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ffmpeg Documentation
ffmpeg Documentation
Table of Contents
The generic syntax is:
| | ffmpeg [[infile options][‘-i’ infile]]... {[outfile options] outfile}...
|
ffmpeg is a very fast video and audio converter that can also grab from
a live audio/video source. It can also convert between arbitrary sample
rates and resize video on the fly with a high quality polyphase filter.
The command line interface is designed to be intuitive, in the sense
that ffmpeg tries to figure out all parameters that can possibly be
derived automatically. You usually only have to specify the target
bitrate you want.
As a general rule, options are applied to the next specified
file. Therefore, order is important, and you can have the same
option on the command line multiple times. Each occurrence is
then applied to the next input or output file.
-
To set the video bitrate of the output file to 64kbit/s:
| | ffmpeg -i input.avi -b 64k output.avi
|
-
To force the frame rate of the output file to 24 fps:
| | ffmpeg -i input.avi -r 24 output.avi
|
-
To force the frame rate of the input file (valid for raw formats only)
to 1 fps and the frame rate of the output file to 24 fps:
| | ffmpeg -r 1 -i input.m2v -r 24 output.avi
|
The format option may be needed for raw input files.
By default ffmpeg tries to convert as losslessly as possible: It
uses the same audio and video parameters for the outputs as the one
specified for the inputs.
All the numerical options, if not specified otherwise, accept in input
a string representing a number, which may contain one of the
International System number postfixes, for example ’K’, ’M’, ’G’.
If ’i’ is appended after the postfix, powers of 2 are used instead of
powers of 10. The ’B’ postfix multiplies the value for 8, and can be
appended after another postfix or used alone. This allows using for
example ’KB’, ’MiB’, ’G’ and ’B’ as postfix.
Options which do not take arguments are boolean options, and set the
corresponding value to true. They can be set to false by prefixing
with "no" the option name, for example using "-nofoo" in the
commandline will set to false the boolean option with name "foo".
These options are shared amongst the ff* tools.
- ‘-L’
Show license.
- ‘-h, -?, -help, --help’
Show help.
- ‘-version’
Show version.
- ‘-formats’
Show available formats.
The fields preceding the format names have the following meanings:
- ‘D’
Decoding available
- ‘E’
Encoding available
- ‘-codecs’
Show available codecs.
The fields preceding the codec names have the following meanings:
- ‘D’
Decoding available
- ‘E’
Encoding available
- ‘V/A/S’
Video/audio/subtitle codec
- ‘S’
Codec supports slices
- ‘D’
Codec supports direct rendering
- ‘T’
Codec can handle input truncated at random locations instead of only at frame boundaries
- ‘-bsfs’
Show available bitstream filters.
- ‘-protocols’
Show available protocols.
- ‘-filters’
Show available libavfilter filters.
- ‘-pix_fmts’
Show available pixel formats.
- ‘-loglevel loglevel’
Set the logging level used by the library.
loglevel is a number or a string containing one of the following values:
- ‘quiet’
- ‘panic’
- ‘fatal’
- ‘error’
- ‘warning’
- ‘info’
- ‘verbose’
- ‘debug’
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
FFMPEG_FORCE_NOCOLOR or NO_COLOR, or can be forced setting
the environment variable FFMPEG_FORCE_COLOR.
The use of the environment variable NO_COLOR is deprecated and
will be dropped in a following Libav version.
- ‘-f fmt’
Force format.
- ‘-i filename’
input file name
- ‘-y’
Overwrite output files.
- ‘-t duration’
Restrict the transcoded/captured video sequence
to the duration specified in seconds.
hh:mm:ss[.xxx] syntax is also supported.
- ‘-fs limit_size’
Set the file size limit.
- ‘-ss position’
Seek to given time position in seconds.
hh:mm:ss[.xxx] syntax is also supported.
- ‘-itsoffset offset’
Set the input time offset in seconds.
[-]hh:mm:ss[.xxx] syntax is also supported.
This option affects all the input files that follow it.
The offset is added to the timestamps of the input files.
Specifying a positive offset means that the corresponding
streams are delayed by ’offset’ seconds.
- ‘-timestamp time’
Set the recording timestamp in the container.
The syntax for time is:
| | now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH[:MM[:SS[.m...]]])|(HH[MM[SS[.m...]]]))[Z|z])
|
If the value is "now" it takes the current time.
Time is local time unless ’Z’ or ’z’ is appended, in which case it is
interpreted as UTC.
If the year-month-day part is not specified it takes the current
year-month-day.
- ‘-metadata key=value’
Set a metadata key/value pair.
For example, for setting the title in the output file:
| | ffmpeg -i in.avi -metadata title="my title" out.flv
|
- ‘-v number’
Set the logging verbosity level.
- ‘-target type’
Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50", "pal-vcd",
"ntsc-svcd", ... ). All the format options (bitrate, codecs,
buffer sizes) are then set automatically. You can just type:
| | ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg
|
Nevertheless you can specify additional options as long as you know
they do not conflict with the standard, as in:
| | ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
|
- ‘-dframes number’
Set the number of data frames to record.
- ‘-scodec codec’
Force subtitle codec (’copy’ to copy stream).
- ‘-newsubtitle’
Add a new subtitle stream to the current output stream.
- ‘-slang code’
Set the ISO 639 language code (3 letters) of the current subtitle stream.
- ‘-b bitrate’
Set the video bitrate in bit/s (default = 200 kb/s).
- ‘-vframes number’
Set the number of video frames to record.
- ‘-r fps’
Set frame rate (Hz value, fraction or abbreviation), (default = 25).
- ‘-s size’
Set frame size. The format is ‘wxh’ (ffserver default = 160x128, ffmpeg default = same as source).
The following abbreviations are recognized:
- ‘sqcif’
128x96
- ‘qcif’
176x144
- ‘cif’
352x288
- ‘4cif’
704x576
- ‘16cif’
1408x1152
- ‘qqvga’
160x120
- ‘qvga’
320x240
- ‘vga’
640x480
- ‘svga’
800x600
- ‘xga’
1024x768
- ‘uxga’
1600x1200
- ‘qxga’
2048x1536
- ‘sxga’
1280x1024
- ‘qsxga’
2560x2048
- ‘hsxga’
5120x4096
- ‘wvga’
852x480
- ‘wxga’
1366x768
- ‘wsxga’
1600x1024
- ‘wuxga’
1920x1200
- ‘woxga’
2560x1600
- ‘wqsxga’
3200x2048
- ‘wquxga’
3840x2400
- ‘whsxga’
6400x4096
- ‘whuxga’
7680x4800
- ‘cga’
320x200
- ‘ega’
640x350
- ‘hd480’
852x480
- ‘hd720’
1280x720
- ‘hd1080’
1920x1080
- ‘-aspect aspect’
Set the video display aspect ratio specified by aspect.
aspect can be a floating point number string, or a string of the
form num:den, where num and den are the
numerator and denominator of the aspect ratio. For example "4:3",
"16:9", "1.3333", and "1.7777" are valid argument values.
- ‘-croptop size’
- ‘-cropbottom size’
- ‘-cropleft size’
- ‘-cropright size’
All the crop options have been removed. Use -vf
crop=width:height:x:y instead.
- ‘-padtop size’
- ‘-padbottom size’
- ‘-padleft size’
- ‘-padright size’
- ‘-padcolor hex_color’
All the pad options have been removed. Use -vf
pad=width:height:x:y:color instead.
- ‘-vn’
Disable video recording.
- ‘-bt tolerance’
Set video bitrate tolerance (in bits, default 4000k).
Has a minimum value of: (target_bitrate/target_framerate).
In 1-pass mode, bitrate tolerance specifies how far ratecontrol is
willing to deviate from the target average bitrate value. This is
not related to min/max bitrate. Lowering tolerance too much has
an adverse effect on quality.
- ‘-maxrate bitrate’
Set max video bitrate (in bit/s).
Requires -bufsize to be set.
- ‘-minrate bitrate’
Set min video bitrate (in bit/s).
Most useful in setting up a CBR encode:
| | ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
|
It is of little use elsewise.
- ‘-bufsize size’
Set video buffer verifier buffer size (in bits).
- ‘-vcodec codec’
Force video codec to codec. Use the copy special value to
tell that the raw codec data must be copied as is.
- ‘-sameq’
Use same quantizer as source (implies VBR).
- ‘-pass n’
Select the pass number (1 or 2). It is used to do two-pass
video encoding. The statistics of the video are recorded in the first
pass into a log file (see also the option -passlogfile),
and in the second pass that log file is used to generate the video
at the exact requested bitrate.
On pass 1, you may just deactivate audio and set output to null,
examples for Windows and Unix:
| | ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y NUL
ffmpeg -i foo.mov -vcodec libxvid -pass 1 -an -f rawvideo -y /dev/null
|
- ‘-passlogfile prefix’
Set two-pass log file name prefix to prefix, the default file name
prefix is “ffmpeg2pass”. The complete file name will be
‘PREFIX-N.log’, where N is a number specific to the output
stream.
- ‘-newvideo’
Add a new video stream to the current output stream.
- ‘-vlang code’
Set the ISO 639 language code (3 letters) of the current video stream.
- ‘-vf filter_graph’
filter_graph is a description of the filter graph to apply to
the input video.
Use the option "-filters" to show all the available filters (including
also sources and sinks).
- ‘-pix_fmt format’
Set pixel format. Use ’list’ as parameter to show all the supported
pixel formats.
- ‘-sws_flags flags’
Set SwScaler flags.
- ‘-g gop_size’
Set the group of pictures size.
- ‘-intra’
Use only intra frames.
- ‘-vdt n’
Discard threshold.
- ‘-qscale q’
Use fixed video quantizer scale (VBR).
- ‘-qmin q’
minimum video quantizer scale (VBR)
- ‘-qmax q’
maximum video quantizer scale (VBR)
- ‘-qdiff q’
maximum difference between the quantizer scales (VBR)
- ‘-qblur blur’
video quantizer scale blur (VBR) (range 0.0 - 1.0)
- ‘-qcomp compression’
video quantizer scale compression (VBR) (default 0.5).
Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
- ‘-lmin lambda’
minimum video lagrange factor (VBR)
- ‘-lmax lambda’
max video lagrange factor (VBR)
- ‘-mblmin lambda’
minimum macroblock quantizer scale (VBR)
- ‘-mblmax lambda’
maximum macroblock quantizer scale (VBR)
These four options (lmin, lmax, mblmin, mblmax) use ’lambda’ units,
but you may use the QP2LAMBDA constant to easily convert from ’q’ units:
| | ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
|
- ‘-rc_init_cplx complexity’
initial complexity for single pass encoding
- ‘-b_qfactor factor’
qp factor between P- and B-frames
- ‘-i_qfactor factor’
qp factor between P- and I-frames
- ‘-b_qoffset offset’
qp offset between P- and B-frames
- ‘-i_qoffset offset’
qp offset between P- and I-frames
- ‘-rc_eq equation’
Set rate control equation (see section "Expression Evaluation")
(default = tex^qComp).
When computing the rate control equation expression, besides the
standard functions defined in the section "Expression Evaluation", the
following functions are available:
- bits2qp(bits)
- qp2bits(qp)
and the following constants are available:
- iTex
- pTex
- tex
- mv
- fCode
- iCount
- mcVar
- var
- isI
- isP
- isB
- avgQP
- qComp
- avgIITex
- avgPITex
- avgPPTex
- avgBPTex
- avgTex
- ‘-rc_override override’
rate control override for specific intervals
- ‘-me_method method’
Set motion estimation method to method.
Available methods are (from lowest to best quality):
- ‘zero’
Try just the (0, 0) vector.
- ‘phods’
- ‘log’
- ‘x1’
- ‘hex’
- ‘umh’
- ‘epzs’
(default method)
- ‘full’
exhaustive search (slow and marginally better than epzs)
- ‘-dct_algo algo’
Set DCT algorithm to algo. Available values are:
- ‘0’
FF_DCT_AUTO (default)
- ‘1’
FF_DCT_FASTINT
- ‘2’
FF_DCT_INT
- ‘3’
FF_DCT_MMX
- ‘4’
FF_DCT_MLIB
- ‘5’
FF_DCT_ALTIVEC
- ‘-idct_algo algo’
Set IDCT algorithm to algo. Available values are:
- ‘0’
FF_IDCT_AUTO (default)
- ‘1’
FF_IDCT_INT
- ‘2’
FF_IDCT_SIMPLE
- ‘3’
FF_IDCT_SIMPLEMMX
- ‘4’
FF_IDCT_LIBMPEG2MMX
- ‘5’
FF_IDCT_PS2
- ‘6’
FF_IDCT_MLIB
- ‘7’
FF_IDCT_ARM
- ‘8’
FF_IDCT_ALTIVEC
- ‘9’
FF_IDCT_SH4
- ‘10’
FF_IDCT_SIMPLEARM
- ‘-er n’
Set error resilience to n.
- ‘1’
FF_ER_CAREFUL (default)
- ‘2’
FF_ER_COMPLIANT
- ‘3’
FF_ER_AGGRESSIVE
- ‘4’
FF_ER_VERY_AGGRESSIVE
- ‘-ec bit_mask’
Set error concealment to bit_mask. bit_mask is a bit mask of
the following values:
- ‘1’
FF_EC_GUESS_MVS (default = enabled)
- ‘2’
FF_EC_DEBLOCK (default = enabled)
- ‘-bf frames’
Use ’frames’ B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).
- ‘-mbd mode’
macroblock decision
- ‘0’
FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in ffmpeg).
- ‘1’
FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.
- ‘2’
FF_MB_DECISION_RD: rate distortion
- ‘-4mv’
Use four motion vector by macroblock (MPEG-4 only).
- ‘-part’
Use data partitioning (MPEG-4 only).
- ‘-bug param’
Work around encoder bugs that are not auto-detected.
- ‘-strict strictness’
How strictly to follow the standards.
- ‘-aic’
Enable Advanced intra coding (h263+).
- ‘-umv’
Enable Unlimited Motion Vector (h263+)
- ‘-deinterlace’
Deinterlace pictures.
- ‘-ilme’
Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
Use this option if your input file is interlaced and you want
to keep the interlaced format for minimum losses.
The alternative is to deinterlace the input stream with
‘-deinterlace’, but deinterlacing introduces losses.
- ‘-psnr’
Calculate PSNR of compressed frames.
- ‘-vstats’
Dump video coding statistics to ‘vstats_HHMMSS.log’.
- ‘-vstats_file file’
Dump video coding statistics to file.
- ‘-top n’
top=1/bottom=0/auto=-1 field first
- ‘-dc precision’
Intra_dc_precision.
- ‘-vtag fourcc/tag’
Force video tag/fourcc.
- ‘-qphist’
Show QP histogram.
- ‘-vbsf bitstream_filter’
Bitstream filters available are "dump_extra", "remove_extra", "noise", "h264_mp4toannexb", "imxdump", "mjpegadump", "mjpeg2jpeg".
| | ffmpeg -i h264.mp4 -vcodec copy -vbsf h264_mp4toannexb -an out.h264
|
- ‘-force_key_frames time[,time...]’
Force key frames at the specified timestamps, more precisely at the first
frames after each specified time.
This option can be useful to ensure that a seek point is present at a
chapter mark or any other designated place in the output file.
The timestamps must be specified in ascending order.
- ‘-aframes number’
Set the number of audio frames to record.
- ‘-ar freq’
Set the audio sampling frequency. For input streams it is set by
default to 44100 Hz, for output streams it is set by default to the
frequency of the input stream. If the input file has audio streams
with different frequencies, the behaviour is undefined.
- ‘-ab bitrate’
Set the audio bitrate in bit/s (default = 64k).
- ‘-aq q’
Set the audio quality (codec-specific, VBR).
- ‘-ac channels’
Set the number of audio channels. For input streams it is set by
default to 1, for output streams it is set by default to the same
number of audio channels in input. If the input file has audio streams
with different channel count, the behaviour is undefined.
- ‘-an’
Disable audio recording.
- ‘-acodec codec’
Force audio codec to codec. Use the copy special value to
specify that the raw codec data must be copied as is.
- ‘-newaudio’
Add a new audio track to the output file. If you want to specify parameters,
do so before -newaudio (-acodec, -ab, etc..).
Mapping will be done automatically, if the number of output streams is equal to
the number of input streams, else it will pick the first one that matches. You
can override the mapping using -map as usual.
Example:
| | ffmpeg -i file.mpg -vcodec copy -acodec ac3 -ab 384k test.mpg -acodec mp2 -ab 192k -newaudio
|
- ‘-alang code’
Set the ISO 639 language code (3 letters) of the current audio stream.
- ‘-atag fourcc/tag’
Force audio tag/fourcc.
- ‘-audio_service_type type’
Set the type of service that the audio stream contains.
- ‘ma’
Main Audio Service (default)
- ‘ef’
Effects
- ‘vi’
Visually Impaired
- ‘hi’
Hearing Impaired
- ‘di’
Dialogue
- ‘co’
Commentary
- ‘em’
Emergency
- ‘vo’
Voice Over
- ‘ka’
Karaoke
- ‘-absf bitstream_filter’
Bitstream filters available are "dump_extra", "remove_extra", "noise", "mp3comp", "mp3decomp".
- ‘-scodec codec’
Force subtitle codec (’copy’ to copy stream).
- ‘-newsubtitle’
Add a new subtitle stream to the current output stream.
- ‘-slang code’
Set the ISO 639 language code (3 letters) of the current subtitle stream.
- ‘-sn’
Disable subtitle recording.
- ‘-sbsf bitstream_filter’
Bitstream filters available are "mov2textsub", "text2movsub".
| | ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt
|
- ‘-vc channel’
Set video grab channel (DV1394 only).
- ‘-tvstd standard’
Set television standard (NTSC, PAL (SECAM)).
- ‘-isync’
Synchronize read on input.
- ‘-map input_file_id.input_stream_id[:sync_file_id.sync_stream_id]’
-
Designate an input stream as a source for the output file. Each input
stream is identified by the input file index input_file_id and
the input stream index input_stream_id within the input
file. Both indexes start at 0. If specified,
sync_file_id.sync_stream_id sets which input stream
is used as a presentation sync reference.
The -map options must be specified just after the output file.
If any -map options are used, the number of -map options
on the command line must match the number of streams in the output
file. The first -map option on the command line specifies the
source for output stream 0, the second -map option specifies
the source for output stream 1, etc.
For example, if you have two audio streams in the first input file,
these streams are identified by "0.0" and "0.1". You can use
-map to select which stream to place in an output file. For
example:
| | ffmpeg -i INPUT out.wav -map 0.1
|
will map the input stream in ‘INPUT’ identified by "0.1" to
the (single) output stream in ‘out.wav’.
For example, to select the stream with index 2 from input file
‘a.mov’ (specified by the identifier "0.2"), and stream with
index 6 from input ‘b.mov’ (specified by the identifier "1.6"),
and copy them to the output file ‘out.mov’:
| | ffmpeg -i a.mov -i b.mov -vcodec copy -acodec copy out.mov -map 0.2 -map 1.6
|
To add more streams to the output file, you can use the
-newaudio, -newvideo, -newsubtitle options.
- ‘-map_meta_data outfile[,metadata]:infile[,metadata]’
Deprecated, use -map_metadata instead.
- ‘-map_metadata outfile[,metadata]:infile[,metadata]’
Set metadata information of outfile from infile. Note that those
are file indices (zero-based), not filenames.
Optional metadata parameters specify, which metadata to copy - (g)lobal
(i.e. metadata that applies to the whole file), per-(s)tream, per-(c)hapter or
per-(p)rogram. All metadata specifiers other than global must be followed by the
stream/chapter/program number. If metadata specifier is omitted, it defaults to
global.
By default, global metadata is copied from the first input file to all output files,
per-stream and per-chapter metadata is copied along with streams/chapters. These
default mappings are disabled by creating any mapping of the relevant type. A negative
file index can be used to create a dummy mapping that just disables automatic copying.
For example to copy metadata from the first stream of the input file to global metadata
of the output file:
| | ffmpeg -i in.ogg -map_metadata 0:0,s0 out.mp3
|
- ‘-map_chapters outfile:infile’
Copy chapters from infile to outfile. If no chapter mapping is specified,
then chapters are copied from the first input file with at least one chapter to all
output files. Use a negative file index to disable any chapter copying.
- ‘-debug’
Print specific debug info.
- ‘-benchmark’
Show benchmarking information at the end of an encode.
Shows CPU time used and maximum memory consumption.
Maximum memory consumption is not supported on all systems,
it will usually display as 0 if not supported.
- ‘-dump’
Dump each input packet.
- ‘-hex’
When dumping packets, also dump the payload.
- ‘-bitexact’
Only use bit exact algorithms (for codec testing).
- ‘-ps size’
Set RTP payload size in bytes.
- ‘-re’
Read input at native frame rate. Mainly used to simulate a grab device.
- ‘-loop_input’
Loop over the input stream. Currently it works only for image
streams. This option is used for automatic FFserver testing.
- ‘-loop_output number_of_times’
Repeatedly loop output for formats that support looping such as animated GIF
(0 will loop the output infinitely).
- ‘-threads count’
Thread count.
- ‘-vsync parameter’
Video sync method.
- ‘0’
Each frame is passed with its timestamp from the demuxer to the muxer.
- ‘1’
Frames will be duplicated and dropped to achieve exactly the requested
constant framerate.
- ‘2’
Frames are passed through with their timestamp or dropped so as to
prevent 2 frames from having the same timestamp.
- ‘-1’
Chooses between 1 and 2 depending on muxer capabilities. This is the
default method.
With -map you can select from which stream the timestamps should be
taken. You can leave either video or audio unchanged and sync the
remaining stream(s) to the unchanged one.
- ‘-async samples_per_second’
Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
the parameter is the maximum samples per second by which the audio is changed.
-async 1 is a special case where only the start of the audio stream is corrected
without any later correction.
- ‘-copyts’
Copy timestamps from input to output.
- ‘-copytb’
Copy input stream time base from input to output when stream copying.
- ‘-shortest’
Finish encoding when the shortest input stream ends.
- ‘-dts_delta_threshold’
Timestamp discontinuity delta threshold.
- ‘-muxdelay seconds’
Set the maximum demux-decode delay.
- ‘-muxpreload seconds’
Set the initial demux-decode delay.
- ‘-streamid output-stream-index:new-value’
Assign a new stream-id value to an output stream. This option should be
specified prior to the output filename to which it applies.
For the situation where multiple output files exist, a streamid
may be reassigned to a different value.
For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for
an output mpegts file:
| | ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts
|
A preset file contains a sequence of option=value pairs,
one for each line, specifying a sequence of options which would be
awkward to specify on the command line. Lines starting with the hash
(’#’) character are ignored and are used to provide comments. Check
the ‘ffpresets’ directory in the Libav source tree for examples.
Preset files are specified with the vpre, apre,
spre, and fpre options. The fpre option takes the
filename of the preset instead of a preset name as input and can be
used for any kind of codec. For the vpre, apre, and
spre options, the options specified in a preset file are
applied to the currently selected codec of the same type as the preset
option.
The argument passed to the vpre, apre, and spre
preset options identifies the preset file to use according to the
following rules:
First ffmpeg searches for a file named arg.ffpreset in the
directories ‘$FFMPEG_DATADIR’ (if set), and ‘$HOME/.ffmpeg’, and in
the datadir defined at configuration time (usually ‘PREFIX/share/ffmpeg’)
in that order. For example, if the argument is libx264-max, it will
search for the file ‘libx264-max.ffpreset’.
If no such file is found, then ffmpeg will search for a file named
codec_name-arg.ffpreset in the above-mentioned
directories, where codec_name is the name of the codec to which
the preset file options will be applied. For example, if you select
the video codec with -vcodec libx264 and use -vpre max,
then it will search for the file ‘libx264-max.ffpreset’.
-
For streaming at very low bitrate application, use a low frame rate
and a small GOP size. This is especially true for RealVideo where
the Linux player does not seem to be very fast, so it can miss
frames. An example is:
| | ffmpeg -g 3 -r 3 -t 10 -b 50k -s qcif -f rv10 /tmp/b.rm
|
-
The parameter ’q’ which is displayed while encoding is the current
quantizer. The value 1 indicates that a very good quality could
be achieved. The value 31 indicates the worst quality. If q=31 appears
too often, it means that the encoder cannot compress enough to meet
your bitrate. You must either increase the bitrate, decrease the
frame rate or decrease the frame size.
-
If your computer is not fast enough, you can speed up the
compression at the expense of the compression ratio. You can use
’-me zero’ to speed up motion estimation, and ’-intra’ to disable
motion estimation completely (you have only I-frames, which means it
is about as good as JPEG compression).
-
To have very low audio bitrates, reduce the sampling frequency
(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
-
To have a constant quality (but a variable bitrate), use the option
’-qscale n’ when ’n’ is between 1 (excellent quality) and 31 (worst
quality).
-
When converting video files, you can use the ’-sameq’ option which
uses the same quality factor in the encoder as in the decoder.
It allows almost lossless encoding.
If you specify the input format and device then ffmpeg can grab video
and audio directly.
| | ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg
|
Note that you must activate the right video source and channel before
launching ffmpeg with any TV viewer such as xawtv
(http://linux.bytesex.org/xawtv/) by Gerd Knorr. You also
have to set the audio recording levels correctly with a
standard mixer.
Grab the X11 display with ffmpeg via
| | ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg
|
0.0 is display.screen number of your X11 server, same as
the DISPLAY environment variable.
| | ffmpeg -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg
|
0.0 is display.screen number of your X11 server, same as the DISPLAY environment
variable. 10 is the x-offset and 20 the y-offset for the grabbing.
Any supported file format and protocol can serve as input to ffmpeg:
Examples:
-
You can use YUV files as input:
| | ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
|
It will use the files:
| | /tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
|
The Y files use twice the resolution of the U and V files. They are
raw files, without header. They can be generated by all decent video
decoders. You must specify the size of the image with the ‘-s’ option
if ffmpeg cannot guess it.
-
You can input from a raw YUV420P file:
| | ffmpeg -i /tmp/test.yuv /tmp/out.avi
|
test.yuv is a file containing raw YUV planar data. Each frame is composed
of the Y plane followed by the U and V planes at half vertical and
horizontal resolution.
-
You can output to a raw YUV420P file:
| | ffmpeg -i mydivx.avi hugefile.yuv
|
-
You can set several input files and output files:
| | ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
|
Converts the audio file a.wav and the raw YUV video file a.yuv
to MPEG file a.mpg.
-
You can also do audio and video conversions at the same time:
| | ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
|
Converts a.wav to MPEG audio at 22050 Hz sample rate.
-
You can encode to several formats at the same time and define a
mapping from input stream to output streams:
| | ffmpeg -i /tmp/a.wav -ab 64k /tmp/a.mp2 -ab 128k /tmp/b.mp2 -map 0:0 -map 0:0
|
Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. ’-map
file:index’ specifies which input stream is used for each output
stream, in the order of the definition of output streams.
-
You can transcode decrypted VOBs:
| | ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800k -g 300 -bf 2 -acodec libmp3lame -ab 128k snatch.avi
|
This is a typical DVD ripping example; the input is a VOB file, the
output an AVI file with MPEG-4 video and MP3 audio. Note that in this
command we use B-frames so the MPEG-4 stream is DivX5 compatible, and
GOP size is 300 which means one intra frame every 10 seconds for 29.97fps
input video. Furthermore, the audio stream is MP3-encoded so you need
to enable LAME support by passing --enable-libmp3lame to configure.
The mapping is particularly useful for DVD transcoding
to get the desired audio language.
NOTE: To see the supported input formats, use ffmpeg -formats.
-
You can extract images from a video, or create a video from many images:
For extracting images from a video:
| | ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
|
This will extract one video frame per second from the video and will
output them in files named ‘foo-001.jpeg’, ‘foo-002.jpeg’,
etc. Images will be rescaled to fit the new WxH values.
If you want to extract just a limited number of frames, you can use the
above command in combination with the -vframes or -t option, or in
combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
| | ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
|
The syntax foo-%03d.jpeg specifies to use a decimal number
composed of three digits padded with zeroes to express the sequence
number. It is the same syntax supported by the C printf function, but
only formats accepting a normal integer are suitable.
-
You can put many streams of the same type in the output:
| | ffmpeg -i test1.avi -i test2.avi -vcodec copy -acodec copy -vcodec copy -acodec copy test12.avi -newvideo -newaudio
|
In addition to the first video and audio streams, the resulting
output file ‘test12.avi’ will contain the second video
and the second audio stream found in the input streams list.
The -newvideo, -newaudio and -newsubtitle
options have to be specified immediately after the name of the output
file to which you want to add them.
When evaluating an arithemetic expression, Libav uses an internal
formula evaluator, implemented through the ‘libavutil/eval.h’
interface.
An expression may contain unary, binary operators, constants, and
functions.
Two expressions expr1 and expr2 can be combined to form
another expression "expr1;expr2".
expr1 and expr2 are evaluated in turn, and the new
expression evaluates to the value of expr2.
The following binary operators are available: +, -,
*, /, ^.
The following unary operators are available: +, -.
The following functions are available:
- ‘sinh(x)’
- ‘cosh(x)’
- ‘tanh(x)’
- ‘sin(x)’
- ‘cos(x)’
- ‘tan(x)’
- ‘atan(x)’
- ‘asin(x)’
- ‘acos(x)’
- ‘exp(x)’
- ‘log(x)’
- ‘abs(x)’
- ‘squish(x)’
- ‘gauss(x)’
- ‘isnan(x)’
Return 1.0 if x is NAN, 0.0 otherwise.
- ‘mod(x, y)’
- ‘max(x, y)’
- ‘min(x, y)’
- ‘eq(x, y)’
- ‘gte(x, y)’
- ‘gt(x, y)’
- ‘lte(x, y)’
- ‘lt(x, y)’
- ‘st(var, expr)’
Allow to store the value of the expression expr in an internal
variable. var specifies the number of the variable where to
store the value, and it is a value ranging from 0 to 9. The function
returns the value stored in the internal variable.
- ‘ld(var)’
Allow to load the value of the internal variable with number
var, which was previosly stored with st(var, expr).
The function returns the loaded value.
- ‘while(cond, expr)’
Evaluate expression expr while the expression cond is
non-zero, and returns the value of the last expr evaluation, or
NAN if cond was always false.
- ‘ceil(expr)’
Round the value of expression expr upwards to the nearest
integer. For example, "ceil(1.5)" is "2.0".
- ‘floor(expr)’
Round the value of expression expr downwards to the nearest
integer. For example, "floor(-1.5)" is "-2.0".
- ‘trunc(expr)’
Round the value of expression expr towards zero to the nearest
integer. For example, "trunc(-1.5)" is "-1.0".
Note that:
* works like AND
+ works like OR
thus
is equivalent to
When A evaluates to either 1 or 0, that is the same as
In your C code, you can extend the list of unary and binary functions,
and define recognized constants, so that they are available for your
expressions.
The evaluator also recognizes the International System number
postfixes. If ’i’ is appended after the postfix, powers of 2 are used
instead of powers of 10. The ’B’ postfix multiplies the value for 8,
and can be appended after another postfix or used alone. This allows
using for example ’KB’, ’MiB’, ’G’ and ’B’ as postfix.
Follows the list of available International System postfixes, with
indication of the corresponding powers of 10 and of 2.
- ‘y’
-24 / -80
- ‘z’
-21 / -70
- ‘a’
-18 / -60
- ‘f’
-15 / -50
- ‘p’
-12 / -40
- ‘n’
-9 / -30
- ‘u’
-6 / -20
- ‘m’
-3 / -10
- ‘c’
-2
- ‘d’
-1
- ‘h’
2
- ‘k’
3 / 10
- ‘K’
3 / 10
- ‘M’
6 / 20
- ‘G’
9 / 30
- ‘T’
12 / 40
- ‘P’
15 / 40
- ‘E’
18 / 50
- ‘Z’
21 / 60
- ‘Y’
24 / 70
Encoders are configured elements in Libav which allow the encoding of
multimedia streams.
When you configure your Libav build, all the supported native encoders
are enabled by default. Encoders requiring an external library must be enabled
manually via the corresponding --enable-lib option. You can list all
available encoders using the configure option --list-encoders.
You can disable all the encoders with the configure option
--disable-encoders and selectively enable / disable single encoders
with the options --enable-encoder=ENCODER /
--disable-encoder=ENCODER.
The option -codecs of the ff* tools will display the list of
enabled encoders.
A description of some of the currently available audio encoders
follows.
AC-3 audio encoders.
These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as
the undocumented RealAudio 3 (a.k.a. dnet).
The ac3 encoder uses floating-point math, while the ac3_fixed
encoder only uses fixed-point integer math. This does not mean that one is
always faster, just that one or the other may be better suited to a
particular system. The floating-point encoder will generally produce better
quality audio for a given bitrate. The ac3_fixed encoder is not the
default codec for any of the output formats, so it must be specified explicitly
using the option -acodec ac3_fixed in order to use it.
The AC-3 metadata options are used to set parameters that describe the audio,
but in most cases do not affect the audio encoding itself. Some of the options
do directly affect or influence the decoding and playback of the resulting
bitstream, while others are just for informational purposes. A few of the
options will add bits to the output stream that could otherwise be used for
audio data, and will thus affect the quality of the output. Those will be
indicated accordingly with a note in the option list below.
These parameters are described in detail in several publicly-available
documents.
- ‘-per_frame_metadata boolean’
Allow Per-Frame Metadata. Specifies if the encoder should check for changing
metadata for each frame.
- ‘0’
The metadata values set at initialization will be used for every frame in the
stream. (default)
- ‘1’
Metadata values can be changed before encoding each frame.
- ‘-center_mixlev level’
Center Mix Level. The amount of gain the decoder should apply to the center
channel when downmixing to stereo. This field will only be written to the
bitstream if a center channel is present. The value is specified as a scale
factor. There are 3 valid values:
- ‘0.707’
Apply -3dB gain
- ‘0.595’
Apply -4.5dB gain (default)
- ‘0.500’
Apply -6dB gain
- ‘-surround_mixlev level’
Surround Mix Level. The amount of gain the decoder should apply to the surround
channel(s) when downmixing to stereo. This field will only be written to the
bitstream if one or more surround channels are present. The value is specified
as a scale factor. There are 3 valid values:
- ‘0.707’
Apply -3dB gain
- ‘0.500’
Apply -6dB gain (default)
- ‘0.000’
Silence Surround Channel(s)
Audio Production Information is optional information describing the mixing
environment. Either none or both of the fields are written to the bitstream.
- ‘-mixing_level number’
Mixing Level. Specifies peak sound pressure level (SPL) in the production
environment when the mix was mastered. Valid values are 80 to 111, or -1 for
unknown or not indicated. The default value is -1, but that value cannot be
used if the Audio Production Information is written to the bitstream. Therefore,
if the room_type option is not the default value, the mixing_level
option must not be -1.
- ‘-room_type type’
Room Type. Describes the equalization used during the final mixing session at
the studio or on the dubbing stage. A large room is a dubbing stage with the
industry standard X-curve equalization; a small room has flat equalization.
This field will not be written to the bitstream if both the mixing_level
option and the room_type option have the default values.
- ‘0’
- ‘notindicated’
Not Indicated (default)
- ‘1’
- ‘large’
Large Room
- ‘2’
- ‘small’
Small Room
- ‘-copyright boolean’
Copyright Indicator. Specifies whether a copyright exists for this audio.
- ‘0’
- ‘off’
No Copyright Exists (default)
- ‘1’
- ‘on’
Copyright Exists
- ‘-dialnorm value’
Dialogue Normalization. Indicates how far the average dialogue level of the
program is below digital 100% full scale (0 dBFS). This parameter determines a
level shift during audio reproduction that sets the average volume of the
dialogue to a preset level. The goal is to match volume level between program
sources. A value of -31dB will result in no volume level change, relative to
the source volume, during audio reproduction. Valid values are whole numbers in
the range -31 to -1, with -31 being the default.
- ‘-dsur_mode mode’
Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround
(Pro Logic). This field will only be written to the bitstream if the audio
stream is stereo. Using this option does NOT mean the encoder will actually
apply Dolby Surround processing.
- ‘0’
- ‘notindicated’
Not Indicated (default)
- ‘1’
- ‘off’
Not Dolby Surround Encoded
- ‘2’
- ‘on’
Dolby Surround Encoded
- ‘-original boolean’
Original Bit Stream Indicator. Specifies whether this audio is from the
original source and not a copy.
- ‘0’
- ‘off’
Not Original Source
- ‘1’
- ‘on’
Original Source (default)
The extended bitstream options are part of the Alternate Bit Stream Syntax as
specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts.
If any one parameter in a group is specified, all values in that group will be
written to the bitstream. Default values are used for those that are written
but have not been specified. If the mixing levels are written, the decoder
will use these values instead of the ones specified in the center_mixlev
and surround_mixlev options if it supports the Alternate Bit Stream
Syntax.
- ‘-dmix_mode mode’
Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt
(Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.
- ‘0’
- ‘notindicated’
Not Indicated (default)
- ‘1’
- ‘ltrt’
Lt/Rt Downmix Preferred
- ‘2’
- ‘loro’
Lo/Ro Downmix Preferred
- ‘-ltrt_cmixlev level’
Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lt/Rt mode.
- ‘1.414’
Apply +3dB gain
- ‘1.189’
Apply +1.5dB gain
- ‘1.000’
Apply 0dB gain
- ‘0.841’
Apply -1.5dB gain
- ‘0.707’
Apply -3.0dB gain
- ‘0.595’
Apply -4.5dB gain (default)
- ‘0.500’
Apply -6.0dB gain
- ‘0.000’
Silence Center Channel
- ‘-ltrt_surmixlev level’
Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lt/Rt mode.
- ‘0.841’
Apply -1.5dB gain
- ‘0.707’
Apply -3.0dB gain
- ‘0.595’
Apply -4.5dB gain
- ‘0.500’
Apply -6.0dB gain (default)
- ‘0.000’
Silence Surround Channel(s)
- ‘-loro_cmixlev level’
Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lo/Ro mode.
- ‘1.414’
Apply +3dB gain
- ‘1.189’
Apply +1.5dB gain
- ‘1.000’
Apply 0dB gain
- ‘0.841’
Apply -1.5dB gain
- ‘0.707’
Apply -3.0dB gain
- ‘0.595’
Apply -4.5dB gain (default)
- ‘0.500’
Apply -6.0dB gain
- ‘0.000’
Silence Center Channel
- ‘-loro_surmixlev level’
Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lo/Ro mode.
- ‘0.841’
Apply -1.5dB gain
- ‘0.707’
Apply -3.0dB gain
- ‘0.595’
Apply -4.5dB gain
- ‘0.500’
Apply -6.0dB gain (default)
- ‘0.000’
Silence Surround Channel(s)
- ‘-dsurex_mode mode’
Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX
(7.1 matrixed to 5.1). Using this option does NOT mean the encoder will actually
apply Dolby Surround EX processing.
- ‘0’
- ‘notindicated’
Not Indicated (default)
- ‘1’
- ‘on’
Dolby Surround EX On
- ‘2’
- ‘off’
Dolby Surround EX Off
- ‘-dheadphone_mode mode’
Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone
encoding (multi-channel matrixed to 2.0 for use with headphones). Using this
option does NOT mean the encoder will actually apply Dolby Headphone
processing.
- ‘0’
- ‘notindicated’
Not Indicated (default)
- ‘1’
- ‘on’
Dolby Headphone On
- ‘2’
- ‘off’
Dolby Headphone Off
- ‘-ad_conv_type type’
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D
conversion.
- ‘0’
- ‘standard’
Standard A/D Converter (default)
- ‘1’
- ‘hdcd’
HDCD A/D Converter
- ‘-stereo_rematrixing boolean’
Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This
is an optional AC-3 feature that increases quality by selectively encoding
the left/right channels as mid/side. This option is enabled by default, and it
is highly recommended that it be left as enabled except for testing purposes.
Floating-Point-Only AC-3 Encoding Options
These options are only valid for the floating-point encoder and do not exist
for the fixed-point encoder due to the corresponding features not being
implemented in fixed-point.
- ‘-channel_coupling boolean’
Enables/Disables use of channel coupling, which is an optional AC-3 feature
that increases quality by combining high frequency information from multiple
channels into a single channel. The per-channel high frequency information is
sent with less accuracy in both the frequency and time domains. This allows
more bits to be used for lower frequencies while preserving enough information
to reconstruct the high frequencies. This option is enabled by default for the
floating-point encoder and should generally be left as enabled except for
testing purposes or to increase encoding speed.
- ‘-1’
- ‘auto’
Selected by Encoder (default)
- ‘0’
- ‘off’
Disable Channel Coupling
- ‘1’
- ‘on’
Enable Channel Coupling
- ‘-cpl_start_band number’
Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a
value higher than the bandwidth is used, it will be reduced to 1 less than the
coupling end band. If auto is used, the start band will be determined by
the encoder based on the bit rate, sample rate, and channel layout. This option
has no effect if channel coupling is disabled.
- ‘-1’
- ‘auto’
Selected by Encoder (default)
Demuxers are configured elements in Libav which allow to read the
multimedia streams from a particular type of file.
When you configure your Libav build, all the supported demuxers
are enabled by default. You can list all available ones using the
configure option "–list-demuxers".
You can disable all the demuxers using the configure option
"–disable-demuxers", and selectively enable a single demuxer with
the option "–enable-demuxer=DEMUXER", or disable it
with the option "–disable-demuxer=DEMUXER".
The option "-formats" of the ff* tools will display the list of
enabled demuxers.
The description of some of the currently available demuxers follows.
Image file demuxer.
This demuxer reads from a list of image files specified by a pattern.
The pattern may contain the string "%d" or "%0Nd", which
specifies the position of the characters representing a sequential
number in each filename matched by the pattern. If the form
"%d0Nd" is used, the string representing the number in each
filename is 0-padded and N is the total number of 0-padded
digits representing the number. The literal character ’%’ can be
specified in the pattern with the string "%%".
If the pattern contains "%d" or "%0Nd", the first filename of
the file list specified by the pattern must contain a number
inclusively contained between 0 and 4, all the following numbers must
be sequential. This limitation may be hopefully fixed.
The pattern may contain a suffix which is used to automatically
determine the format of the images contained in the files.
For example the pattern "img-%03d.bmp" will match a sequence of
filenames of the form ‘img-001.bmp’, ‘img-002.bmp’, ...,
‘img-010.bmp’, etc.; the pattern "i%%m%%g-%d.jpg" will match a
sequence of filenames of the form ‘i%m%g-1.jpg’,
‘i%m%g-2.jpg’, ..., ‘i%m%g-10.jpg’, etc.
The size, the pixel format, and the format of each image must be the
same for all the files in the sequence.
The following example shows how to use ‘ffmpeg’ for creating a
video from the images in the file sequence ‘img-001.jpeg’,
‘img-002.jpeg’, ..., assuming an input framerate of 10 frames per
second:
| | ffmpeg -r 10 -f image2 -i 'img-%03d.jpeg' out.avi
|
Note that the pattern must not necessarily contain "%d" or
"%0Nd", for example to convert a single image file
‘img.jpeg’ you can employ the command:
| | ffmpeg -f image2 -i img.jpeg img.png
|
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing ’a’ or ’v’ in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
Muxers are configured elements in Libav which allow writing
multimedia streams to a particular type of file.
When you configure your Libav build, all the supported muxers
are enabled by default. You can list all available muxers using the
configure option --list-muxers.
You can disable all the muxers with the configure option
--disable-muxers and selectively enable / disable single muxers
with the options --enable-muxer=MUXER /
--disable-muxer=MUXER.
The option -formats of the ff* tools will display the list of
enabled muxers.
A description of some of the currently available muxers follows.
CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC of all the input audio
and video frames. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
CRC.
The output of the muxer consists of a single line of the form:
CRC=0xCRC, where CRC is a hexadecimal number 0-padded to
8 digits containing the CRC for all the decoded input frames.
For example to compute the CRC of the input, and store it in the file
‘out.crc’:
| | ffmpeg -i INPUT -f crc out.crc
|
You can print the CRC to stdout with the command:
You can select the output format of each frame with ‘ffmpeg’ by
specifying the audio and video codec and format. For example to
compute the CRC of the input audio converted to PCM unsigned 8-bit
and the input video converted to MPEG-2 video, use the command:
| | ffmpeg -i INPUT -acodec pcm_u8 -vcodec mpeg2video -f crc -
|
See also the framecrc muxer (see framecrc).
Per-frame CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC for each decoded audio
and video frame. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
CRC.
The output of the muxer consists of a line for each audio and video
frame of the form: stream_index, frame_dts,
frame_size, 0xCRC, where CRC is a hexadecimal
number 0-padded to 8 digits containing the CRC of the decoded frame.
For example to compute the CRC of each decoded frame in the input, and
store it in the file ‘out.crc’:
| | ffmpeg -i INPUT -f framecrc out.crc
|
You can print the CRC of each decoded frame to stdout with the command:
| | ffmpeg -i INPUT -f framecrc -
|
You can select the output format of each frame with ‘ffmpeg’ by
specifying the audio and video codec and format. For example, to
compute the CRC of each decoded input audio frame converted to PCM
unsigned 8-bit and of each decoded input video frame converted to
MPEG-2 video, use the command:
| | ffmpeg -i INPUT -acodec pcm_u8 -vcodec mpeg2video -f framecrc -
|
See also the crc muxer (see crc).
Image file muxer.
The image file muxer writes video frames to image files.
The output filenames are specified by a pattern, which can be used to
produce sequentially numbered series of files.
The pattern may contain the string "%d" or "%0Nd", this string
specifies the position of the characters representing a numbering in
the filenames. If the form "%0Nd" is used, the string
representing the number in each filename is 0-padded to N
digits. The literal character ’%’ can be specified in the pattern with
the string "%%".
If the pattern contains "%d" or "%0Nd", the first filename of
the file list specified will contain the number 1, all the following
numbers will be sequential.
The pattern may contain a suffix which is used to automatically
determine the format of the image files to write.
For example the pattern "img-%03d.bmp" will specify a sequence of
filenames of the form ‘img-001.bmp’, ‘img-002.bmp’, ...,
‘img-010.bmp’, etc.
The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
form ‘img%-1.jpg’, ‘img%-2.jpg’, ..., ‘img%-10.jpg’,
etc.
The following example shows how to use ‘ffmpeg’ for creating a
sequence of files ‘img-001.jpeg’, ‘img-002.jpeg’, ...,
taking one image every second from the input video:
| | ffmpeg -i in.avi -r 1 -f image2 'img-%03d.jpeg'
|
Note that with ‘ffmpeg’, if the format is not specified with the
-f option and the output filename specifies an image file
format, the image2 muxer is automatically selected, so the previous
command can be written as:
| | ffmpeg -i in.avi -r 1 'img-%03d.jpeg'
|
Note also that the pattern must not necessarily contain "%d" or
"%0Nd", for example to create a single image file
‘img.jpeg’ from the input video you can employ the command:
| | ffmpeg -i in.avi -f image2 -vframes 1 img.jpeg
|
MPEG transport stream muxer.
This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
The muxer options are:
- ‘-mpegts_original_network_id number’
Set the original_network_id (default 0x0001). This is unique identifier
of a network in DVB. Its main use is in the unique identification of a
service through the path Original_Network_ID, Transport_Stream_ID.
- ‘-mpegts_transport_stream_id number’
Set the transport_stream_id (default 0x0001). This identifies a
transponder in DVB.
- ‘-mpegts_service_id number’
Set the service_id (default 0x0001) also known as program in DVB.
- ‘-mpegts_pmt_start_pid number’
Set the first PID for PMT (default 0x1000, max 0x1f00).
- ‘-mpegts_start_pid number’
Set the first PID for data packets (default 0x0100, max 0x0f00).
The recognized metadata settings in mpegts muxer are service_provider
and service_name. If they are not set the default for
service_provider is "Libav" and the default for
service_name is "Service01".
| | ffmpeg -i file.mpg -acodec copy -vcodec copy \
-mpegts_original_network_id 0x1122 \
-mpegts_transport_stream_id 0x3344 \
-mpegts_service_id 0x5566 \
-mpegts_pmt_start_pid 0x1500 \
-mpegts_start_pid 0x150 \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
-y out.ts
|
Null muxer.
This muxer does not generate any output file, it is mainly useful for
testing or benchmarking purposes.
For example to benchmark decoding with ‘ffmpeg’ you can use the
command:
| | ffmpeg -benchmark -i INPUT -f null out.null
|
Note that the above command does not read or write the ‘out.null’
file, but specifying the output file is required by the ‘ffmpeg’
syntax.
Alternatively you can write the command as:
| | ffmpeg -benchmark -i INPUT -f null -
|
Matroska container muxer.
This muxer implements the matroska and webm container specs.
The recognized metadata settings in this muxer are:
- ‘title=title name’
Name provided to a single track
- ‘language=language name’
Specifies the language of the track in the Matroska languages form
- ‘STEREO_MODE=mode’
Stereo 3D video layout of two views in a single video track
- ‘mono’
video is not stereo
- ‘left_right’
Both views are arranged side by side, Left-eye view is on the left
- ‘bottom_top’
Both views are arranged in top-bottom orientation, Left-eye view is at bottom
- ‘top_bottom’
Both views are arranged in top-bottom orientation, Left-eye view is on top
- ‘checkerboard_rl’
Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
- ‘checkerboard_lr’
Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
- ‘row_interleaved_rl’
Each view is constituted by a row based interleaving, Right-eye view is first row
- ‘row_interleaved_lr’
Each view is constituted by a row based interleaving, Left-eye view is first row
- ‘col_interleaved_rl’
Both views are arranged in a column based interleaving manner, Right-eye view is first column
- ‘col_interleaved_lr’
Both views are arranged in a column based interleaving manner, Left-eye view is first column
- ‘anaglyph_cyan_red’
All frames are in anaglyph format viewable through red-cyan filters
- ‘right_left’
Both views are arranged side by side, Right-eye view is on the left
- ‘anaglyph_green_magenta’
All frames are in anaglyph format viewable through green-magenta filters
- ‘block_lr’
Both eyes laced in one Block, Left-eye view is first
- ‘block_rl’
Both eyes laced in one Block, Right-eye view is first
For example a 3D WebM clip can be created using the following command line:
| | ffmpeg -i sample_left_right_clip.mpg -an -vcodec libvpx -metadata STEREO_MODE=left_right -y stereo_clip.webm
|
Input devices are configured elements in Libav which allow to access
the data coming from a multimedia device attached to your system.
When you configure your Libav build, all the supported input devices
are enabled by default. You can list all available ones using the
configure option "–list-indevs".
You can disable all the input devices using the configure option
"–disable-indevs", and selectively enable an input device using the
option "–enable-indev=INDEV", or you can disable a particular
input device using the option "–disable-indev=INDEV".
The option "-formats" of the ff* tools will display the list of
supported input devices (amongst the demuxers).
A description of the currently available input devices follows.
ALSA (Advanced Linux Sound Architecture) input device.
To enable this input device during configuration you need libasound
installed on your system.
This device allows capturing from an ALSA device. The name of the
device to capture has to be an ALSA card identifier.
An ALSA identifier has the syntax:
where the DEV and SUBDEV components are optional.
The three arguments (in order: CARD,DEV,SUBDEV)
specify card number or identifier, device number and subdevice number
(-1 means any).
To see the list of cards currently recognized by your system check the
files ‘/proc/asound/cards’ and ‘/proc/asound/devices’.
For example to capture with ‘ffmpeg’ from an ALSA device with
card id 0, you may run the command:
| | ffmpeg -f alsa -i hw:0 alsaout.wav
|
For more information see:
http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html
BSD video input device.
Linux DV 1394 input device.
Linux framebuffer input device.
The Linux framebuffer is a graphic hardware-independent abstraction
layer to show graphics on a computer monitor, typically on the
console. It is accessed through a file device node, usually
‘/dev/fb0’.
For more detailed information read the file
Documentation/fb/framebuffer.txt included in the Linux source tree.
To record from the framebuffer device ‘/dev/fb0’ with
‘ffmpeg’:
| | ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi
|
You can take a single screenshot image with the command:
| | ffmpeg -f fbdev -vframes 1 -r 1 -i /dev/fb0 screenshot.jpeg
|
See also http://linux-fbdev.sourceforge.net/, and fbset(1).
JACK input device.
To enable this input device during configuration you need libjack
installed on your system.
A JACK input device creates one or more JACK writable clients, one for
each audio channel, with name client_name:input_N, where
client_name is the name provided by the application, and N
is a number which identifies the channel.
Each writable client will send the acquired data to the Libav input
device.
Once you have created one or more JACK readable clients, you need to
connect them to one or more JACK writable clients.
To connect or disconnect JACK clients you can use the
‘jack_connect’ and ‘jack_disconnect’ programs, or do it
through a graphical interface, for example with ‘qjackctl’.
To list the JACK clients and their properties you can invoke the command
‘jack_lsp’.
Follows an example which shows how to capture a JACK readable client
with ‘ffmpeg’.
| | # Create a JACK writable client with name "ffmpeg".
$ ffmpeg -f jack -i ffmpeg -y out.wav
# Start the sample jack_metro readable client.
$ jack_metro -b 120 -d 0.2 -f 4000
# List the current JACK clients.
$ jack_lsp -c
system:capture_1
system:capture_2
system:playback_1
system:playback_2
ffmpeg:input_1
metro:120_bpm
# Connect metro to the ffmpeg writable client.
$ jack_connect metro:120_bpm ffmpeg:input_1
|
For more information read:
http://jackaudio.org/
IIDC1394 input device, based on libdc1394 and libraw1394.
Open Sound System input device.
The filename to provide to the input device is the device node
representing the OSS input device, and is usually set to
‘/dev/dsp’.
For example to grab from ‘/dev/dsp’ using ‘ffmpeg’ use the
command:
| | ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
|
For more information about OSS see:
http://manuals.opensound.com/usersguide/dsp.html
sndio input device.
To enable this input device during configuration you need libsndio
installed on your system.
The filename to provide to the input device is the device node
representing the sndio input device, and is usually set to
‘/dev/audio0’.
For example to grab from ‘/dev/audio0’ using ‘ffmpeg’ use the
command:
| | ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
|
Video4Linux and Video4Linux2 input video devices.
The name of the device to grab is a file device node, usually Linux
systems tend to automatically create such nodes when the device
(e.g. an USB webcam) is plugged into the system, and has a name of the
kind ‘/dev/videoN’, where N is a number associated to
the device.
Video4Linux and Video4Linux2 devices only support a limited set of
widthxheight sizes and framerates. You can check which are
supported for example with the command ‘dov4l’ for Video4Linux
devices and the command ‘v4l-info’ for Video4Linux2 devices.
If the size for the device is set to 0x0, the input device will
try to autodetect the size to use.
Only for the video4linux2 device, if the frame rate is set to 0/0 the
input device will use the frame rate value already set in the driver.
Video4Linux support is deprecated since Linux 2.6.30, and will be
dropped in later versions.
Follow some usage examples of the video4linux devices with the ff*
tools.
| | # Grab and show the input of a video4linux device, frame rate is set
# to the default of 25/1.
ffplay -s 320x240 -f video4linux /dev/video0
# Grab and show the input of a video4linux2 device, autoadjust size.
ffplay -f video4linux2 /dev/video0
# Grab and record the input of a video4linux2 device, autoadjust size,
# frame rate value defaults to 0/0 so it is read from the video4linux2
# driver.
ffmpeg -f video4linux2 -i /dev/video0 out.mpeg
|
VfW (Video for Windows) capture input device.
The filename passed as input is the capture driver number, ranging from
0 to 9. You may use "list" as filename to print a list of drivers. Any
other filename will be interpreted as device number 0.
X11 video input device.
This device allows to capture a region of an X11 display.
The filename passed as input has the syntax:
| | [hostname]:display_number.screen_number[+x_offset,y_offset]
|
hostname:display_number.screen_number specifies the
X11 display name of the screen to grab from. hostname can be
ommitted, and defaults to "localhost". The environment variable
DISPLAY contains the default display name.
x_offset and y_offset specify the offsets of the grabbed
area with respect to the top-left border of the X11 screen. They
default to 0.
Check the X11 documentation (e.g. man X) for more detailed information.
Use the ‘dpyinfo’ program for getting basic information about the
properties of your X11 display (e.g. grep for "name" or "dimensions").
For example to grab from ‘:0.0’ using ‘ffmpeg’:
| | ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
# Grab at position 10,20.
ffmpeg -f x11grab -25 -s cif -i :0.0+10,20 out.mpg
|
Output devices are configured elements in Libav which allow to write
multimedia data to an output device attached to your system.
When you configure your Libav build, all the supported output devices
are enabled by default. You can list all available ones using the
configure option "–list-outdevs".
You can disable all the output devices using the configure option
"–disable-outdevs", and selectively enable an output device using the
option "–enable-outdev=OUTDEV", or you can disable a particular
input device using the option "–disable-outdev=OUTDEV".
The option "-formats" of the ff* tools will display the list of
enabled output devices (amongst the muxers).
A description of the currently available output devices follows.
ALSA (Advanced Linux Sound Architecture) output device.
OSS (Open Sound System) output device.
sndio audio output device.
Protocols are configured elements in Libav which allow to access
resources which require the use of a particular protocol.
When you configure your Libav build, all the supported protocols are
enabled by default. You can list all available ones using the
configure option "–list-protocols".
You can disable all the protocols using the configure option
"–disable-protocols", and selectively enable a protocol using the
option "–enable-protocol=PROTOCOL", or you can disable a
particular protocol using the option
"–disable-protocol=PROTOCOL".
The option "-protocols" of the ff* tools will display the list of
supported protocols.
A description of the currently available protocols follows.
Read Apple HTTP Live Streaming compliant segmented stream as
a uniform one. The M3U8 playlists describing the segments can be
remote HTTP resources or local files, accessed using the standard
file protocol.
HTTP is default, specific protocol can be declared by specifying
"+proto" after the applehttp URI scheme name, where proto
is either "file" or "http".
| | applehttp://host/path/to/remote/resource.m3u8
applehttp+http://host/path/to/remote/resource.m3u8
applehttp+file://path/to/local/resource.m3u8
|
Physical concatenation protocol.
Allow to read and seek from many resource in sequence as if they were
a unique resource.
A URL accepted by this protocol has the syntax:
| | concat:URL1|URL2|...|URLN
|
where URL1, URL2, ..., URLN are the urls of the
resource to be concatenated, each one possibly specifying a distinct
protocol.
For example to read a sequence of files ‘split1.mpeg’,
‘split2.mpeg’, ‘split3.mpeg’ with ‘ffplay’ use the
command:
| | ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
|
Note that you may need to escape the character "|" which is special for
many shells.
File access protocol.
Allow to read from or read to a file.
For example to read from a file ‘input.mpeg’ with ‘ffmpeg’
use the command:
| | ffmpeg -i file:input.mpeg output.mpeg
|
The ff* tools default to the file protocol, that is a resource
specified with the name "FILE.mpeg" is interpreted as the URL
"file:FILE.mpeg".
Gopher protocol.
HTTP (Hyper Text Transfer Protocol).
MMS (Microsoft Media Server) protocol over TCP.
MMS (Microsoft Media Server) protocol over HTTP.
The required syntax is:
| | mmsh://server[:port][/app][/playpath]
|
MD5 output protocol.
Computes the MD5 hash of the data to be written, and on close writes
this to the designated output or stdout if none is specified. It can
be used to test muxers without writing an actual file.
Some examples follow.
| | # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
ffmpeg -i input.flv -f avi -y md5:output.avi.md5
# Write the MD5 hash of the encoded AVI file to stdout.
ffmpeg -i input.flv -f avi -y md5:
|
Note that some formats (typically MOV) require the output protocol to
be seekable, so they will fail with the MD5 output protocol.
UNIX pipe access protocol.
Allow to read and write from UNIX pipes.
The accepted syntax is:
number is the number corresponding to the file descriptor of the
pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number
is not specified, by default the stdout file descriptor will be used
for writing, stdin for reading.
For example to read from stdin with ‘ffmpeg’:
| | cat test.wav | ffmpeg -i pipe:0
# ...this is the same as...
cat test.wav | ffmpeg -i pipe:
|
For writing to stdout with ‘ffmpeg’:
| | ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
# ...this is the same as...
ffmpeg -i test.wav -f avi pipe: | cat > test.avi
|
Note that some formats (typically MOV), require the output protocol to
be seekable, so they will fail with the pipe output protocol.
Real-Time Messaging Protocol.
The Real-Time Messaging Protocol (RTMP) is used for streaming multimeâ€
dia content across a TCP/IP network.
The required syntax is:
| | rtmp://server[:port][/app][/playpath]
|
The accepted parameters are:
- ‘server’
The address of the RTMP server.
- ‘port’
The number of the TCP port to use (by default is 1935).
- ‘app’
It is the name of the application to access. It usually corresponds to
the path where the application is installed on the RTMP server
(e.g. ‘/ondemand/’, ‘/flash/live/’, etc.).
- ‘playpath’
It is the path or name of the resource to play with reference to the
application specified in app, may be prefixed by "mp4:".
For example to read with ‘ffplay’ a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
| | ffplay rtmp://myserver/vod/sample
|
Real-Time Messaging Protocol and its variants supported through
librtmp.
Requires the presence of the librtmp headers and library during
configuration. You need to explicitely configure the build with
"–enable-librtmp". If enabled this will replace the native RTMP
protocol.
This protocol provides most client functions and a few server
functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
variants of these encrypted types (RTMPTE, RTMPTS).
The required syntax is:
| | rtmp_proto://server[:port][/app][/playpath] options
|
where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe",
"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
server, port, app and playpath have the same
meaning as specified for the RTMP native protocol.
options contains a list of space-separated options of the form
key=val.
See the librtmp manual page (man 3 librtmp) for more information.
For example, to stream a file in real-time to an RTMP server using
‘ffmpeg’:
| | ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
|
To play the same stream using ‘ffplay’:
| | ffplay "rtmp://myserver/live/mystream live=1"
|
Real-Time Protocol.
RTSP is not technically a protocol handler in libavformat, it is a demuxer
and muxer. The demuxer supports both normal RTSP (with data transferred
over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
data transferred over RDT).
The muxer can be used to send a stream using RTSP ANNOUNCE to a server
supporting it (currently Darwin Streaming Server and Mischa Spiegelmock’s
RTSP server, http://github.com/revmischa/rtsp-server).
The required syntax for a RTSP url is:
| | rtsp://hostname[:port]/path[?options]
|
options is a &-separated list. The following options
are supported:
- ‘udp’
Use UDP as lower transport protocol.
- ‘tcp’
Use TCP (interleaving within the RTSP control channel) as lower
transport protocol.
- ‘multicast’
Use UDP multicast as lower transport protocol.
- ‘http’
Use HTTP tunneling as lower transport protocol, which is useful for
passing proxies.
- ‘filter_src’
Accept packets only from negotiated peer address and port.
Multiple lower transport protocols may be specified, in that case they are
tried one at a time (if the setup of one fails, the next one is tried).
For the muxer, only the tcp and udp options are supported.
When receiving data over UDP, the demuxer tries to reorder received packets
(since they may arrive out of order, or packets may get lost totally). In
order for this to be enabled, a maximum delay must be specified in the
max_delay field of AVFormatContext.
When watching multi-bitrate Real-RTSP streams with ‘ffplay’, the
streams to display can be chosen with -vst n and
-ast n for video and audio respectively, and can be switched
on the fly by pressing v and a.
Example command lines:
To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
| | ffplay -max_delay 500000 rtsp://server/video.mp4?udp
|
To watch a stream tunneled over HTTP:
| | ffplay rtsp://server/video.mp4?http
|
To send a stream in realtime to a RTSP server, for others to watch:
| | ffmpeg -re -i input -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
|
Session Announcement Protocol (RFC 2974). This is not technically a
protocol handler in libavformat, it is a muxer and demuxer.
It is used for signalling of RTP streams, by announcing the SDP for the
streams regularly on a separate port.
The syntax for a SAP url given to the muxer is:
| | sap://destination[:port][?options]
|
The RTP packets are sent to destination on port port,
or to port 5004 if no port is specified.
options is a &-separated list. The following options
are supported:
- ‘announce_addr=address’
Specify the destination IP address for sending the announcements to.
If omitted, the announcements are sent to the commonly used SAP
announcement multicast address 224.2.127.254 (sap.mcast.net), or
ff0e::2:7ffe if destination is an IPv6 address.
- ‘announce_port=port’
Specify the port to send the announcements on, defaults to
9875 if not specified.
- ‘ttl=ttl’
Specify the time to live value for the announcements and RTP packets,
defaults to 255.
- ‘same_port=0|1’
If set to 1, send all RTP streams on the same port pair. If zero (the
default), all streams are sent on unique ports, with each stream on a
port 2 numbers higher than the previous.
VLC/Live555 requires this to be set to 1, to be able to receive the stream.
The RTP stack in libavformat for receiving requires all streams to be sent
on unique ports.
Example command lines follow.
To broadcast a stream on the local subnet, for watching in VLC:
| | ffmpeg -re -i input -f sap sap://224.0.0.255?same_port=1
|
Similarly, for watching in ffplay:
| | ffmpeg -re -i input -f sap sap://224.0.0.255
|
And for watching in ffplay, over IPv6:
| | ffmpeg -re -i input -f sap sap://[ff0e::1:2:3:4]
|
The syntax for a SAP url given to the demuxer is:
address is the multicast address to listen for announcements on,
if omitted, the default 224.2.127.254 (sap.mcast.net) is used. port
is the port that is listened on, 9875 if omitted.
The demuxers listens for announcements on the given address and port.
Once an announcement is received, it tries to receive that particular stream.
Example command lines follow.
To play back the first stream announced on the normal SAP multicast address:
To play back the first stream announced on one the default IPv6 SAP multicast address:
| | ffplay sap://[ff0e::2:7ffe]
|
Trasmission Control Protocol.
The required syntax for a TCP url is:
| | tcp://hostname:port[?options]
|
- ‘listen’
Listen for an incoming connection
| | ffmpeg -i input -f format tcp://hostname:port?listen
ffplay tcp://hostname:port
|
User Datagram Protocol.
The required syntax for a UDP url is:
| | udp://hostname:port[?options]
|
options contains a list of &-seperated options of the form key=val.
Follow the list of supported options.
- ‘buffer_size=size’
set the UDP buffer size in bytes
- ‘localport=port’
override the local UDP port to bind with
- ‘pkt_size=size’
set the size in bytes of UDP packets
- ‘reuse=1|0’
explicitly allow or disallow reusing UDP sockets
- ‘ttl=ttl’
set the time to live value (for multicast only)
- ‘connect=1|0’
Initialize the UDP socket with connect(). In this case, the
destination address can’t be changed with ff_udp_set_remote_url later.
If the destination address isn’t known at the start, this option can
be specified in ff_udp_set_remote_url, too.
This allows finding out the source address for the packets with getsockname,
and makes writes return with AVERROR(ECONNREFUSED) if "destination
unreachable" is received.
For receiving, this gives the benefit of only receiving packets from
the specified peer address/port.
Some usage examples of the udp protocol with ‘ffmpeg’ follow.
To stream over UDP to a remote endpoint:
| | ffmpeg -i input -f format udp://hostname:port
|
To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
| | ffmpeg -i input -f mpegts udp://hostname:port?pkt_size=188&buffer_size=65535
|
To receive over UDP from a remote endpoint:
| | ffmpeg -i udp://[multicast-address]:port
|
When you configure your Libav build, all the supported bitstream
filters are enabled by default. You can list all available ones using
the configure option --list-bsfs.
You can disable all the bitstream filters using the configure option
--disable-bsfs, and selectively enable any bitstream filter using
the option --enable-bsf=BSF, or you can disable a particular
bitstream filter using the option --disable-bsf=BSF.
The option -bsfs of the ff* tools will display the list of
all the supported bitstream filters included in your build.
Below is a description of the currently available bitstream filters.
Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
MJPEG is a video codec wherein each video frame is essentially a
JPEG image. The individual frames can be extracted without loss,
e.g. by
| | ffmpeg -i ../some_mjpeg.avi -vcodec copy frames_%d.jpg
|
Unfortunately, these chunks are incomplete JPEG images, because
they lack the DHT segment required for decoding. Quoting from
http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml:
Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
commented that "MJPEG, or at least the MJPEG in AVIs having the
MJPG fourcc, is restricted JPEG with a fixed – and *omitted* –
Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2,
and it must use basic Huffman encoding, not arithmetic or
progressive. . . . You can indeed extract the MJPEG frames and
decode them with a regular JPEG decoder, but you have to prepend
the DHT segment to them, or else the decoder won’t have any idea
how to decompress the data. The exact table necessary is given in
the OpenDML spec."
This bitstream filter patches the header of frames extracted from an MJPEG
stream (carrying the AVI1 header ID and lacking a DHT segment) to
produce fully qualified JPEG images.
| | ffmpeg -i mjpeg-movie.avi -vcodec copy -vbsf mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -vcodec copy rotated.avi
|
A filtergraph is a directed graph of connected filters. It can contain
cycles, and there can be multiple links between a pair of
filters. Each link has one input pad on one side connecting it to one
filter from which it takes its input, and one output pad on the other
side connecting it to the one filter accepting its output.
Each filter in a filtergraph is an instance of a filter class
registered in the application, which defines the features and the
number of input and output pads of the filter.
A filter with no input pads is called a "source", a filter with no
output pads is called a "sink".
A filtergraph can be represented using a textual representation, which
is recognized by the -vf and -af options of the ff*
tools, and by the av_parse_graph() function defined in
‘libavfilter/avfiltergraph’.
A filterchain consists of a sequence of connected filters, each one
connected to the previous one in the sequence. A filterchain is
represented by a list of ","-separated filter descriptions.
A filtergraph consists of a sequence of filterchains. A sequence of
filterchains is represented by a list of ";"-separated filterchain
descriptions.
A filter is represented by a string of the form:
[in_link_1]...[in_link_N]filter_name=arguments[out_link_1]...[out_link_M]
filter_name is the name of the filter class of which the
described filter is an instance of, and has to be the name of one of
the filter classes registered in the program.
The name of the filter class is optionally followed by a string
"=arguments".
arguments is a string which contains the parameters used to
initialize the filter instance, and are described in the filter
descriptions below.
The list of arguments can be quoted using the character "’" as initial
and ending mark, and the character ’\’ for escaping the characters
within the quoted text; otherwise the argument string is considered
terminated when the next special character (belonging to the set
"[]=;,") is encountered.
The name and arguments of the filter are optionally preceded and
followed by a list of link labels.
A link label allows to name a link and associate it to a filter output
or input pad. The preceding labels in_link_1
... in_link_N, are associated to the filter input pads,
the following labels out_link_1 ... out_link_M, are
associated to the output pads.
When two link labels with the same name are found in the
filtergraph, a link between the corresponding input and output pad is
created.
If an output pad is not labelled, it is linked by default to the first
unlabelled input pad of the next filter in the filterchain.
For example in the filterchain:
| | nullsrc, split[L1], [L2]overlay, nullsink
|
the split filter instance has two output pads, and the overlay filter
instance two input pads. The first output pad of split is labelled
"L1", the first input pad of overlay is labelled "L2", and the second
output pad of split is linked to the second input pad of overlay,
which are both unlabelled.
In a complete filterchain all the unlabelled filter input and output
pads must be connected. A filtergraph is considered valid if all the
filter input and output pads of all the filterchains are connected.
Follows a BNF description for the filtergraph syntax:
| | NAME ::= sequence of alphanumeric characters and '_'
LINKLABEL ::= "[" NAME "]"
LINKLABELS ::= LINKLABEL [LINKLABELS]
FILTER_ARGUMENTS ::= sequence of chars (eventually quoted)
FILTER ::= [LINKNAMES] NAME ["=" ARGUMENTS] [LINKNAMES]
FILTERCHAIN ::= FILTER [,FILTERCHAIN]
FILTERGRAPH ::= FILTERCHAIN [;FILTERGRAPH]
|
When you configure your Libav build, you can disable any of the
existing filters using –disable-filters.
The configure output will show the audio filters included in your
build.
Below is a description of the currently available audio filters.
Pass the audio source unchanged to the output.
Below is a description of the currently available audio sources.
Null audio source, never return audio frames. It is mainly useful as a
template and to be employed in analysis / debugging tools.
It accepts as optional parameter a string of the form
sample_rate:channel_layout.
sample_rate specify the sample rate, and defaults to 44100.
channel_layout specify the channel layout, and can be either an
integer or a string representing a channel layout. The default value
of channel_layout is 3, which corresponds to CH_LAYOUT_STEREO.
Check the channel_layout_map definition in
‘libavcodec/audioconvert.c’ for the mapping between strings and
channel layout values.
Follow some examples:
| | # set the sample rate to 48000 Hz and the channel layout to CH_LAYOUT_MONO.
anullsrc=48000:4
# same as
anullsrc=48000:mono
|
Below is a description of the currently available audio sinks.
Null audio sink, do absolutely nothing with the input audio. It is
mainly useful as a template and to be employed in analysis / debugging
tools.
When you configure your Libav build, you can disable any of the
existing filters using –disable-filters.
The configure output will show the video filters included in your
build.
Below is a description of the currently available video filters.
Detect frames that are (almost) completely black. Can be useful to
detect chapter transitions or commercials. Output lines consist of
the frame number of the detected frame, the percentage of blackness,
the position in the file if known or -1 and the timestamp in seconds.
In order to display the output lines, you need to set the loglevel at
least to the AV_LOG_INFO value.
The filter accepts the syntax:
| | blackframe[=amount:[threshold]]
|
amount is the percentage of the pixels that have to be below the
threshold, and defaults to 98.
threshold is the threshold below which a pixel value is
considered black, and defaults to 32.
Copy the input source unchanged to the output. Mainly useful for
testing purposes.
Crop the input video to out_w:out_h:x:y.
The parameters are expressions containing the following constants:
- ‘E, PI, PHI’
the corresponding mathematical approximated values for e
(euler number), pi (greek PI), PHI (golden ratio)
- ‘x, y’
the computed values for x and y. They are evaluated for
each new frame.
- ‘in_w, in_h’
the input width and heigth
- ‘iw, ih’
same as in_w and in_h
- ‘out_w, out_h’
the output (cropped) width and heigth
- ‘ow, oh’
same as out_w and out_h
- ‘n’
the number of input frame, starting from 0
- ‘pos’
the position in the file of the input frame, NAN if unknown
- ‘t’
timestamp expressed in seconds, NAN if the input timestamp is unknown
The out_w and out_h parameters specify the expressions for
the width and height of the output (cropped) video. They are
evaluated just at the configuration of the filter.
The default value of out_w is "in_w", and the default value of
out_h is "in_h".
The expression for out_w may depend on the value of out_h,
and the expression for out_h may depend on out_w, but they
cannot depend on x and y, as x and y are
evaluated after out_w and out_h.
The x and y parameters specify the expressions for the
position of the top-left corner of the output (non-cropped) area. They
are evaluated for each frame. If the evaluated value is not valid, it
is approximated to the nearest valid value.
The default value of x is "(in_w-out_w)/2", and the default
value for y is "(in_h-out_h)/2", which set the cropped area at
the center of the input image.
The expression for x may depend on y, and the expression
for y may depend on x.
Follow some examples:
| | # crop the central input area with size 100x100
crop=100:100
# crop the central input area with size 2/3 of the input video
"crop=2/3*in_w:2/3*in_h"
# crop the input video central square
crop=in_h
# delimit the rectangle with the top-left corner placed at position
# 100:100 and the right-bottom corner corresponding to the right-bottom
# corner of the input image.
crop=in_w-100:in_h-100:100:100
# crop 10 pixels from the left and right borders, and 20 pixels from
# the top and bottom borders
"crop=in_w-2*10:in_h-2*20"
# keep only the bottom right quarter of the input image
"crop=in_w/2:in_h/2:in_w/2:in_h/2"
# crop height for getting Greek harmony
"crop=in_w:1/PHI*in_w"
# trembling effect
"crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)"
# erratic camera effect depending on timestamp
"crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)"
# set x depending on the value of y
"crop=in_w/2:in_h/2:y:10+10*sin(n/10)"
|
Auto-detect crop size.
Calculate necessary cropping parameters and prints the recommended
parameters through the logging system. The detected dimensions
correspond to the non-black area of the input video.
It accepts the syntax:
| | cropdetect[=limit[:round[:reset]]]
|
- ‘limit’
Threshold, which can be optionally specified from nothing (0) to
everything (255), defaults to 24.
- ‘round’
Value which the width/height should be divisible by, defaults to
16. The offset is automatically adjusted to center the video. Use 2 to
get only even dimensions (needed for 4:2:2 video). 16 is best when
encoding to most video codecs.
- ‘reset’
Counter that determines after how many frames cropdetect will reset
the previously detected largest video area and start over to detect
the current optimal crop area. Defaults to 0.
This can be useful when channel logos distort the video area. 0
indicates never reset and return the largest area encountered during
playback.
Draw a colored box on the input image.
It accepts the syntax:
| | drawbox=x:y:width:height:color
|
- ‘x, y’
Specify the top left corner coordinates of the box. Default to 0.
- ‘width, height’
Specify the width and height of the box, if 0 they are interpreted as
the input width and height. Default to 0.
- ‘color’
Specify the color of the box to write, it can be the name of a color
(case insensitive match) or a 0xRRGGBB[AA] sequence.
Follow some examples:
| | # draw a black box around the edge of the input image
drawbox
# draw a box with color red and an opacity of 50%
drawbox=10:20:200:60:red@0.5"
|
Draw text string or text from specified file on top of video using the
libfreetype library.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libfreetype.
The filter also recognizes strftime() sequences in the provided text
and expands them accordingly. Check the documentation of strftime().
The filter accepts parameters as a list of key=value pairs,
separated by ":".
The description of the accepted parameters follows.
- ‘fontfile’
The font file to be used for drawing text. Path must be included.
This parameter is mandatory.
- ‘text’
The text string to be drawn. The text must be a sequence of UTF-8
encoded characters.
This parameter is mandatory if no file is specified with the parameter
textfile.
- ‘textfile’
A text file containing text to be drawn. The text must be a sequence
of UTF-8 encoded characters.
This parameter is mandatory if no text string is specified with the
parameter text.
If both text and textfile are specified, an error is thrown.
- ‘x, y’
The offsets where text will be drawn within the video frame.
Relative to the top/left border of the output image.
The default value of x and y is 0.
- ‘fontsize’
The font size to be used for drawing text.
The default value of fontsize is 16.
- ‘fontcolor’
The color to be used for drawing fonts.
Either a string (e.g. "red") or in 0xRRGGBB[AA] format
(e.g. "0xff000033"), possibly followed by an alpha specifier.
The default value of fontcolor is "black".
- ‘boxcolor’
The color to be used for drawing box around text.
Either a string (e.g. "yellow") or in 0xRRGGBB[AA] format
(e.g. "0xff00ff"), possibly followed by an alpha specifier.
The default value of boxcolor is "white".
- ‘box’
Used to draw a box around text using background color.
Value should be either 1 (enable) or 0 (disable).
The default value of box is 0.
- ‘shadowx, shadowy’
The x and y offsets for the text shadow position with respect to the
position of the text. They can be either positive or negative
values. Default value for both is "0".
- ‘shadowcolor’
The color to be used for drawing a shadow behind the drawn text. It
can be a color name (e.g. "yellow") or a string in the 0xRRGGBB[AA]
form (e.g. "0xff00ff"), possibly followed by an alpha specifier.
The default value of shadowcolor is "black".
- ‘ft_load_flags’
Flags to be used for loading the fonts.
The flags map the corresponding flags supported by libfreetype, and are
a combination of the following values:
- default
- no_scale
- no_hinting
- render
- no_bitmap
- vertical_layout
- force_autohint
- crop_bitmap
- pedantic
- ignore_global_advance_width
- no_recurse
- ignore_transform
- monochrome
- linear_design
- no_autohint
- end table
Default value is "render".
For more information consult the documentation for the FT_LOAD_*
libfreetype flags.
- ‘tabsize’
The size in number of spaces to use for rendering the tab.
Default value is 4.
For example the command:
| | drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'"
|
will draw "Test Text" with font FreeSerif, using the default values
for the optional parameters.
The command:
| | drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\
x=100: y=50: fontsize=24: fontcolor=yellow@0.2: box=1: boxcolor=red@0.2"
|
will draw ’Test Text’ with font FreeSerif of size 24 at position x=100
and y=50 (counting from the top-left corner of the screen), text is
yellow with a red box around it. Both the text and the box have an
opacity of 20%.
Note that the double quotes are not necessary if spaces are not used
within the parameter list.
For more information about libfreetype, check:
http://www.freetype.org/.
Apply fade-in/out effect to input video.
It accepts the parameters:
type:start_frame:nb_frames
type specifies if the effect type, can be either "in" for
fade-in, or "out" for a fade-out effect.
start_frame specifies the number of the start frame for starting
to apply the fade effect.
nb_frames specifies the number of frames for which the fade
effect has to last. At the end of the fade-in effect the output video
will have the same intensity as the input video, at the end of the
fade-out transition the output video will be completely black.
A few usage examples follow, usable too as test scenarios.
| | # fade in first 30 frames of video
fade=in:0:30
# fade out last 45 frames of a 200-frame video
fade=out:155:45
# fade in first 25 frames and fade out last 25 frames of a 1000-frame video
fade=in:0:25, fade=out:975:25
# make first 5 frames black, then fade in from frame 5-24
fade=in:5:20
|
Transform the field order of the input video.
It accepts one parameter which specifies the required field order that
the input interlaced video will be transformed to. The parameter can
assume one of the following values:
- ‘0 or bff’
output bottom field first
- ‘1 or tff’
output top field first
Default value is "tff".
Transformation is achieved by shifting the picture content up or down
by one line, and filling the remaining line with appropriate picture content.
This method is consistent with most broadcast field order converters.
If the input video is not flagged as being interlaced, or it is already
flagged as being of the required output field order then this filter does
not alter the incoming video.
This filter is very useful when converting to or from PAL DV material,
which is bottom field first.
For example:
| | ./ffmpeg -i in.vob -vf "fieldorder=bff" out.dv
|
Buffer input images and send them when they are requested.
This filter is mainly useful when auto-inserted by the libavfilter
framework.
The filter does not take parameters.
Convert the input video to one of the specified pixel formats.
Libavfilter will try to pick one that is supported for the input to
the next filter.
The filter accepts a list of pixel format names, separated by ":",
for example "yuv420p:monow:rgb24".
Some examples follow:
| | # convert the input video to the format "yuv420p"
format=yuv420p
# convert the input video to any of the formats in the list
format=yuv420p:yuv444p:yuv410p
|
Apply a frei0r effect to the input video.
To enable compilation of this filter you need to install the frei0r
header and configure Libav with –enable-frei0r.
The filter supports the syntax:
| | filter_name[{:|=}param1:param2:...:paramN]
|
filter_name is the name to the frei0r effect to load. If the
environment variable FREI0R_PATH is defined, the frei0r effect
is searched in each one of the directories specified by the colon
separated list in FREIOR_PATH, otherwise in the standard frei0r
paths, which are in this order: ‘HOME/.frei0r-1/lib/’,
‘/usr/local/lib/frei0r-1/’, ‘/usr/lib/frei0r-1/’.
param1, param2, ... , paramN specify the parameters
for the frei0r effect.
A frei0r effect parameter can be a boolean (whose values are specified
with "y" and "n"), a double, a color (specified by the syntax
R/G/B, R, G, and B being float
numbers from 0.0 to 1.0) or by an av_parse_color() color
description), a position (specified by the syntax X/Y,
X and Y being float numbers) and a string.
The number and kind of parameters depend on the loaded effect. If an
effect parameter is not specified the default value is set.
Some examples follow:
| | # apply the distort0r effect, set the first two double parameters
frei0r=distort0r:0.5:0.01
# apply the colordistance effect, takes a color as first parameter
frei0r=colordistance:0.2/0.3/0.4
frei0r=colordistance:violet
frei0r=colordistance:0x112233
# apply the perspective effect, specify the top left and top right
# image positions
frei0r=perspective:0.2/0.2:0.8/0.2
|
For more information see:
http://piksel.org/frei0r
Fix the banding artifacts that are sometimes introduced into nearly flat
regions by truncation to 8bit colordepth.
Interpolate the gradients that should go where the bands are, and
dither them.
This filter is designed for playback only. Do not use it prior to
lossy compression, because compression tends to lose the dither and
bring back the bands.
The filter takes two optional parameters, separated by ’:’:
strength:radius
strength is the maximum amount by which the filter will change
any one pixel. Also the threshold for detecting nearly flat
regions. Acceptable values range from .51 to 255, default value is
1.2, out-of-range values will be clipped to the valid range.
radius is the neighborhood to fit the gradient to. A larger
radius makes for smoother gradients, but also prevents the filter from
modifying the pixels near detailed regions. Acceptable values are
8-32, default value is 16, out-of-range values will be clipped to the
valid range.
| | # default parameters
gradfun=1.2:16
# omitting radius
gradfun=1.2
|
Flip the input video horizontally.
For example to horizontally flip the video in input with
‘ffmpeg’:
| | ffmpeg -i in.avi -vf "hflip" out.avi
|
High precision/quality 3d denoise filter. This filter aims to reduce
image noise producing smooth images and making still images really
still. It should enhance compressibility.
It accepts the following optional parameters:
luma_spatial:chroma_spatial:luma_tmp:chroma_tmp
- ‘luma_spatial’
a non-negative float number which specifies spatial luma strength,
defaults to 4.0
- ‘chroma_spatial’
a non-negative float number which specifies spatial chroma strength,
defaults to 3.0*luma_spatial/4.0
- ‘luma_tmp’
a float number which specifies luma temporal strength, defaults to
6.0*luma_spatial/4.0
- ‘chroma_tmp’
a float number which specifies chroma temporal strength, defaults to
luma_tmp*chroma_spatial/luma_spatial
Force libavfilter not to use any of the specified pixel formats for the
input to the next filter.
The filter accepts a list of pixel format names, separated by ":",
for example "yuv420p:monow:rgb24".
Some examples follow:
| | # force libavfilter to use a format different from "yuv420p" for the
# input to the vflip filter
noformat=yuv420p,vflip
# convert the input video to any of the formats not contained in the list
noformat=yuv420p:yuv444p:yuv410p
|
Pass the video source unchanged to the output.
Apply video transform using libopencv.
To enable this filter install libopencv library and headers and
configure Libav with –enable-libopencv.
The filter takes the parameters: filter_name{:=}filter_params.
filter_name is the name of the libopencv filter to apply.
filter_params specifies the parameters to pass to the libopencv
filter. If not specified the default values are assumed.
Refer to the official libopencv documentation for more precise
informations:
http://opencv.willowgarage.com/documentation/c/image_filtering.html
Follows the list of supported libopencv filters.
Dilate an image by using a specific structuring element.
This filter corresponds to the libopencv function cvDilate.
It accepts the parameters: struct_el:nb_iterations.
struct_el represents a structuring element, and has the syntax:
colsxrows+anchor_xxanchor_y/shape
cols and rows represent the number of colums and rows of
the structuring element, anchor_x and anchor_y the anchor
point, and shape the shape for the structuring element, and
can be one of the values "rect", "cross", "ellipse", "custom".
If the value for shape is "custom", it must be followed by a
string of the form "=filename". The file with name
filename is assumed to represent a binary image, with each
printable character corresponding to a bright pixel. When a custom
shape is used, cols and rows are ignored, the number
or columns and rows of the read file are assumed instead.
The default value for struct_el is "3x3+0x0/rect".
nb_iterations specifies the number of times the transform is
applied to the image, and defaults to 1.
Follow some example:
| | # use the default values
ocv=dilate
# dilate using a structuring element with a 5x5 cross, iterate two times
ocv=dilate=5x5+2x2/cross:2
# read the shape from the file diamond.shape, iterate two times
# the file diamond.shape may contain a pattern of characters like this:
# *
# ***
# *****
# ***
# *
# the specified cols and rows are ignored (but not the anchor point coordinates)
ocv=0x0+2x2/custom=diamond.shape:2
|
Erode an image by using a specific structuring element.
This filter corresponds to the libopencv function cvErode.
The filter accepts the parameters: struct_el:nb_iterations,
with the same meaning and use of those of the dilate filter
(see dilate).
Smooth the input video.
The filter takes the following parameters:
type:param1:param2:param3:param4.
type is the type of smooth filter to apply, and can be one of
the following values: "blur", "blur_no_scale", "median", "gaussian",
"bilateral". The default value is "gaussian".
param1, param2, param3, and param4 are
parameters whose meanings depend on smooth type. param1 and
param2 accept integer positive values or 0, param3 and
param4 accept float values.
The default value for param1 is 3, the default value for the
other parameters is 0.
These parameters correspond to the parameters assigned to the
libopencv function cvSmooth.
Overlay one video on top of another.
It takes two inputs and one output, the first input is the "main"
video on which the second input is overlayed.
It accepts the parameters: x:y.
x is the x coordinate of the overlayed video on the main video,
y is the y coordinate. The parameters are expressions containing
the following parameters:
- ‘main_w, main_h’
main input width and height
- ‘W, H’
same as main_w and main_h
- ‘overlay_w, overlay_h’
overlay input width and height
- ‘w, h’
same as overlay_w and overlay_h
Be aware that frames are taken from each input video in timestamp
order, hence, if their initial timestamps differ, it is a a good idea
to pass the two inputs through a setpts=PTS-STARTPTS filter to
have them begin in the same zero timestamp, as it does the example for
the movie filter.
Follow some examples:
| | # draw the overlay at 10 pixels from the bottom right
# corner of the main video.
overlay=main_w-overlay_w-10:main_h-overlay_h-10
# insert a transparent PNG logo in the bottom left corner of the input
movie=logo.png [logo];
[in][logo] overlay=10:main_h-overlay_h-10 [out]
# insert 2 different transparent PNG logos (second logo on bottom
# right corner):
movie=logo1.png [logo1];
movie=logo2.png [logo2];
[in][logo1] overlay=10:H-h-10 [in+logo1];
[in+logo1][logo2] overlay=W-w-10:H-h-10 [out]
# add a transparent color layer on top of the main video,
# WxH specifies the size of the main input to the overlay filter
color=red.3:WxH [over]; [in][over] overlay [out]
|
You can chain togheter more overlays but the efficiency of such
approach is yet to be tested.
Add paddings to the input image, and places the original input at the
given coordinates x, y.
It accepts the following parameters:
width:height:x:y:color.
The parameters width, height, x, and y are
expressions containing the following constants:
- ‘E, PI, PHI’
the corresponding mathematical approximated values for e
(euler number), pi (greek PI), phi (golden ratio)
- ‘in_w, in_h’
the input video width and heigth
- ‘iw, ih’
same as in_w and in_h
- ‘out_w, out_h’
the output width and heigth, that is the size of the padded area as
specified by the width and height expressions
- ‘ow, oh’
same as out_w and out_h
- ‘x, y’
x and y offsets as specified by the x and y
expressions, or NAN if not yet specified
- ‘a’
input display aspect ratio, same as iw / ih
- ‘hsub, vsub’
horizontal and vertical chroma subsample values. For example for the
pixel format "yuv422p" hsub is 2 and vsub is 1.
Follows the description of the accepted parameters.
- ‘width, height’
-
Specify the size of the output image with the paddings added. If the
value for width or height is 0, the corresponding input size
is used for the output.
The width expression can reference the value set by the
height expression, and viceversa.
The default value of width and height is 0.
- ‘x, y’
-
Specify the offsets where to place the input image in the padded area
with respect to the top/left border of the output image.
The x expression can reference the value set by the y
expression, and viceversa.
The default value of x and y is 0.
- ‘color’
-
Specify the color of the padded area, it can be the name of a color
(case insensitive match) or a 0xRRGGBB[AA] sequence.
The default value of color is "black".
Some examples follow:
| | # Add paddings with color "violet" to the input video. Output video
# size is 640x480, the top-left corner of the input video is placed at
# column 0, row 40.
pad=640:480:0:40:violet
# pad the input to get an output with dimensions increased bt 3/2,
# and put the input video at the center of the padded area
pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2"
# pad the input to get a squared output with size equal to the maximum
# value between the input width and height, and put the input video at
# the center of the padded area
pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2"
# pad the input to get a final w/h ratio of 16:9
pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"
# double output size and put the input video in the bottom-right
# corner of the output padded area
pad="2*iw:2*ih:ow-iw:oh-ih"
|
Pixel format descriptor test filter, mainly useful for internal
testing. The output video should be equal to the input video.
For example:
| | format=monow, pixdesctest
|
can be used to test the monowhite pixel format descriptor definition.
Scale the input video to width:height and/or convert the image format.
The parameters width and height are expressions containing
the following constants:
- ‘E, PI, PHI’
the corresponding mathematical approximated values for e
(euler number), pi (greek PI), phi (golden ratio)
- ‘in_w, in_h’
the input width and heigth
- ‘iw, ih’
same as in_w and in_h
- ‘out_w, out_h’
the output (cropped) width and heigth
- ‘ow, oh’
same as out_w and out_h
- ‘a’
input display aspect ratio, same as iw / ih
- ‘hsub, vsub’
horizontal and vertical chroma subsample values. For example for the
pixel format "yuv422p" hsub is 2 and vsub is 1.
If the input image format is different from the format requested by
the next filter, the scale filter will convert the input to the
requested format.
If the value for width or height is 0, the respective input
size is used for the output.
If the value for width or height is -1, the scale filter will
use, for the respective output size, a value that maintains the aspect
ratio of the input image.
The default value of width and height is 0.
Some examples follow:
| | # scale the input video to a size of 200x100.
scale=200:100
# scale the input to 2x
scale=2*iw:2*ih
# the above is the same as
scale=2*in_w:2*in_h
# scale the input to half size
scale=iw/2:ih/2
# increase the width, and set the height to the same size
scale=3/2*iw:ow
# seek for Greek harmony
scale=iw:1/PHI*iw
scale=ih*PHI:ih
# increase the height, and set the width to 3/2 of the height
scale=3/2*oh:3/5*ih
# increase the size, but make the size a multiple of the chroma
scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub"
# increase the width to a maximum of 500 pixels, keep the same input aspect ratio
scale='min(500\, iw*3/2):-1'
|
Set the Display Aspect Ratio for the filter output video.
This is done by changing the specified Sample (aka Pixel) Aspect
Ratio, according to the following equation:
DAR = HORIZONTAL_RESOLUTION / VERTICAL_RESOLUTION * SAR
Keep in mind that this filter does not modify the pixel dimensions of
the video frame. Also the display aspect ratio set by this filter may
be changed by later filters in the filterchain, e.g. in case of
scaling or if another "setdar" or a "setsar" filter is applied.
The filter accepts a parameter string which represents the wanted
display aspect ratio.
The parameter can be a floating point number string, or an expression
of the form num:den, where num and den are the
numerator and denominator of the aspect ratio.
If the parameter is not specified, it is assumed the value "0:1".
For example to change the display aspect ratio to 16:9, specify:
| | setdar=16:9
# the above is equivalent to
setdar=1.77777
|
See also the "setsar" filter documentation (see setsar).
Change the PTS (presentation timestamp) of the input video frames.
Accept in input an expression evaluated through the eval API, which
can contain the following constants:
- ‘PTS’
the presentation timestamp in input
- ‘PI’
Greek PI
- ‘PHI’
golden ratio
- ‘E’
Euler number
- ‘N’
the count of the input frame, starting from 0.
- ‘STARTPTS’
the PTS of the first video frame
- ‘INTERLACED’
tell if the current frame is interlaced
- ‘POS’
original position in the file of the frame, or undefined if undefined
for the current frame
- ‘PREV_INPTS’
previous input PTS
- ‘PREV_OUTPTS’
previous output PTS
Some examples follow:
| | # start counting PTS from zero
setpts=PTS-STARTPTS
# fast motion
setpts=0.5*PTS
# slow motion
setpts=2.0*PTS
# fixed rate 25 fps
setpts=N/(25*TB)
# fixed rate 25 fps with some jitter
setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'
|
Set the Sample (aka Pixel) Aspect Ratio for the filter output video.
Note that as a consequence of the application of this filter, the
output display aspect ratio will change according to the following
equation:
DAR = HORIZONTAL_RESOLUTION / VERTICAL_RESOLUTION * SAR
Keep in mind that the sample aspect ratio set by this filter may be
changed by later filters in the filterchain, e.g. if another "setsar"
or a "setdar" filter is applied.
The filter accepts a parameter string which represents the wanted
sample aspect ratio.
The parameter can be a floating point number string, or an expression
of the form num:den, where num and den are the
numerator and denominator of the aspect ratio.
If the parameter is not specified, it is assumed the value "0:1".
For example to change the sample aspect ratio to 10:11, specify:
Set the timebase to use for the output frames timestamps.
It is mainly useful for testing timebase configuration.
It accepts in input an arithmetic expression representing a rational.
The expression can contain the constants "PI", "E", "PHI", "AVTB" (the
default timebase), and "intb" (the input timebase).
The default value for the input is "intb".
Follow some examples.
| | # set the timebase to 1/25
settb=1/25
# set the timebase to 1/10
settb=0.1
#set the timebase to 1001/1000
settb=1+0.001
#set the timebase to 2*intb
settb=2*intb
#set the default timebase value
settb=AVTB
|
Pass the images of input video on to next video filter as multiple
slices.
| | ./ffmpeg -i in.avi -vf "slicify=32" out.avi
|
The filter accepts the slice height as parameter. If the parameter is
not specified it will use the default value of 16.
Adding this in the beginning of filter chains should make filtering
faster due to better use of the memory cache.
Transpose rows with columns in the input video and optionally flip it.
It accepts a parameter representing an integer, which can assume the
values:
- ‘0’
Rotate by 90 degrees counterclockwise and vertically flip (default), that is:
| | L.R L.l
. . -> . .
l.r R.r
|
- ‘1’
Rotate by 90 degrees clockwise, that is:
| | L.R l.L
. . -> . .
l.r r.R
|
- ‘2’
Rotate by 90 degrees counterclockwise, that is:
| | L.R R.r
. . -> . .
l.r L.l
|
- ‘3’
Rotate by 90 degrees clockwise and vertically flip, that is:
| | L.R r.R
. . -> . .
l.r l.L
|
Sharpen or blur the input video.
It accepts the following parameters:
luma_msize_x:luma_msize_y:luma_amount:chroma_msize_x:chroma_msize_y:chroma_amount
Negative values for the amount will blur the input video, while positive
values will sharpen. All parameters are optional and default to the
equivalent of the string ’5:5:1.0:0:0:0.0’.
- ‘luma_msize_x’
Set the luma matrix horizontal size. It can be an integer between 3
and 13, default value is 5.
- ‘luma_msize_y’
Set the luma matrix vertical size. It can be an integer between 3
and 13, default value is 5.
- ‘luma_amount’
Set the luma effect strength. It can be a float number between -2.0
and 5.0, default value is 1.0.
- ‘chroma_msize_x’
Set the chroma matrix horizontal size. It can be an integer between 3
and 13, default value is 0.
- ‘chroma_msize_y’
Set the chroma matrix vertical size. It can be an integer between 3
and 13, default value is 0.
- ‘luma_amount’
Set the chroma effect strength. It can be a float number between -2.0
and 5.0, default value is 0.0.
| | # Strong luma sharpen effect parameters
unsharp=7:7:2.5
# Strong blur of both luma and chroma parameters
unsharp=7:7:-2:7:7:-2
# Use the default values with ffmpeg
./ffmpeg -i in.avi -vf "unsharp" out.mp4
|
Flip the input video vertically.
| | ./ffmpeg -i in.avi -vf "vflip" out.avi
|
Deinterlace the input video ("yadif" means "yet another deinterlacing
filter").
It accepts the optional parameters: mode:parity.
mode specifies the interlacing mode to adopt, accepts one of the
following values:
- ‘0’
output 1 frame for each frame
- ‘1’
output 1 frame for each field
- ‘2’
like 0 but skips spatial interlacing check
- ‘3’
like 1 but skips spatial interlacing check
Default value is 0.
parity specifies the picture field parity assumed for the input
interlaced video, accepts one of the following values:
- ‘0’
assume top field first
- ‘1’
assume bottom field first
- ‘-1’
enable automatic detection
Default value is -1.
If interlacing is unknown or decoder does not export this information,
top field first will be assumed.
Below is a description of the currently available video sources.
Buffer video frames, and make them available to the filter chain.
This source is mainly intended for a programmatic use, in particular
through the interface defined in ‘libavfilter/vsrc_buffer.h’.
It accepts the following parameters:
width:height:pix_fmt_string:timebase_num:timebase_den:sample_aspect_ratio_num:sample_aspect_ratio.den
All the parameters need to be explicitely defined.
Follows the list of the accepted parameters.
- ‘width, height’
Specify the width and height of the buffered video frames.
- ‘pix_fmt_string’
A string representing the pixel format of the buffered video frames.
It may be a number corresponding to a pixel format, or a pixel format
name.
- ‘timebase_num, timebase_den’
Specify numerator and denomitor of the timebase assumed by the
timestamps of the buffered frames.
- ‘sample_aspect_ratio.num, sample_aspect_ratio.den’
Specify numerator and denominator of the sample aspect ratio assumed
by the video frames.
For example:
| | buffer=320:240:yuv410p:1:24:1:1
|
will instruct the source to accept video frames with size 320x240 and
with format "yuv410p", assuming 1/24 as the timestamps timebase and
square pixels (1:1 sample aspect ratio).
Since the pixel format with name "yuv410p" corresponds to the number 6
(check the enum PixelFormat definition in ‘libavutil/pixfmt.h’),
this example corresponds to:
Provide an uniformly colored input.
It accepts the following parameters:
color:frame_size:frame_rate
Follows the description of the accepted parameters.
- ‘color’
Specify the color of the source. It can be the name of a color (case
insensitive match) or a 0xRRGGBB[AA] sequence, possibly followed by an
alpha specifier. The default value is "black".
- ‘frame_size’
Specify the size of the sourced video, it may be a string of the form
widthxheigth, or the name of a size abbreviation. The
default value is "320x240".
- ‘frame_rate’
Specify the frame rate of the sourced video, as the number of frames
generated per second. It has to be a string in the format
frame_rate_num/frame_rate_den, an integer number, a float
number or a valid video frame rate abbreviation. The default value is
"25".
For example the following graph description will generate a red source
with an opacity of 0.2, with size "qcif" and a frame rate of 10
frames per second, which will be overlayed over the source connected
to the pad with identifier "in".
| | "color=red@0.2:qcif:10 [color]; [in][color] overlay [out]"
|
Read a video stream from a movie container.
It accepts the syntax: movie_name[:options] where
movie_name is the name of the resource to read (not necessarily
a file but also a device or a stream accessed through some protocol),
and options is an optional sequence of key=value
pairs, separated by ":".
The description of the accepted options follows.
- ‘format_name, f’
Specifies the format assumed for the movie to read, and can be either
the name of a container or an input device. If not specified the
format is guessed from movie_name or by probing.
- ‘seek_point, sp’
Specifies the seek point in seconds, the frames will be output
starting from this seek point, the parameter is evaluated with
av_strtod so the numerical value may be suffixed by an IS
postfix. Default value is "0".
- ‘stream_index, si’
Specifies the index of the video stream to read. If the value is -1,
the best suited video stream will be automatically selected. Default
value is "-1".
This filter allows to overlay a second video on top of main input of
a filtergraph as shown in this graph:
| | input -----------> deltapts0 --> overlay --> output
^
|
movie --> scale--> deltapts1 -------+
|
Some examples follow:
| | # skip 3.2 seconds from the start of the avi file in.avi, and overlay it
# on top of the input labelled as "in".
movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [movie];
[in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out]
# read from a video4linux2 device, and overlay it on top of the input
# labelled as "in"
movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [movie];
[in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out]
|
Null video source, never return images. It is mainly useful as a
template and to be employed in analysis / debugging tools.
It accepts as optional parameter a string of the form
width:height:timebase.
width and height specify the size of the configured
source. The default values of width and height are
respectively 352 and 288 (corresponding to the CIF size format).
timebase specifies an arithmetic expression representing a
timebase. The expression can contain the constants "PI", "E", "PHI",
"AVTB" (the default timebase), and defaults to the value "AVTB".
Provide a frei0r source.
To enable compilation of this filter you need to install the frei0r
header and configure Libav with –enable-frei0r.
The source supports the syntax:
| | size:rate:src_name[{=|:}param1:param2:...:paramN]
|
size is the size of the video to generate, may be a string of the
form widthxheight or a frame size abbreviation.
rate is the rate of the video to generate, may be a string of
the form num/den or a frame rate abbreviation.
src_name is the name to the frei0r source to load. For more
information regarding frei0r and how to set the parameters read the
section "frei0r" (see frei0r) in the description of the video
filters.
Some examples follow:
| | # generate a frei0r partik0l source with size 200x200 and framerate 10
# which is overlayed on the overlay filter main input
frei0r_src=200x200:10:partik0l=1234 [overlay]; [in][overlay] overlay
|
Below is a description of the currently available video sinks.
Null video sink, do absolutely nothing with the input video. It is
mainly useful as a template and to be employed in analysis / debugging
tools.
Libav is able to dump metadata from media files into a simple UTF-8-encoded
INI-like text file and then load it back using the metadata muxer/demuxer.
The file format is as follows:
-
A file consists of a header and a number of metadata tags divided into sections,
each on its own line.
-
The header is a ’;FFMETADATA’ string, followed by a version number (now 1).
-
Metadata tags are of the form ’key=value’
-
Immediately after header follows global metadata
-
After global metadata there may be sections with per-stream/per-chapter
metadata.
-
A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in
brackets (’[’, ’]’) and ends with next section or end of file.
-
At the beginning of a chapter section there may be an optional timebase to be
used for start/end values. It must be in form ’TIMEBASE=num/den’, where num and
den are integers. If the timebase is missing then start/end times are assumed to
be in milliseconds.
Next a chapter section must contain chapter start and end times in form
’START=num’, ’END=num’, where num is a positive integer.
-
Empty lines and lines starting with ’;’ or ’#’ are ignored.
-
Metadata keys or values containing special characters (’=’, ’;’, ’#’, ’\’ and a
newline) must be escaped with a backslash ’\’.
-
Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of
the tag (in the example above key is ’foo ’, value is ’ bar’).
A ffmetadata file might look like this:
| | ;FFMETADATA1
title=bike\\shed
;this is a comment
artist=Libav troll team
[CHAPTER]
TIMEBASE=1/1000
START=0
#chapter ends at 0:01:00
END=60000
title=chapter \#1
[STREAM]
title=multi\
line
|
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