Difference between revisions of "FFmpeg"

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(Package installation: Mention AUR version with ALL codecs)
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[[pacman|Install]] {{Pkg|ffmpeg}} from the [[official repositories]].
 
[[pacman|Install]] {{Pkg|ffmpeg}} from the [[official repositories]].
  
A drop-in replacement fork called {{AUR|libav}} is available in [[AUR]]. The binary it provides is called '''avconv''' instead of '''ffmpeg'''.
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{{Note|The version in the official repositories does not include all codecs due to license constraints. Notably, the Fraunhofer AAC codec and the AAC+ codec are not included. {{AUR|ffmpeg-full}} from [[AUR]] supplies all codecs.}}
  
Note that the version in the official repositories does not include all codecs due to license constraints. Notably, the Fraunhofer AAC codec and the AAC+ codec are not included. {{AUR|ffmpeg-full}} from [[AUR]] provides a drop-in replacement with all codecs enabled.
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A drop-in replacement fork called {{AUR|libav}} is available in AUR. The binary it provides is called '''avconv''' instead of '''ffmpeg'''.
  
 
== Encoding examples ==
 
== Encoding examples ==

Revision as of 20:34, 22 June 2013

Summary help replacing me
This article attempts to walk users through the installation, usage and configuration of FFmpeg.

From the FFmpeg homepage:

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 ffmpeg from the official repositories.

Note: The version in the official repositories does not include all codecs due to license constraints. Notably, the Fraunhofer AAC codec and the AAC+ codec are not included. ffmpeg-fullAUR from AUR supplies all codecs.

A drop-in replacement fork called libavAUR is available in AUR. The binary it provides is called avconv instead of ffmpeg.

Encoding examples

Screen cast to .webm

Using x11grab to video grab your display and using ALSA for sound. First we create lossless raw file test.mkv.

$ ffmpeg -f x11grab -r 30 -i :0.0 -f alsa -i hw:0,0 -acodec flac -vcodec ffvhuff test.mkv

Then we process this test.mkv file into a smaller test.webm end product. Complex switches like c:a and c:v convert the stream into what's needed for WebM.

$ ffmpeg -y -i test.mkv -c:a libvorbis -q:a 3 -c:v libvpx -b:v 2000k test.webm

See https://github.com/kaihendry/recordmydesktop2.0/blob/master/r2d2.sh for a more fleshed out example.

VOB to any container

Concatenate the desired VOB files into a single VOB file:

$ cat video-1.VOB video-2.VOB video-3.VOB > output.VOB

Concatenate and then pipe the output VOB to FFmpeg to use a different format:

$ cat video-1.VOB video-2.VOB video-3.VOB > output.VOB | ffmpeg -i ...

x264 lossless

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

$ ffmpeg -i input -vcodec libx264 -preset ultrafast -qp 0 -acodec copy output.mkv

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 -vcodec libx264 -preset veryslow -qp 0 -acodec copy output.mkv

Both examples will provide the same quality output.

Single-pass MPEG-2 (near lossless)

Allow FFmpeg to automatically set DVD standardized parameters. Encode to DVD MPEG-2 at a frame rate of 30 frames/second:

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

Encode to DVD MPEG-2 at a frame rate of 24 frames/second:

$ ffmpeg -i video.VOB -target film-dvd -sameq 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. A sane range is 18-28 and 23 is default. 18 is considered to be visually lossless. Use the slowest -preset you have patience for. See the x264 Encoding Guide for more information.

$ ffmpeg -i video -vcodec libx264 -preset slow -crf 22 -acodec libmp3lame -aq 4 output.mkv

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

YouTube

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 custom alias 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:

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

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

Softsubs to hardsubs

If have a softsubbed video (eg. ASS/SSA subs in a mkv container like most anime) you can 'burn' these subs into a new file to be played on a device which does not support subs or is to weak to display complex subs.

  • Recompile ffmpeg with --enable-libass if it is not already enabled in your ffmpeg build. See ABS for easy recompiling.
  • Pull out subs from your file. This command assumes that track #2 is the ASS/SSA track. Use mkvinfo if it is not.
$ mkvextract tracks your file.mkv 2:your file.ass
  • Recode file with ffmpeg. See sections above for suitable options. It is very important to disable sub-recording and enable sub-rendering:
$ ffmpeg ... -sn -vf ass=subtitles.ass

Output is set as *.mp4 since the default Android 4.2 player dislikes *.mkv. (But VLC on Android works with mkv). Example:

$ ffmpeg -i video.mkv -sn -vcodec libx264 -crf 18 -preset slow -vf ass=subtitles.ass -acodec copy output.mp4

Preset files

Creating presets

Populate ~/.ffmpeg with the default preset files:

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

Create new and/or modify the default preset files:

~/.ffmpeg/libavcodec-vhq.ffpreset
vtag=DX50
mbd=2
trellis=2
flags=+cbp+mv0
pre_dia_size=4
dia_size=4
precmp=4
cmp=4
subcmp=4
preme=2
qns=2

Using preset files

Enable the -vpre option after declaring the desired -vcodec

libavcodec-vhq.ffpreset

  • 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

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.

Adding subtitles

FFmpeg does not currently support muxing subtitle files into existing streams. See MEncoder for subtitle muxing support.

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

Package removal

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

See also