to WAV 16khz mono 16bit:
ffmpeg -i 111.mp3 -acodec pcm_s16le -ac 1 -ar 16000 out.wav
Convert any MP3 file to WAV 20khz mono 16bit for ADDAC WAV Player:
ffmpeg -i 111.mp3 -acodec pcm_s16le -ac 1 -ar 22050 out.wav
cd into dir for batch process:
for i in *.mp3; do ffmpeg -i "$i" -acodec pcm_s16le -ac 1 -ar 22050 "${i%.mp3}-encoded.wav"; done
Picking the 30 seconds fragment at an offset of 1 minute:
In seconds
ffmpeg -i input.mp3 -ss 60 -t 30 output.wav
In HH:MM:SS format
ffmpeg -i input.mp3 -ss 0:01:00 -t 0:00:30 output.wav
ffmpeg -i input-video.avi -vn -acodec copy output-audio.aac
vn is no video.
acodec copy says use the same audio stream that's already in there.
ffmpeg -i video.mp4 -f mp3 -ab 192000 -vn music.mp3
The -i option in the above command is simple: it is the path to the input file. The second option -f mp3 tells ffmpeg that the ouput is in mp3 format. The third option i.e -ab 192000 tells ffmpeg that we want the output to be encoded at 192Kbps and -vn tells ffmpeg that we dont want video. The last param is the name of the output file.
Add the fade-in effect (4 seconds):
ffmpeg -i test01.mp3 -af "afade=t=in:ss=0:d=4" test02.mp3
Add the fade-out effect (5 seconds, starting at 65 seconds) to the new file:
ffmpeg -i test02.mp3 -af "afade=t=out:st=65:d=5" test03.mp3
strip audio stream away from video
ffmpeg -i INPUT.mp4 -codec copy -an OUTPUT.mp4
combine the two streams together (new audio with originally exisiting video)
ffmpeg -i INPUT.mp4 -i AUDIO.wav -shortest -c:v copy -c:a aac -b:a 256k OUTPUT.mp4
You say you want to "extract audio from them (mp3 or ogg)". But what if the audio in the mp4 file is not one of those? you'd have to transcode anyway. So why not leave the audio format detection up to ffmpeg?
To convert one file:
ffmpeg -i videofile.mp4 -vn -acodec libvorbis audiofile.ogg
To convert many files:
for vid in *.mp4; do ffmpeg -i "$vid" -vn -acodec libvorbis "${vid%.mp4}.ogg"; done
You can of course select any ffmpeg parameters for audio encoding that you like, to set things like bitrate and so on.
Use -acodec libmp3lame and change the extension from .ogg to .mp3 for mp3 encoding.
If what you want is to really extract the audio, you can simply "copy" the audio track to a file using -acodec copy. Of course, the main difference is that transcoding is slow and cpu-intensive, while copying is really quick as you're just moving bytes from one file to another. Here's how to copy just the audio track (assuming it's in mp3 format):
ffmpeg -i videofile.mp4 -vn -acodec copy audiofile.mp3
Note that in this case, the audiofile format has to be consistent with what the container has (i.e. if the audio is AAC format, you have to say audiofile.aac). You can use the ffprobe command to see which formats you have, this may provide some information:
for file in *; do ffprobe $file 2>&1 |grep Audio; done
A possible way to automatically parse the audio codec and name the audio file accordingly would be:
for file in *mp4 *avi; do ffmpeg -i "$file" -vn -acodec copy "$file".ffprobe "$file" 2>&1 |sed -rn 's/.Audio: (...), ./\1/p'; done
Note that this command uses sed to parse output from ffprobe for each file, it assumes a 3-letter audio codec name (e.g. mp3, ogg, aac) and will break with anything different.
Encoding multiple files
You can use a Bash "for loop" to encode all files in a directory:
$ mkdir newfiles
$ for f in *.m4a; do ffmpeg -i "$f" -codec:v copy -codec:a libmp3lame -q:a 2 newfiles/"${f%.m4a}.mp3"; done
ffmpeg -i input.m4a -acodec libmp3lame -ab 128k output.mp3
m4a to mp3 conversion with ffmpeg and lame
A batch file version of the same command would be:
for f in *.m4a; do ffmpeg -i "$f" -acodec libmp3lame -ab 256k "${f%.m4a}.mp3"; done
$ vf [ss][filename][outputFileName]
where vf is a custom bash script as follows:
$ ffmpeg -ss $1 -i $2 -qmin 1 -q:v 1 -qscale:v 2 -frames:v 1 -huffman optimal $3.jpg
ss offset = frame number divided by FPS of video = the decimal (in milliseconds) ffmpeg needs i.e. 130.5
file names in folder, if they contain spaces, must be properly escaped
ls * | perl -ne 'print "file $_"' | ffmpeg -f concat -i - -c copy merged.mp4
$ ffmpeg -i video.flv image%d.jpg
$ ffmpeg -f image2 -i image%d.jpg imagestovideo.mpg
$ ffmpeg -loop 1 -i image.png -c:v libx264 -t 15 -pix_fmt yuv420p -vf scale=320:240 out.mp4
$ ffmpeg -i example.mp4 -f webm -c:v libvpx -b:v 1M -acodec libvorbis example.webm -hide_banner
more info
ffmpeg -i audio.xxx -c:a flac audio.flac
You can modify a video file directly without having to re-encode the video stream. However the audio stream will have to be re-encoded.
Left channel to mono:
ffmpeg -i video.mp4 -map_channel 0.1.0 -c:v copy mono.mp4
Left channel to stereo:
ffmpeg -i video.mp4 -map_channel 0.1.0 -map_channel 0.1.0 -c:v copy stereo.mp4
If you want to use the right channel, write 0.1.1 instead of 0.1.0.
Here's a command line that will slice to 30 seconds without transcoding:
ffmpeg -t 30 -i inputfile.mp3 -acodec copy outputfile.mp3
$ ffmpeg -r 60 -f image2 -s 1280x720 -i pic%05d.png -i MP3FILE.mp3 -vcodec libx264 -b 4M -vpre normal -acodec copy OUTPUT.mp4
Do you need to cut video with re-encoding or without re-encoding mode? You can try to following below command.
Synopsis: ffmpeg -i [input_file] -ss [start_seconds] -t [duration_seconds] [output_file]
Example:
ffmpeg -i source.mp4 -ss 00:00:05 -t 00:00:10 -c copy cut_video.mp4
Example:
ffmpeg -ss 00:00:05 -i source.mp4 -t 00:00:10 -async 1 -strict -2 cut_video.mp4
Place "-ss" before "-i" to avoid annoying frame missing problem
If you want to cut off section from the beginning, simply drop -t 00:00:10 from the command
Example:
ffmpeg -i input.avi -vcodec libx264 -crf 24 output.avi
It reduced a 100mb video to 9mb.. Very little change in video quality.
Example:
ffmpeg -i video.mov -vf eq=saturation=0 -s 640x480 -c:v libx264 -crf 24 output.mp4
make a grayscale version and scale to 640x480
ffmpeg -i file.mkv
check for streams that you want (video/audio). be sure to convert/specify DTS 6 channel audio stream
ffmpeg -i input.mkv -strict experimental -map 0:0 -map 0:1 -c:v copy -c:a:1 libmp3lame -b:a 192k -ac 6 output.mp4
ffmpeg -i source.mov -i watermark.png -filter_complex "overlay=x=(main_w-overlay_w)/2:y=(main_h-overlay_h)/2" output.mp4
ffmpeg -i input.mp4 -ss 00:00:05 -c copy -t 12 sliced-output.mp4
ffmpeg -i input.mp4 -ss 00:00:05 -c copy -to 00:00:07 sliced-output.mp4
-ss: the time to start
-t: duration
-to: time to end
`$ cat mylist.txt file '/path/to/file1' file '/path/to/file2' file '/path/to/file3'
$ ffmpeg -f concat -i mylist.txt -c copy output`
$ffmpeg -i input.mp4 -vcodec h264 -acodec mp2 output.mp4
more commands
http://www.catswhocode.com/blog/19-ffmpeg-commands-for-all-needs