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From the project home page:

FFmpeg is a complete, cross-platform solution to record, convert and stream audio and video. It includes libavcodec - the leading audio/video codec library.

Package installation

Install the ffmpeg package.

For the development version, install the ffmpeg-gitAUR package. There is also ffmpeg-fullAUR, which is built with as much optional features enabled as possible

Encoding examples

Screen cast

FFmpeg includes the x11grab and ALSA virtual devices that enable capturing the entire user display and audio input.

To create test.mkv with lossless encoding:

$ ffmpeg -f x11grab -video_size 1920x1080 -i $DISPLAY -f alsa -i default -c:v ffvhuff -c:a flac test.mkv

where -video_size specifies the size of the area to capture. Check the FFmpeg manual for examples of how to change the screen or position of the capture area.

To implicitely encode to a shareable size use :

$ ffmpeg -f x11grab -s 1920x1080 -r 25 -i $DISPLAY   -f alsa -i default   -c:v libx264 -b:v 200k -s 1280x720 test.mp4

You may want to adjust the parameters (left-to-right): input format, [input]size, framerate, input (in this case display, but could be a file too), input format, input, codec:video, bitrate:video, [output]size of output. Without context the meaning of the parameters may seem ambigious. See the manpage for the synopsis.

Recording webcam

FFmpeg supports grabbing input from Video4Linux2 devices. The following command will record a video from the webcam, assuming that the webcam is correctly recognized under /dev/video0:

$ ffmpeg -f v4l2 -s 640x480 -i /dev/video0 output.mpg

The above produces a silent video. It is also possible to include audio sources from a microphone. The following command will include a stream from the default ALSA recording device into the video:

$ ffmpeg -f alsa -i default -f v4l2 -s 640x480 -i /dev/video0 output.mpg

To use PulseAudio with an ALSA backend:

$ ffmpeg -f alsa -i pulse -f v4l2 -s 640x480 -i /dev/video0 output.mpg

For a higher quality capture, try encoding the output using higher quality codecs:

$ ffmpeg -f alsa -i default -f v4l2 -s 640x480 -i /dev/video0 -acodec flac \
-vcodec libx264 output.mkv

VOB to any container

Concatenate the desired VOB files into a single stream and mux them to MPEG-2:

$ cat f0.VOB f1.VOB f2.VOB | ffmpeg -i - out.mp2

x264 lossless

The ultrafast preset will provide the fastest encoding and is useful for quick capturing (such as screencasting):

$ ffmpeg -i input -c:v libx264 -preset ultrafast -qp 0 -c:a copy output

On the opposite end of the preset spectrum is veryslow and will encode slower than ultrafast but provide a smaller output file size:

$ ffmpeg -i input -c:v libx264 -preset veryslow -qp 0 -c:a copy output

Both examples will provide the same quality output.


In encoding x265 files, you may need to specify the aspect ratio of the file via -aspect <width:height>. Example :

 ffmpeg -i input -c:v libx265 -aspect 1920:1080 -preset veryslow -x265-params crf=20 output

Single-pass MPEG-2 (near lossless)

Allow FFmpeg to automatically set DVD standardized parameters. Encode to DVD MPEG-2 at ~30 FPS:

$ ffmpeg -i video.VOB -target ntsc-dvd output.mpg

Encode to DVD MPEG-2 at ~24 FPS:

$ ffmpeg -i video.VOB -target film-dvd output.mpg

x264: constant rate factor

Used when you want a specific quality output. General usage is to use the highest -crf value that still provides an acceptable quality. Lower values are higher quality; 0 is lossless, 18 is visually lossless, and 23 is the default value. A sane range is between 18 and 28. Use the slowest -preset you have patience for. See the x264 Encoding Guide for more information.

$ ffmpeg -i video -c:v libx264 -tune film -preset slow -crf 22 -x264opts fast_pskip=0 -c:a libmp3lame -aq 4 output.mkv

-tune option can be used to match the type and content of the of media being encoded.


FFmpeg is very useful to encode videos and strip their size before you upload them on YouTube. The following single line of code takes an input file and outputs a mkv container.

$ ffmpeg -i video -c:v libx264 -crf 18 -preset slow -pix_fmt yuv420p -c:a copy output.mkv

For more information see the forums. You can also create a shell function ytconvert which takes the name of the input file as first argument and the name of the .mkv container as second argument. To do so add the following to your ~/.bashrc:

ytconvert() {
        ffmpeg -i "$1" -c:v libx264 -crf 18 -preset slow -pix_fmt yuv420p -c:a copy "$2.mkv"

See also Arch Linux forum thread.

Two-pass x264 (very high-quality)

Audio deactivated as only video statistics are recorded during the first of multiple pass runs:

$ ffmpeg -i video.VOB -an -vcodec libx264 -pass 1  -preset veryslow \
-threads 0 -b 3000k -x264opts frameref=15:fast_pskip=0 -f rawvideo -y /dev/null

Container format is automatically detected and muxed into from the output file extenstion (.mkv):

$ ffmpeg -i video.VOB -acodec libvo-aacenc -ab 256k -ar 96000 -vcodec libx264 \
-pass 2 -preset veryslow -threads 0 -b 3000k -x264opts frameref=15:fast_pskip=0 video.mkv
Tip: If you receive Unknown encoder 'libvo-aacenc' error (given the fact that your ffmpeg is compiled with libvo-aacenc enabled), you may want to try -acodec libvo_aacenc, an underscore instead of hyphen.

Two-pass MPEG-4 (very high-quality)

Audio deactivated as only video statistics are logged during the first of multiple pass runs:

$ ffmpeg -i video.VOB -an -vcodec mpeg4 -pass 1 -mbd 2 -trellis 2 -flags +cbp+mv0 \
-pre_dia_size 4 -dia_size 4 -precmp 4 -cmp 4 -subcmp 4 -preme 2 -qns 2 -b 3000k \
-f rawvideo -y /dev/null

Container format is automatically detected and muxed into from the output file extenstion (.avi):

$ ffmpeg -i video.VOB -acodec copy -vcodec mpeg4 -vtag DX50 -pass 2 -mbd 2 -trellis 2 \
-flags +cbp+mv0 -pre_dia_size 4 -dia_size 4 -precmp 4 -cmp 4 -subcmp 4 -preme 2 -qns 2 \
-b 3000k video.avi
  • Introducing threads=n>1 for -vcodec mpeg4 may skew the effects of motion estimation and lead to reduced video quality and compression efficiency.
  • The two-pass MPEG-4 example above also supports output to the MP4 container (replace .avi with .mp4).

Determining bitrates with fixed output file sizes

  • (Desired File Size in MB - Audio File Size in MB) x 8192 kb/MB / Length of Media in Seconds (s) = Bitrate in kb/s
  • (3900 MB - 275 MB) = 3625 MB x 8192 kb/MB / 8830 s = 3363 kb/s required to achieve an approximate total output file size of 3900 MB

x264 video stabilization

Video stablization using the vbid.stab plugin entails two passes.

First pass

The first pass records stabilization parameters to a file and/or a test video for visual analysis.

  • Records stabilization parameters to a file only
$ ffmpeg -i input -vf vidstabdetect=stepsize=4:mincontrast=0:result=transforms.trf -f null -
  • Records stabilization parameters to a file and create test video "output-stab" for visual analysis
$ ffmpeg -i input -vf vidstabdetect=stepsize=4:mincontrast=0:result=transforms.trf -f output-stab

Second pass

The second pass parses the stabilization parameters generated from the first pass and applies them to produce "output-stab_final". You will want to apply any additional filters at this point so as to aboid subsequent transcoding to preserve as much video quality as possible. The following example performs the following in addition to video stabilization:

  • unsharp is recommended by the author of vid.stab. Here we are simply using the defaults of 5:5:1.0:5:5:1.0
  • Tip: fade=t=in:st=0:d=4
    fade in from black starting from the beginning of the file for four seconds
  • Tip: fade=t=out:st=60:d=4
    fade out to black starting from sixty seconds into the video for four seconds
  • -c:a pcm_s16le XAVC-S codec records in pcm_s16be which is losslessly transcoded to pcm_s16le
$  ffmpeg -i input -vf vidstabtransform=smoothing=30:interpol=bicubic:input=transforms.trf,unsharp,fade=t=in:st=0:d=4,fade=t=out:st=60:d=4 -c:v libx264 -tune film -preset veryslow -crf 8 -x264opts fast_pskip=0 -c:a pcm_s16le output-stab_final



Subtitles embedded in container files, such as MPEG-2 and Matroska, can be extracted and converted into SRT, SSA, among other subtitle formats.

  • Inspect a file to determine if it contains a subtitle stream:
$ ffprobe foo.mkv
Stream #0:0(und): Video: h264 (High), yuv420p, 1920x800 [SAR 1:1 DAR 12:5], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
  CREATION_TIME   : 2012-06-05 05:04:15
  LANGUAGE        : und
Stream #0:1(und): Audio: aac, 44100 Hz, stereo, fltp (default)
 CREATION_TIME   : 2012-06-05 05:10:34
 LANGUAGE        : und
 HANDLER_NAME    : GPAC ISO Audio Handler
Stream #0:2: Subtitle: ssa (default)
  • foo.mkv has an embedded SSA subtitle which can be extracted into an independent file:
$ ffmpeg -i foo.mkv foo.ssa


(instructions based on an FFmpeg wiki article)

Hardsubbing entails merging subtitles with the video. Hardsubs can't be disabled, nor language switched.

  • Overlay foo.mpg with the subtitles in foo.ssa:
$ ffmpeg -i foo.mpg -c copy -vf subtitles=foo.ssa out.mpg

Volume gain

Change the audio volume in multiples of 256 where 256 = 100% (normal) volume. Additional values such as 400 are also valid options.

-vol 256  = 100%
-vol 512  = 200%
-vol 768  = 300%
-vol 1024 = 400%
-vol 2048 = 800%

To double the volume (512 = 200%) of an MP3 file:

$ ffmpeg -i file.mp3 -vol 512 louder_file.mp3

To quadruple the volume (1024 = 400%) of an Ogg file:

$ ffmpeg -i file.ogg -vol 1024 louder_file.ogg

Note that gain metadata is only written to the output file. Unlike mp3gain or ogggain, the source sound file is untouched.

Extracting audio

$ ffmpeg -i video.mpg
Input #0, avi, from 'video.mpg':
  Duration: 01:58:28.96, start: 0.000000, bitrate: 3000 kb/s
    Stream #0.0: Video: mpeg4, yuv420p, 720x480 [PAR 1:1 DAR 16:9], 29.97 tbr, 29.97 tbn, 29.97 tbc
    Stream #0.1: Audio: ac3, 48000 Hz, stereo, s16, 384 kb/s
    Stream #0.2: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s
    Stream #0.3: Audio: dts, 48000 Hz, 5.1 768 kb/s

Extract the first (-map 0:1) AC-3 encoded audio stream exactly as it was multiplexed into the file:

$ ffmpeg -i video.mpg -map 0:1 -acodec copy -vn video.ac3

Convert the third (-map 0:3) DTS audio stream to an AAC file with a bitrate of 192 kb/s and a sampling rate of 96000 Hz:

$ ffmpeg -i video.mpg -map 0:3 -acodec libvo-aacenc -ab 192k -ar 96000 -vn output.aac

-vn disables the processing of the video stream.

Extract audio stream with certain time interval:

$ ffmpeg -ss 00:01:25 -t 00:00:05 -i video.mpg -map 0:1 -acodec copy -vn output.ac3

-ss specifies the start point, and -t specifies the duration.

Stripping audio

  1. Copy the first video stream (-map 0:0) along with the second AC-3 audio stream (-map 0:2).
  2. Convert the AC-3 audio stream to two-channel MP3 with a bitrate of 128 kb/s and a sampling rate of 48000 Hz.
$ ffmpeg -i video.mpg -map 0:0 -map 0:2 -vcodec copy -acodec libmp3lame \
-ab 128k -ar 48000 -ac 2 video.mkv
$ ffmpeg -i video.mkv
Input #0, avi, from 'video.mpg':
  Duration: 01:58:28.96, start: 0.000000, bitrate: 3000 kb/s
    Stream #0.0: Video: mpeg4, yuv420p, 720x480 [PAR 1:1 DAR 16:9], 29.97 tbr, 29.97 tbn, 29.97 tbc
    Stream #0.1: Audio: mp3, 48000 Hz, stereo, s16, 128 kb/s
Note: Removing undesired audio streams allows for additional bits to be allocated towards improving video quality.

Splitting files

You can use the copy codec to perform operations on a file without changing the encoding. For example, this allows you to easily split any kind of media file into two

$ ffmpeg -i file.ext -t 00:05:30 -c copy part1.ext -ss 00:05:30 -c copy part2.ext

Preset files

Populate ~/.ffmpeg with the default preset files:

$ cp -iR /usr/share/ffmpeg ~/.ffmpeg

Create new and/or modify the default preset files:


Using preset files

Enable the -vpre option after declaring the desired -vcodec


  • libavcodec = Name of the vcodec/acodec
  • vhq = Name of specific preset to be called out
  • ffpreset = FFmpeg preset filetype suffix
Two-pass MPEG-4 (very high quality)

First pass of a multipass (bitrate) ratecontrol transcode:

$ ffmpeg -i video.mpg -an -vcodec mpeg4 -pass 1 -vpre vhq -f rawvideo -y /dev/null

Ratecontrol based on the video statistics logged from the first pass:

$ ffmpeg -i video.mpg -acodec libvorbis -aq 8 -ar 48000 -vcodec mpeg4 \
-pass 2 -vpre vhq -b 3000k output.mp4
  • libvorbis quality settings (VBR)
  • -aq 4 = 128 kb/s
  • -aq 5 = 160 kb/s
  • -aq 6 = 192 kb/s
  • -aq 7 = 224 kb/s
  • -aq 8 = 256 kb/s

Package removal

pacman will not remove configuration files outside of the defaults that were created during package installation. This includes user-created preset files.


The FFmpeg package includes FFserver, which can be used to stream media over a network. To use it, you first need to create the config file /etc/ffserver.conf to define your feeds and streams. Each feed specifies how the media will be sent to ffserver and each stream specifies how a particular feed will be transcoded for streaming over the network. You can start with the sample configuration file or check the ffserver(1) man page for feed and stream examples. Here is a simple configuration file for streaming flash video:

HTTPPort 8090
MaxHTTPConnections 2000
MaxClients 1000
MaxBandwidth 10000
CustomLog -

<Feed av_feed.ffm>
        File /tmp/av_feed.ffm
        FileMaxSize 1G
        ACL allow

<Stream av_stream.flv>
        Feed av_feed.ffm
        Format flv

        VideoCodec libx264
        VideoFrameRate 25
        VideoSize hd1080
        VideoBitRate 400
        AVOptionVideo qmin 10
        AVOptionVideo qmax 42
        AVOptionVideo flags +global_header

        AudioCodec libmp3lame
        AVOptionAudio flags +global_header

        Preroll 15

<Stream stat.html>
        Format status
        ACL allow localhost
        ACL allow

<Redirect index.html>
        URL http://www.ffmpeg.org/

Once you have created your config file, you can start the server and send media to your feeds. For the previous config example, this would look like

$ ffserver &
$ ffmpeg -i myvideo.mkv http://localhost:8090/av_feed.ffm

You can then stream your media using the URL http://yourserver.net:8090/av_stream.flv.

See also