Difference between revisions of "FFmpeg"

Revision as of 20:23, 27 November 2013

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

Various flavors and related projects can be installed from the official repositories and the AUR:

ffmpeg – official package

Notable variants:

ffmpeg-git^AUR – development version
ffmpeg-full^AUR – built with as much optional features enabled as possible

Forks:

ffmbc^AUR – targeted for broadcasting usage
libav-git^AUR – the binary it provides is called avconv instead of ffmpeg

Encoding examples

Screen cast

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

First create test.mkv with lossless encoding:

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

then we process it into a smaller Wikipedia:WebM file:

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

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 Wikipedia: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 -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 ~30 FPS:

$ ffmpeg -i video.VOB -target ntsc-dvd -q:a 0 -q:v 0 output.mpg

Encode to DVD MPEG-2 at ~24 FPS:

$ ffmpeg -i video.VOB -target film-dvd -q:a 0 -q:v 0 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 -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

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

Subtitles

Extracting

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)
  Metadata:
  CREATION_TIME   : 2012-06-05 05:04:15
  LANGUAGE        : und
Stream #0:1(und): Audio: aac, 44100 Hz, stereo, fltp (default)
 Metadata:
 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

Hardsubbing

(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

Copy the first video stream (-map 0:0) along with the second AC-3 audio stream (-map 0:2).
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.

Preset files

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

aoTuV is generally preferred over libvorbis provided by Xiph.Org and is provided by libvorbis-aotuv in the AUR.

Package removal

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

@@ Line 24: / Line 24: @@
 === Screen cast  ===
-FFmpeg includes the [http://www.ffmpeg.org/ffmpeg-devices.html#x11grab x11grab] and [[ALSA]] virtual formats that enable capturing the entire user display and audio output.
+FFmpeg includes the [http://www.ffmpeg.org/ffmpeg-devices.html#x11grab x11grab] and [http://www.ffmpeg.org/ffmpeg-devices.html#alsa-1 ALSA] virtual devices that enable capturing the entire user display and audio output.
 First create {{ic|test.mkv}} with lossless encoding:
@@ Line 30: / Line 30: @@
   $ ffmpeg -f x11grab -i $DISPLAY -f alsa -i default -c:v ffvhuff -c:a flac test.mkv
-Then we process it into a smaller [[Wikipedia:WebM]].
+then we process it into a smaller [[Wikipedia:WebM]] file:
   $ ffmpeg -i test.mkv  -c:v libvpx -c:a libvorbis  -b:v 2000k -q:a 3 test.webm