ffmpeg-all man page on DragonFly

ffmpeg-all man page on DragonFly

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FFMPEG-ALL(1) FFMPEG-ALL(1)

NAME
ffmpeg - ffmpeg video converter

SYNOPSIS
ffmpeg [global_options] {[input_file_options] -i input_file} ...
{[output_file_options] output_file} ...

DESCRIPTION
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.

ffmpeg reads from an arbitrary number of input "files" (which can be
regular files, pipes, network streams, grabbing devices, etc.),
specified by the "-i" option, and writes to an arbitrary number of
output "files", which are specified by a plain output filename.
Anything found on the command line which cannot be interpreted as an
option is considered to be an output filename.

Each input or output file can, in principle, contain any number of
streams of different types (video/audio/subtitle/attachment/data). The
allowed number and/or types of streams may be limited by the container
format. Selecting which streams from which inputs will go into which
output is either done automatically or with the "-map" option (see the
Stream selection chapter).

To refer to input files in options, you must use their indices
(0-based). E.g. the first input file is 0, the second is 1, etc.
Similarly, streams within a file are referred to by their indices. E.g.
"2:3" refers to the fourth stream in the third input file. Also see the
Stream specifiers chapter.

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. Exceptions from this rule are the global
options (e.g. verbosity level), which should be specified first.

Do not mix input and output files -- first specify all input files,
then all output files. Also do not mix options which belong to
different files. All options apply ONLY to the next input or output
file and are reset between files.

· To set the video bitrate of the output file to 64 kbit/s:

ffmpeg -i input.avi -b:v 64k -bufsize 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.

DETAILED DESCRIPTION
The transcoding process in ffmpeg for each output can be described by
the following diagram:

_______ ______________
| | | |
| input | demuxer | encoded data | decoder
| file | ---------> | packets | -----+
|_______| |______________| |
v
_________
| |
| decoded |
| frames |
|_________|
________ ______________ |
| | | | |
| output | <-------- | encoded data | <----+
| file | muxer | packets | encoder
|________| |______________|

ffmpeg calls the libavformat library (containing demuxers) to read
input files and get packets containing encoded data from them. When
there are multiple input files, ffmpeg tries to keep them synchronized
by tracking lowest timestamp on any active input stream.

Encoded packets are then passed to the decoder (unless streamcopy is
selected for the stream, see further for a description). The decoder
produces uncompressed frames (raw video/PCM audio/...) which can be
processed further by filtering (see next section). After filtering, the
frames are passed to the encoder, which encodes them and outputs
encoded packets. Finally those are passed to the muxer, which writes
the encoded packets to the output file.

Filtering
Before encoding, ffmpeg can process raw audio and video frames using
filters from the libavfilter library. Several chained filters form a
filter graph. ffmpeg distinguishes between two types of filtergraphs:
simple and complex.

Simple filtergraphs

Simple filtergraphs are those that have exactly one input and output,
both of the same type. In the above diagram they can be represented by
simply inserting an additional step between decoding and encoding:

_________ ______________
| | | |
| decoded | | encoded data |
| frames |\ _ | packets |
|_________| \ /||______________|
\ __________ /
simple _\|| | / encoder
filtergraph | filtered |/
| frames |
|__________|

Simple filtergraphs are configured with the per-stream -filter option
(with -vf and -af aliases for video and audio respectively). A simple
filtergraph for video can look for example like this:

_______ _____________ _______ ________
| | | | | | | |
| input | ---> | deinterlace | ---> | scale | ---> | output |
|_______| |_____________| |_______| |________|

Note that some filters change frame properties but not frame contents.
E.g. the "fps" filter in the example above changes number of frames,
but does not touch the frame contents. Another example is the "setpts"
filter, which only sets timestamps and otherwise passes the frames
unchanged.

Complex filtergraphs

Complex filtergraphs are those which cannot be described as simply a
linear processing chain applied to one stream. This is the case, for
example, when the graph has more than one input and/or output, or when
output stream type is different from input. They can be represented
with the following diagram:

_________
| |
| input 0 |\ __________
|_________| \ | |
\ _________ /| output 0 |
\ | | / |__________|
_________ \| complex | /
| | | |/
| input 1 |---->| filter |\
|_________| | | \ __________
/| graph | \ | |
/ | | \| output 1 |
_________ / |_________| |__________|
| | /
| input 2 |/
|_________|

Complex filtergraphs are configured with the -filter_complex option.
Note that this option is global, since a complex filtergraph, by its
nature, cannot be unambiguously associated with a single stream or
file.

The -lavfi option is equivalent to -filter_complex.

A trivial example of a complex filtergraph is the "overlay" filter,
which has two video inputs and one video output, containing one video
overlaid on top of the other. Its audio counterpart is the "amix"
filter.

Stream copy
Stream copy is a mode selected by supplying the "copy" parameter to the
-codec option. It makes ffmpeg omit the decoding and encoding step for
the specified stream, so it does only demuxing and muxing. It is useful
for changing the container format or modifying container-level
metadata. The diagram above will, in this case, simplify to this:

_______ ______________ ________
| | | | | |
| input | demuxer | encoded data | muxer | output |
| file | ---------> | packets | -------> | file |
|_______| |______________| |________|

Since there is no decoding or encoding, it is very fast and there is no
quality loss. However, it might not work in some cases because of many
factors. Applying filters is obviously also impossible, since filters
work on uncompressed data.

STREAM SELECTION
By default, ffmpeg includes only one stream of each type (video, audio,
subtitle) present in the input files and adds them to each output file.
It picks the "best" of each based upon the following criteria: for
video, it is the stream with the highest resolution, for audio, it is
the stream with the most channels, for subtitles, it is the first
subtitle stream. In the case where several streams of the same type
rate equally, the stream with the lowest index is chosen.

You can disable some of those defaults by using the "-vn/-an/-sn"
options. For full manual control, use the "-map" option, which disables
the defaults just described.

OPTIONS
All the numerical options, if not specified otherwise, accept a string
representing a number as input, which may be followed by one of the SI
unit prefixes, for example: 'K', 'M', or 'G'.

If 'i' is appended to the SI unit prefix, the complete prefix will be
interpreted as a unit prefix for binary multiples, which are based on
powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
prefix multiplies the value by 8. This allows using, for example: 'KB',
'MiB', 'G' and 'B' as number suffixes.

Options which do not take arguments are boolean options, and set the
corresponding value to true. They can be set to false by prefixing the
option name with "no". For example using "-nofoo" will set the boolean
option with name "foo" to false.

Stream specifiers
Some options are applied per-stream, e.g. bitrate or codec. Stream
specifiers are used to precisely specify which stream(s) a given option
belongs to.

A stream specifier is a string generally appended to the option name
and separated from it by a colon. E.g. "-codec:a:1 ac3" contains the
"a:1" stream specifier, which matches the second audio stream.
Therefore, it would select the ac3 codec for the second audio stream.

A stream specifier can match several streams, so that the option is
applied to all of them. E.g. the stream specifier in "-b:a 128k"
matches all audio streams.

An empty stream specifier matches all streams. For example, "-codec
copy" or "-codec: copy" would copy all the streams without reencoding.

Possible forms of stream specifiers are:

stream_index
Matches the stream with this index. E.g. "-threads:1 4" would set
the thread count for the second stream to 4.

stream_type[:stream_index]
stream_type is one of following: 'v' or 'V' for video, 'a' for
audio, 's' for subtitle, 'd' for data, and 't' for attachments. 'v'
matches all video streams, 'V' only matches video streams which are
not attached pictures, video thumbnails or cover arts. If
stream_index is given, then it matches stream number stream_index
of this type. Otherwise, it matches all streams of this type.

p:program_id[:stream_index]
If stream_index is given, then it matches the stream with number
stream_index in the program with the id program_id. Otherwise, it
matches all streams in the program.

#stream_id or i:stream_id
Match the stream by stream id (e.g. PID in MPEG-TS container).

m:key[:value]
Matches streams with the metadata tag key having the specified
value. If value is not given, matches streams that contain the
given tag with any value.

u Matches streams with usable configuration, the codec must be
defined and the essential information such as video dimension or
audio sample rate must be present.

Note that in ffmpeg, matching by metadata will only work properly
for input files.

Generic options
These options are shared amongst the ff* tools.

-L Show license.

-h, -?, -help, --help [arg]
Show help. An optional parameter may be specified to print help
about a specific item. If no argument is specified, only basic (non
advanced) tool options are shown.

Possible values of arg are:

long
Print advanced tool options in addition to the basic tool
options.

full
Print complete list of options, including shared and private
options for encoders, decoders, demuxers, muxers, filters, etc.

decoder=decoder_name
Print detailed information about the decoder named
decoder_name. Use the -decoders option to get a list of all
decoders.

encoder=encoder_name
Print detailed information about the encoder named
encoder_name. Use the -encoders option to get a list of all
encoders.

demuxer=demuxer_name
Print detailed information about the demuxer named
demuxer_name. Use the -formats option to get a list of all
demuxers and muxers.

muxer=muxer_name
Print detailed information about the muxer named muxer_name.
Use the -formats option to get a list of all muxers and
demuxers.

filter=filter_name
Print detailed information about the filter name filter_name.
Use the -filters option to get a list of all filters.

-version
Show version.

-formats
Show available formats (including devices).

-devices
Show available devices.

-codecs
Show all codecs known to libavcodec.

Note that the term 'codec' is used throughout this documentation as
a shortcut for what is more correctly called a media bitstream
format.

-decoders
Show available decoders.

-encoders
Show all available encoders.

-bsfs
Show available bitstream filters.

-protocols
Show available protocols.

-filters
Show available libavfilter filters.

-pix_fmts
Show available pixel formats.

-sample_fmts
Show available sample formats.

-layouts
Show channel names and standard channel layouts.

-colors
Show recognized color names.

-sources device[,opt1=val1[,opt2=val2]...]
Show autodetected sources of the intput device. Some devices may
provide system-dependent source names that cannot be autodetected.
The returned list cannot be assumed to be always complete.

ffmpeg -sources pulse,server=192.168.0.4

-sinks device[,opt1=val1[,opt2=val2]...]
Show autodetected sinks of the output device. Some devices may
provide system-dependent sink names that cannot be autodetected.
The returned list cannot be assumed to be always complete.

ffmpeg -sinks pulse,server=192.168.0.4

-loglevel [repeat+]loglevel | -v [repeat+]loglevel
Set the logging level used by the library. Adding "repeat+"
indicates that repeated log output should not be compressed to the
first line and the "Last message repeated n times" line will be
omitted. "repeat" can also be used alone. If "repeat" is used
alone, and with no prior loglevel set, the default loglevel will be
used. If multiple loglevel parameters are given, using 'repeat'
will not change the loglevel. loglevel is a string or a number
containing one of the following values:

quiet, -8
Show nothing at all; be silent.

panic, 0
Only show fatal errors which could lead the process to crash,
such as and assert failure. This is not currently used for
anything.

fatal, 8
Only show fatal errors. These are errors after which the
process absolutely cannot continue after.

error, 16
Show all errors, including ones which can be recovered from.

warning, 24
Show all warnings and errors. Any message related to possibly
incorrect or unexpected events will be shown.

info, 32
Show informative messages during processing. This is in
addition to warnings and errors. This is the default value.

verbose, 40
Same as "info", except more verbose.

debug, 48
Show everything, including debugging information.

trace, 56

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
AV_LOG_FORCE_NOCOLOR or NO_COLOR, or can be forced setting the
environment variable AV_LOG_FORCE_COLOR. The use of the
environment variable NO_COLOR is deprecated and will be dropped in
a following FFmpeg version.

-report
Dump full command line and console output to a file named
"program-YYYYMMDD-HHMMSS.log" in the current directory. This file
can be useful for bug reports. It also implies "-loglevel
verbose".

Setting the environment variable FFREPORT to any value has the same
effect. If the value is a ':'-separated key=value sequence, these
options will affect the report; option values must be escaped if
they contain special characters or the options delimiter ':' (see
the ``Quoting and escaping'' section in the ffmpeg-utils manual).

The following options are recognized:

file
set the file name to use for the report; %p is expanded to the
name of the program, %t is expanded to a timestamp, "%%" is
expanded to a plain "%"

level
set the log verbosity level using a numerical value (see
"-loglevel").

For example, to output a report to a file named ffreport.log using
a log level of 32 (alias for log level "info"):

FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output

Errors in parsing the environment variable are not fatal, and will
not appear in the report.

-hide_banner
Suppress printing banner.

All FFmpeg tools will normally show a copyright notice, build
options and library versions. This option can be used to suppress
printing this information.

-cpuflags flags (global)
Allows setting and clearing cpu flags. This option is intended for
testing. Do not use it unless you know what you're doing.

ffmpeg -cpuflags -sse+mmx ...
ffmpeg -cpuflags mmx ...
ffmpeg -cpuflags 0 ...

Possible flags for this option are:

x86
mmx
mmxext
sse
sse2
sse2slow
sse3
sse3slow
ssse3
atom
sse4.1
sse4.2
avx
avx2
xop
fma3
fma4
3dnow
3dnowext
bmi1
bmi2
cmov
ARM
armv5te
armv6
armv6t2
vfp
vfpv3
neon
setend
AArch64
armv8
vfp
neon
PowerPC
altivec
Specific Processors
pentium2
pentium3
pentium4
k6
k62
athlon
athlonxp
k8
-opencl_bench
This option is used to benchmark all available OpenCL devices and
print the results. This option is only available when FFmpeg has
been compiled with "--enable-opencl".

When FFmpeg is configured with "--enable-opencl", the options for
the global OpenCL context are set via -opencl_options. See the
"OpenCL Options" section in the ffmpeg-utils manual for the
complete list of supported options. Amongst others, these options
include the ability to select a specific platform and device to run
the OpenCL code on. By default, FFmpeg will run on the first device
of the first platform. While the options for the global OpenCL
context provide flexibility to the user in selecting the OpenCL
device of their choice, most users would probably want to select
the fastest OpenCL device for their system.

This option assists the selection of the most efficient
configuration by identifying the appropriate device for the user's
system. The built-in benchmark is run on all the OpenCL devices and
the performance is measured for each device. The devices in the
results list are sorted based on their performance with the fastest
device listed first. The user can subsequently invoke ffmpeg using
the device deemed most appropriate via -opencl_options to obtain
the best performance for the OpenCL accelerated code.

Typical usage to use the fastest OpenCL device involve the
following steps.

Run the command:

ffmpeg -opencl_bench

Note down the platform ID (pidx) and device ID (didx) of the first
i.e. fastest device in the list. Select the platform and device
using the command:

ffmpeg -opencl_options platform_idx=<pidx>:device_idx=<didx> ...

-opencl_options options (global)
Set OpenCL environment options. This option is only available when
FFmpeg has been compiled with "--enable-opencl".

options must be a list of key=value option pairs separated by ':'.
See the ``OpenCL Options'' section in the ffmpeg-utils manual for
the list of supported options.

AVOptions
These options are provided directly by the libavformat, libavdevice and
libavcodec libraries. To see the list of available AVOptions, use the
-help option. They are separated into two categories:

generic
These options can be set for any container, codec or device.
Generic options are listed under AVFormatContext options for
containers/devices and under AVCodecContext options for codecs.

private
These options are specific to the given container, device or codec.
Private options are listed under their corresponding
containers/devices/codecs.

For example to write an ID3v2.3 header instead of a default ID3v2.4 to
an MP3 file, use the id3v2_version private option of the MP3 muxer:

ffmpeg -i input.flac -id3v2_version 3 out.mp3

All codec AVOptions are per-stream, and thus a stream specifier should
be attached to them.

Note: the -nooption syntax cannot be used for boolean AVOptions, use
-option 0/-option 1.

Note: the old undocumented way of specifying per-stream AVOptions by
prepending v/a/s to the options name is now obsolete and will be
removed soon.

Main options
-f fmt (input/output)
Force input or output file format. The format is normally auto
detected for input files and guessed from the file extension for
output files, so this option is not needed in most cases.

-i filename (input)
input file name

-y (global)
Overwrite output files without asking.

-n (global)
Do not overwrite output files, and exit immediately if a specified
output file already exists.

-c[:stream_specifier] codec (input/output,per-stream)
-codec[:stream_specifier] codec (input/output,per-stream)
Select an encoder (when used before an output file) or a decoder
(when used before an input file) for one or more streams. codec is
the name of a decoder/encoder or a special value "copy" (output
only) to indicate that the stream is not to be re-encoded.

For example

ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT

encodes all video streams with libx264 and copies all audio
streams.

For each stream, the last matching "c" option is applied, so

ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT

will copy all the streams except the second video, which will be
encoded with libx264, and the 138th audio, which will be encoded
with libvorbis.

-t duration (input/output)
When used as an input option (before "-i"), limit the duration of
data read from the input file.

When used as an output option (before an output filename), stop
writing the output after its duration reaches duration.

duration must be a time duration specification, see the Time
duration section in the ffmpeg-utils(1) manual.

-to and -t are mutually exclusive and -t has priority.

-to position (output)
Stop writing the output at position. position must be a time
duration specification, see the Time duration section in the
ffmpeg-utils(1) manual.

-to and -t are mutually exclusive and -t has priority.

-fs limit_size (output)
Set the file size limit, expressed in bytes.

-ss position (input/output)
When used as an input option (before "-i"), seeks in this input
file to position. Note that in most formats it is not possible to
seek exactly, so ffmpeg will seek to the closest seek point before
position. When transcoding and -accurate_seek is enabled (the
default), this extra segment between the seek point and position
will be decoded and discarded. When doing stream copy or when
-noaccurate_seek is used, it will be preserved.

When used as an output option (before an output filename), decodes
but discards input until the timestamps reach position.

position must be a time duration specification, see the Time
duration section in the ffmpeg-utils(1) manual.

-sseof position (input/output)
Like the "-ss" option but relative to the "end of file". That is
negative values are earlier in the file, 0 is at EOF.

-itsoffset offset (input)
Set the input time offset.

offset must be a time duration specification, see the Time duration
section in the ffmpeg-utils(1) manual.

The offset is added to the timestamps of the input files.
Specifying a positive offset means that the corresponding streams
are delayed by the time duration specified in offset.

-timestamp date (output)
Set the recording timestamp in the container.

date must be a date specification, see the Date section in the
ffmpeg-utils(1) manual.

-metadata[:metadata_specifier] key=value (output,per-metadata)
Set a metadata key/value pair.

An optional metadata_specifier may be given to set metadata on
streams or chapters. See "-map_metadata" documentation for details.

This option overrides metadata set with "-map_metadata". It is also
possible to delete metadata by using an empty value.

For example, for setting the title in the output file:

ffmpeg -i in.avi -metadata title="my title" out.flv

To set the language of the first audio stream:

ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT

-target type (output)
Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50"). type
may be prefixed with "pal-", "ntsc-" or "film-" to use the
corresponding standard. 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 (output)
Set the number of data frames to output. This is an alias for
"-frames:d".

-frames[:stream_specifier] framecount (output,per-stream)
Stop writing to the stream after framecount frames.

-q[:stream_specifier] q (output,per-stream)
-qscale[:stream_specifier] q (output,per-stream)
Use fixed quality scale (VBR). The meaning of q/qscale is codec-
dependent. If qscale is used without a stream_specifier then it
applies only to the video stream, this is to maintain compatibility
with previous behavior and as specifying the same codec specific
value to 2 different codecs that is audio and video generally is
not what is intended when no stream_specifier is used.

-filter[:stream_specifier] filtergraph (output,per-stream)
Create the filtergraph specified by filtergraph and use it to
filter the stream.

filtergraph is a description of the filtergraph to apply to the
stream, and must have a single input and a single output of the
same type of the stream. In the filtergraph, the input is
associated to the label "in", and the output to the label "out".
See the ffmpeg-filters manual for more information about the
filtergraph syntax.

See the -filter_complex option if you want to create filtergraphs
with multiple inputs and/or outputs.

-filter_script[:stream_specifier] filename (output,per-stream)
This option is similar to -filter, the only difference is that its
argument is the name of the file from which a filtergraph
description is to be read.

-pre[:stream_specifier] preset_name (output,per-stream)
Specify the preset for matching stream(s).

-stats (global)
Print encoding progress/statistics. It is on by default, to
explicitly disable it you need to specify "-nostats".

-progress url (global)
Send program-friendly progress information to url.

Progress information is written approximately every second and at
the end of the encoding process. It is made of "key=value" lines.
key consists of only alphanumeric characters. The last key of a
sequence of progress information is always "progress".

-stdin
Enable interaction on standard input. On by default unless standard
input is used as an input. To explicitly disable interaction you
need to specify "-nostdin".

Disabling interaction on standard input is useful, for example, if
ffmpeg is in the background process group. Roughly the same result
can be achieved with "ffmpeg ... < /dev/null" but it requires a
shell.

-debug_ts (global)
Print timestamp information. It is off by default. This option is
mostly useful for testing and debugging purposes, and the output
format may change from one version to another, so it should not be
employed by portable scripts.

See also the framemd5 muxer.

mov, mp4, ismv
MOV/MP4/ISMV (Smooth Streaming) muxer.

The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4 file
has all the metadata about all packets stored in one location (written
at the end of the file, it can be moved to the start for better
playback by adding faststart to the movflags, or using the qt-faststart
tool). A fragmented file consists of a number of fragments, where
packets and metadata about these packets are stored together. Writing a
fragmented file has the advantage that the file is decodable even if
the writing is interrupted (while a normal MOV/MP4 is undecodable if it
is not properly finished), and it requires less memory when writing
very long files (since writing normal MOV/MP4 files stores info about
every single packet in memory until the file is closed). The downside
is that it is less compatible with other applications.

Options

Fragmentation is enabled by setting one of the AVOptions that define
how to cut the file into fragments:

-moov_size bytes
Reserves space for the moov atom at the beginning of the file
instead of placing the moov atom at the end. If the space reserved
is insufficient, muxing will fail.

-movflags frag_keyframe
Start a new fragment at each video keyframe.

-frag_duration duration
Create fragments that are duration microseconds long.

-frag_size size
Create fragments that contain up to size bytes of payload data.

-movflags frag_custom
Allow the caller to manually choose when to cut fragments, by
calling "av_write_frame(ctx, NULL)" to write a fragment with the
packets written so far. (This is only useful with other
applications integrating libavformat, not from ffmpeg.)

-min_frag_duration duration
Don't create fragments that are shorter than duration microseconds
long.

If more than one condition is specified, fragments are cut when one of
the specified conditions is fulfilled. The exception to this is
"-min_frag_duration", which has to be fulfilled for any of the other
conditions to apply.

Additionally, the way the output file is written can be adjusted
through a few other options:

-movflags empty_moov
Write an initial moov atom directly at the start of the file,
without describing any samples in it. Generally, an mdat/moov pair
is written at the start of the file, as a normal MOV/MP4 file,
containing only a short portion of the file. With this option set,
there is no initial mdat atom, and the moov atom only describes the
tracks but has a zero duration.

This option is implicitly set when writing ismv (Smooth Streaming)
files.

-movflags separate_moof
Write a separate moof (movie fragment) atom for each track.
Normally, packets for all tracks are written in a moof atom (which
is slightly more efficient), but with this option set, the muxer
writes one moof/mdat pair for each track, making it easier to
separate tracks.

This option is implicitly set when writing ismv (Smooth Streaming)
files.

-movflags faststart
Run a second pass moving the index (moov atom) to the beginning of
the file. This operation can take a while, and will not work in
various situations such as fragmented output, thus it is not
enabled by default.

-movflags rtphint
Add RTP hinting tracks to the output file.

-movflags disable_chpl
Disable Nero chapter markers (chpl atom). Normally, both Nero
chapters and a QuickTime chapter track are written to the file.
With this option set, only the QuickTime chapter track will be
written. Nero chapters can cause failures when the file is
reprocessed with certain tagging programs, like mp3Tag 2.61a and
iTunes 11.3, most likely other versions are affected as well.

-movflags omit_tfhd_offset
Do not write any absolute base_data_offset in tfhd atoms. This
avoids tying fragments to absolute byte positions in the
file/streams.

-movflags default_base_moof
Similarly to the omit_tfhd_offset, this flag avoids writing the
absolute base_data_offset field in tfhd atoms, but does so by using
the new default-base-is-moof flag instead. This flag is new from
14496-12:2012. This may make the fragments easier to parse in
certain circumstances (avoiding basing track fragment location
calculations on the implicit end of the previous track fragment).

Example

Smooth Streaming content can be pushed in real time to a publishing
point on IIS with this muxer. Example:

ffmpeg -re <<normal input/transcoding options>> -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)

Audible AAX

Audible AAX files are encrypted M4B files, and they can be decrypted by
specifying a 4 byte activation secret.

ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4

mp3
The MP3 muxer writes a raw MP3 stream with the following optional
features:

· An ID3v2 metadata header at the beginning (enabled by default).
Versions 2.3 and 2.4 are supported, the "id3v2_version" private
option controls which one is used (3 or 4). Setting "id3v2_version"
to 0 disables the ID3v2 header completely.

The muxer supports writing attached pictures (APIC frames) to the
ID3v2 header. The pictures are supplied to the muxer in form of a
video stream with a single packet. There can be any number of those
streams, each will correspond to a single APIC frame. The stream
metadata tags title and comment map to APIC description and picture
type respectively. See <http://id3.org/id3v2.4.0-frames> for
allowed picture types.

Note that the APIC frames must be written at the beginning, so the
muxer will buffer the audio frames until it gets all the pictures.
It is therefore advised to provide the pictures as soon as possible
to avoid excessive buffering.

· A Xing/LAME frame right after the ID3v2 header (if present). It is
enabled by default, but will be written only if the output is
seekable. The "write_xing" private option can be used to disable
it. The frame contains various information that may be useful to
the decoder, like the audio duration or encoder delay.

· A legacy ID3v1 tag at the end of the file (disabled by default). It
may be enabled with the "write_id3v1" private option, but as its
capabilities are very limited, its usage is not recommended.

Examples:

Write an mp3 with an ID3v2.3 header and an ID3v1 footer:

ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3

To attach a picture to an mp3 file select both the audio and the
picture stream with "map":

ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3

Write a "clean" MP3 without any extra features:

ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3

mpegts
MPEG transport stream muxer.

This muxer implements ISO 13818-1 and part of ETSI EN 300 468.

The recognized metadata settings in mpegts muxer are "service_provider"
and "service_name". If they are not set the default for
"service_provider" is "FFmpeg" and the default for "service_name" is
"Service01".

Options

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_service_type number
Set the program service_type (default digital_tv), see below a list
of pre defined values.

-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).

-mpegts_m2ts_mode number
Enable m2ts mode if set to 1. Default value is -1 which disables
m2ts mode.

-muxrate number
Set a constant muxrate (default VBR).

-pcr_period numer
Override the default PCR retransmission time (default 20ms),
ignored if variable muxrate is selected.

pat_period number
Maximal time in seconds between PAT/PMT tables.

sdt_period number
Maximal time in seconds between SDT tables.

-pes_payload_size number
Set minimum PES packet payload in bytes.

-mpegts_flags flags
Set flags (see below).

-mpegts_copyts number
Preserve original timestamps, if value is set to 1. Default value
is -1, which results in shifting timestamps so that they start from
0.

-tables_version number
Set PAT, PMT and SDT version (default 0, valid values are from 0 to
31, inclusively). This option allows updating stream structure so
that standard consumer may detect the change. To do so, reopen
output AVFormatContext (in case of API usage) or restart ffmpeg
instance, cyclically changing tables_version value:

ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
...
ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
...

Option mpegts_service_type accepts the following values:

hex_value
Any hexdecimal value between 0x01 to 0xff as defined in ETSI 300
468.

digital_tv
Digital TV service.

digital_radio
Digital Radio service.

teletext
Teletext service.

advanced_codec_digital_radio
Advanced Codec Digital Radio service.

mpeg2_digital_hdtv
MPEG2 Digital HDTV service.

advanced_codec_digital_sdtv
Advanced Codec Digital SDTV service.

advanced_codec_digital_hdtv
Advanced Codec Digital HDTV service.

Option mpegts_flags may take a set of such flags:

resend_headers
Reemit PAT/PMT before writing the next packet.

latm
Use LATM packetization for AAC.

pat_pmt_at_frames
Reemit PAT and PMT at each video frame.

Example

ffmpeg -i file.mpg -c 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

mxf, mxf_d10
MXF muxer.

Options

The muxer options are:

store_user_comments bool
Set if user comments should be stored if available or never. IRT
D-10 does not allow user comments. The default is thus to write
them for mxf but not for mxf_d10

null
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 -

nut
-syncpoints flags
Change the syncpoint usage in nut:

default use the normal low-overhead seeking aids.
none do not use the syncpoints at all, reducing the overhead but
making the stream non-seekable;
Use of this option is not recommended, as the resulting files are very damage
sensitive and seeking is not possible. Also in general the overhead from
syncpoints is negligible. Note, -C<write_index> 0 can be used to disable
all growing data tables, allowing to mux endless streams with limited memory
and without these disadvantages.

timestamped extend the syncpoint with a wallclock field.

The none and timestamped flags are experimental.

-write_index bool
Write index at the end, the default is to write an index.

ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor

ogg
Ogg container muxer.

-page_duration duration
Preferred page duration, in microseconds. The muxer will attempt to
create pages that are approximately duration microseconds long.
This allows the user to compromise between seek granularity and
container overhead. The default is 1 second. A value of 0 will fill
all segments, making pages as large as possible. A value of 1 will
effectively use 1 packet-per-page in most situations, giving a
small seek granularity at the cost of additional container
overhead.

-serial_offset value
Serial value from which to set the streams serial number. Setting
it to different and sufficiently large values ensures that the
produced ogg files can be safely chained.

segment, stream_segment, ssegment
Basic stream segmenter.

This muxer outputs streams to a number of separate files of nearly
fixed duration. Output filename pattern can be set in a fashion similar
to image2, or by using a "strftime" template if the strftime option is
enabled.

"stream_segment" is a variant of the muxer used to write to streaming
output formats, i.e. which do not require global headers, and is
recommended for outputting e.g. to MPEG transport stream segments.
"ssegment" is a shorter alias for "stream_segment".

Every segment starts with a keyframe of the selected reference stream,
which is set through the reference_stream option.

Note that if you want accurate splitting for a video file, you need to
make the input key frames correspond to the exact splitting times
expected by the segmenter, or the segment muxer will start the new
segment with the key frame found next after the specified start time.

The segment muxer works best with a single constant frame rate video.

Optionally it can generate a list of the created segments, by setting
the option segment_list. The list type is specified by the
segment_list_type option. The entry filenames in the segment list are
set by default to the basename of the corresponding segment files.

See also the hls muxer, which provides a more specific implementation
for HLS segmentation.

Options

The segment muxer supports the following options:

reference_stream specifier
Set the reference stream, as specified by the string specifier. If
specifier is set to "auto", the reference is chosen automatically.
Otherwise it must be a stream specifier (see the ``Stream
specifiers'' chapter in the ffmpeg manual) which specifies the
reference stream. The default value is "auto".

segment_format format
Override the inner container format, by default it is guessed by
the filename extension.

segment_format_options options_list
Set output format options using a :-separated list of key=value
parameters. Values containing the ":" special character must be
escaped.

segment_list name
Generate also a listfile named name. If not specified no listfile
is generated.

segment_list_flags flags
Set flags affecting the segment list generation.

It currently supports the following flags:

cache
Allow caching (only affects M3U8 list files).

live
Allow live-friendly file generation.

segment_list_size size
Update the list file so that it contains at most size segments. If
0 the list file will contain all the segments. Default value is 0.

segment_list_entry_prefix prefix
Prepend prefix to each entry. Useful to generate absolute paths.
By default no prefix is applied.

segment_list_type type
Select the listing format.

The following values are recognized:

flat
Generate a flat list for the created segments, one segment per
line.

csv, ext
Generate a list for the created segments, one segment per line,
each line matching the format (comma-separated values):

<segment_filename>,<segment_start_time>,<segment_end_time>

segment_filename is the name of the output file generated by
the muxer according to the provided pattern. CSV escaping
(according to RFC4180) is applied if required.

segment_start_time and segment_end_time specify the segment
start and end time expressed in seconds.

A list file with the suffix ".csv" or ".ext" will auto-select
this format.

ext is deprecated in favor or csv.

ffconcat
Generate an ffconcat file for the created segments. The
resulting file can be read using the FFmpeg concat demuxer.

A list file with the suffix ".ffcat" or ".ffconcat" will auto-
select this format.

m3u8
Generate an extended M3U8 file, version 3, compliant with
<http://tools.ietf.org/id/draft-pantos-http-live-streaming>.

A list file with the suffix ".m3u8" will auto-select this
format.

If not specified the type is guessed from the list file name
suffix.

segment_time time
Set segment duration to time, the value must be a duration
specification. Default value is "2". See also the segment_times
option.

Note that splitting may not be accurate, unless you force the
reference stream key-frames at the given time. See the introductory
notice and the examples below.

segment_atclocktime 1|0
If set to "1" split at regular clock time intervals starting from
00:00 o'clock. The time value specified in segment_time is used for
setting the length of the splitting interval.

For example with segment_time set to "900" this makes it possible
to create files at 12:00 o'clock, 12:15, 12:30, etc.

Default value is "0".

segment_time_delta delta
Specify the accuracy time when selecting the start time for a
segment, expressed as a duration specification. Default value is
"0".

When delta is specified a key-frame will start a new segment if its
PTS satisfies the relation:

PTS >= start_time - time_delta

This option is useful when splitting video content, which is always
split at GOP boundaries, in case a key frame is found just before
the specified split time.

In particular may be used in combination with the ffmpeg option
force_key_frames. The key frame times specified by force_key_frames
may not be set accurately because of rounding issues, with the
consequence that a key frame time may result set just before the
specified time. For constant frame rate videos a value of
1/(2*frame_rate) should address the worst case mismatch between the
specified time and the time set by force_key_frames.

segment_times times
Specify a list of split points. times contains a list of comma
separated duration specifications, in increasing order. See also
the segment_time option.

segment_frames frames
Specify a list of split video frame numbers. frames contains a list
of comma separated integer numbers, in increasing order.

This option specifies to start a new segment whenever a reference
stream key frame is found and the sequential number (starting from
0) of the frame is greater or equal to the next value in the list.

segment_wrap limit
Wrap around segment index once it reaches limit.

segment_start_number number
Set the sequence number of the first segment. Defaults to 0.

strftime 1|0
Use the "strftime" function to define the name of the new segments
to write. If this is selected, the output segment name must contain
a "strftime" function template. Default value is 0.

break_non_keyframes 1|0
If enabled, allow segments to start on frames other than keyframes.
This improves behavior on some players when the time between
keyframes is inconsistent, but may make things worse on others, and
can cause some oddities during seeking. Defaults to 0.

reset_timestamps 1|0
Reset timestamps at the begin of each segment, so that each segment
will start with near-zero timestamps. It is meant to ease the
playback of the generated segments. May not work with some
combinations of muxers/codecs. It is set to 0 by default.

initial_offset offset
Specify timestamp offset to apply to the output packet timestamps.
The argument must be a time duration specification, and defaults to
0.

Examples

· Remux the content of file in.mkv to a list of segments out-000.nut,
out-001.nut, etc., and write the list of generated segments to
out.list:

ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut

· Segment input and set output format options for the output
segments:

ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4

· Segment the input file according to the split points specified by
the segment_times option:

ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut

· Use the ffmpeg force_key_frames option to force key frames in the
input at the specified location, together with the segment option
segment_time_delta to account for possible roundings operated when
setting key frame times.

ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
-f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut

In order to force key frames on the input file, transcoding is
required.

· Segment the input file by splitting the input file according to the
frame numbers sequence specified with the segment_frames option:

ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut

· Convert the in.mkv to TS segments using the "libx264" and "libfaac"
encoders:

ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts

· Segment the input file, and create an M3U8 live playlist (can be
used as live HLS source):

ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
-segment_list_flags +live -segment_time 10 out%03d.mkv

smoothstreaming
Smooth Streaming muxer generates a set of files (Manifest, chunks)
suitable for serving with conventional web server.

window_size
Specify the number of fragments kept in the manifest. Default 0
(keep all).

extra_window_size
Specify the number of fragments kept outside of the manifest before
removing from disk. Default 5.

lookahead_count
Specify the number of lookahead fragments. Default 2.

min_frag_duration
Specify the minimum fragment duration (in microseconds). Default
5000000.

remove_at_exit
Specify whether to remove all fragments when finished. Default 0
(do not remove).

tee
The tee muxer can be used to write the same data to several files or
any other kind of muxer. It can be used, for example, to both stream a
video to the network and save it to disk at the same time.

It is different from specifying several outputs to the ffmpeg command-
line tool because the audio and video data will be encoded only once
with the tee muxer; encoding can be a very expensive process. It is not
useful when using the libavformat API directly because it is then
possible to feed the same packets to several muxers directly.

The slave outputs are specified in the file name given to the muxer,
separated by '|'. If any of the slave name contains the '|' separator,
leading or trailing spaces or any special character, it must be escaped
(see the "Quoting and escaping" section in the ffmpeg-utils(1) manual).

Muxer options can be specified for each slave by prepending them as a
list of key=value pairs separated by ':', between square brackets. If
the options values contain a special character or the ':' separator,
they must be escaped; note that this is a second level escaping.

The following special options are also recognized:

f Specify the format name. Useful if it cannot be guessed from the
output name suffix.

bsfs[/spec]
Specify a list of bitstream filters to apply to the specified
output.

It is possible to specify to which streams a given bitstream filter
applies, by appending a stream specifier to the option separated by
"/". spec must be a stream specifier (see Format stream
specifiers). If the stream specifier is not specified, the
bitstream filters will be applied to all streams in the output.

Several bitstream filters can be specified, separated by ",".

select
Select the streams that should be mapped to the slave output,
specified by a stream specifier. If not specified, this defaults to
all the input streams.

Examples

· Encode something and both archive it in a WebM file and stream it
as MPEG-TS over UDP (the streams need to be explicitly mapped):

ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
"archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"

· Use ffmpeg to encode the input, and send the output to three
different destinations. The "dump_extra" bitstream filter is used
to add extradata information to all the output video keyframes
packets, as requested by the MPEG-TS format. The select option is
applied to out.aac in order to make it contain only audio packets.

ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"

· As below, but select only stream "a:1" for the audio output. Note
that a second level escaping must be performed, as ":" is a special
character used to separate options.

ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"

Note: some codecs may need different options depending on the output
format; the auto-detection of this can not work with the tee muxer. The
main example is the global_header flag.

webm_dash_manifest
WebM DASH Manifest muxer.

This muxer implements the WebM DASH Manifest specification to generate
the DASH manifest XML. It also supports manifest generation for DASH
live streams.

For more information see:

· WebM DASH Specification:
<https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>

· ISO DASH Specification:
<http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip>

Options

This muxer supports the following options:

adaptation_sets
This option has the following syntax: "id=x,streams=a,b,c
id=y,streams=d,e" where x and y are the unique identifiers of the
adaptation sets and a,b,c,d and e are the indices of the
corresponding audio and video streams. Any number of adaptation
sets can be added using this option.

live
Set this to 1 to create a live stream DASH Manifest. Default: 0.

chunk_start_index
Start index of the first chunk. This will go in the startNumber
attribute of the SegmentTemplate element in the manifest. Default:
0.

chunk_duration_ms
Duration of each chunk in milliseconds. This will go in the
duration attribute of the SegmentTemplate element in the manifest.
Default: 1000.

utc_timing_url
URL of the page that will return the UTC timestamp in ISO format.
This will go in the value attribute of the UTCTiming element in the
manifest. Default: None.

time_shift_buffer_depth
Smallest time (in seconds) shifting buffer for which any
Representation is guaranteed to be available. This will go in the
timeShiftBufferDepth attribute of the MPD element. Default: 60.

minimum_update_period
Minimum update period (in seconds) of the manifest. This will go in
the minimumUpdatePeriod attribute of the MPD element. Default: 0.

Example

ffmpeg -f webm_dash_manifest -i video1.webm \
-f webm_dash_manifest -i video2.webm \
-f webm_dash_manifest -i audio1.webm \
-f webm_dash_manifest -i audio2.webm \
-map 0 -map 1 -map 2 -map 3 \
-c copy \
-f webm_dash_manifest \
-adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
manifest.xml

webm_chunk
WebM Live Chunk Muxer.

This muxer writes out WebM headers and chunks as separate files which
can be consumed by clients that support WebM Live streams via DASH.

Options

This muxer supports the following options:

chunk_start_index
Index of the first chunk (defaults to 0).

header
Filename of the header where the initialization data will be
written.

audio_chunk_duration
Duration of each audio chunk in milliseconds (defaults to 5000).

Example

ffmpeg -f v4l2 -i /dev/video0 \
-f alsa -i hw:0 \
-map 0:0 \
-c:v libvpx-vp9 \
-s 640x360 -keyint_min 30 -g 30 \
-f webm_chunk \
-header webm_live_video_360.hdr \
-chunk_start_index 1 \
webm_live_video_360_%d.chk \
-map 1:0 \
-c:a libvorbis \
-b:a 128k \
-f webm_chunk \
-header webm_live_audio_128.hdr \
-chunk_start_index 1 \
-audio_chunk_duration 1000 \
webm_live_audio_128_%d.chk

METADATA
FFmpeg 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:

1. A file consists of a header and a number of metadata tags divided
into sections, each on its own line.

2. The header is a ;FFMETADATA string, followed by a version number
(now 1).

3. Metadata tags are of the form key=value

4. Immediately after header follows global metadata

5. After global metadata there may be sections with
per-stream/per-chapter metadata.

6. 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.

7. 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.

8. Empty lines and lines starting with ; or # are ignored.

9. Metadata keys or values containing special characters (=, ;, #, \
and a newline) must be escaped with a backslash \.

10. 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=FFmpeg troll team

[CHAPTER]
TIMEBASE=1/1000
START=0
#chapter ends at 0:01:00
END=60000
title=chapter \#1
[STREAM]
title=multi\
line

By using the ffmetadata muxer and demuxer it is possible to extract
metadata from an input file to an ffmetadata file, and then transcode
the file into an output file with the edited ffmetadata file.

Extracting an ffmetadata file with ffmpeg goes as follows:

ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE

Reinserting edited metadata information from the FFMETADATAFILE file
can be done as:

ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT

PROTOCOLS
Protocols are configured elements in FFmpeg that enable access to
resources that require specific protocols.

When you configure your FFmpeg 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.

async
Asynchronous data filling wrapper for input stream.

Fill data in a background thread, to decouple I/O operation from demux
thread.

async:<URL>
async:http://host/resource
async:cache:http://host/resource

bluray
Read BluRay playlist.

The accepted options are:

angle
BluRay angle

chapter
Start chapter (1...N)

playlist
Playlist to read (BDMV/PLAYLIST/?????.mpls)

Examples:

Read longest playlist from BluRay mounted to /mnt/bluray:

bluray:/mnt/bluray

Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start
from chapter 2:

-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray

cache
Caching wrapper for input stream.

Cache the input stream to temporary file. It brings seeking capability
to live streams.

cache:<URL>

concat
Physical concatenation protocol.

Read and seek from many resources 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.

crypto
AES-encrypted stream reading protocol.

The accepted options are:

key Set the AES decryption key binary block from given hexadecimal
representation.

iv Set the AES decryption initialization vector binary block from
given hexadecimal representation.

Accepted URL formats:

crypto:<URL>
crypto+<URL>

data
Data in-line in the URI. See
<http://en.wikipedia.org/wiki/Data_URI_scheme>.

For example, to convert a GIF file given inline with ffmpeg:

ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png

file
File access protocol.

Read from or write to a file.

A file URL can have the form:

file:<filename>

where filename is the path of the file to read.

An URL that does not have a protocol prefix will be assumed to be a
file URL. Depending on the build, an URL that looks like a Windows path
with the drive letter at the beginning will also be assumed to be a
file URL (usually not the case in builds for unix-like systems).

For example to read from a file input.mpeg with ffmpeg use the command:

ffmpeg -i file:input.mpeg output.mpeg

This protocol accepts the following options:

truncate
Truncate existing files on write, if set to 1. A value of 0
prevents truncating. Default value is 1.

blocksize
Set I/O operation maximum block size, in bytes. Default value is
"INT_MAX", which results in not limiting the requested block size.
Setting this value reasonably low improves user termination request
reaction time, which is valuable for files on slow medium.

ftp
FTP (File Transfer Protocol).

Read from or write to remote resources using FTP protocol.

Following syntax is required.

ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

This protocol accepts the following options.

timeout
Set timeout in microseconds of socket I/O operations used by the
underlying low level operation. By default it is set to -1, which
means that the timeout is not specified.

ftp-anonymous-password
Password used when login as anonymous user. Typically an e-mail
address should be used.

ftp-write-seekable
Control seekability of connection during encoding. If set to 1 the
resource is supposed to be seekable, if set to 0 it is assumed not
to be seekable. Default value is 0.

NOTE: Protocol can be used as output, but it is recommended to not do
it, unless special care is taken (tests, customized server
configuration etc.). Different FTP servers behave in different way
during seek operation. ff* tools may produce incomplete content due to
server limitations.

gopher
Gopher protocol.

hls
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.
The nested protocol is declared by specifying "+proto" after the hls
URI scheme name, where proto is either "file" or "http".

hls+http://host/path/to/remote/resource.m3u8
hls+file://path/to/local/resource.m3u8

Using this protocol is discouraged - the hls demuxer should work just
as well (if not, please report the issues) and is more complete. To
use the hls demuxer instead, simply use the direct URLs to the m3u8
files.

http
HTTP (Hyper Text Transfer Protocol).

This protocol accepts the following options:

seekable
Control seekability of connection. If set to 1 the resource is
supposed to be seekable, if set to 0 it is assumed not to be
seekable, if set to -1 it will try to autodetect if it is seekable.
Default value is -1.

chunked_post
If set to 1 use chunked Transfer-Encoding for posts, default is 1.

content_type
Set a specific content type for the POST messages.

headers
Set custom HTTP headers, can override built in default headers. The
value must be a string encoding the headers.

multiple_requests
Use persistent connections if set to 1, default is 0.

post_data
Set custom HTTP post data.

user-agent
user_agent
Override the User-Agent header. If not specified the protocol will
use a string describing the libavformat build. ("Lavf/<version>")

timeout
Set timeout in microseconds of socket I/O operations used by the
underlying low level operation. By default it is set to -1, which
means that the timeout is not specified.

mime_type
Export the MIME type.

icy If set to 1 request ICY (SHOUTcast) metadata from the server. If
the server supports this, the metadata has to be retrieved by the
application by reading the icy_metadata_headers and
icy_metadata_packet options. The default is 1.

icy_metadata_headers
If the server supports ICY metadata, this contains the ICY-specific
HTTP reply headers, separated by newline characters.

icy_metadata_packet
If the server supports ICY metadata, and icy was set to 1, this
contains the last non-empty metadata packet sent by the server. It
should be polled in regular intervals by applications interested in
mid-stream metadata updates.

cookies
Set the cookies to be sent in future requests. The format of each
cookie is the same as the value of a Set-Cookie HTTP response
field. Multiple cookies can be delimited by a newline character.

offset
Set initial byte offset.

end_offset
Try to limit the request to bytes preceding this offset.

method
When used as a client option it sets the HTTP method for the
request.

When used as a server option it sets the HTTP method that is going
to be expected from the client(s). If the expected and the
received HTTP method do not match the client will be given a Bad
Request response. When unset the HTTP method is not checked for
now. This will be replaced by autodetection in the future.

listen
If set to 1 enables experimental HTTP server. This can be used to
send data when used as an output option, or read data from a client
with HTTP POST when used as an input option. If set to 2 enables
experimental mutli-client HTTP server. This is not yet implemented
in ffmpeg.c or ffserver.c and thus must not be used as a command
line option.

# Server side (sending):
ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>

# Client side (receiving):
ffmpeg -i http://<server>:<port> -c copy somefile.ogg

# Client can also be done with wget:
wget http://<server>:<port> -O somefile.ogg

# Server side (receiving):
ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg

# Client side (sending):
ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>

# Client can also be done with wget:
wget --post-file=somefile.ogg http://<server>:<port>

HTTP Cookies

Some HTTP requests will be denied unless cookie values are passed in
with the request. The cookies option allows these cookies to be
specified. At the very least, each cookie must specify a value along
with a path and domain. HTTP requests that match both the domain and
path will automatically include the cookie value in the HTTP Cookie
header field. Multiple cookies can be delimited by a newline.

The required syntax to play a stream specifying a cookie is:

ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8

Icecast
Icecast protocol (stream to Icecast servers)

This protocol accepts the following options:

ice_genre
Set the stream genre.

ice_name
Set the stream name.

ice_description
Set the stream description.

ice_url
Set the stream website URL.

ice_public
Set if the stream should be public. The default is 0 (not public).

user_agent
Override the User-Agent header. If not specified a string of the
form "Lavf/<version>" will be used.

password
Set the Icecast mountpoint password.

content_type
Set the stream content type. This must be set if it is different
from audio/mpeg.

legacy_icecast
This enables support for Icecast versions < 2.4.0, that do not
support the HTTP PUT method but the SOURCE method.

icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>

mmst
MMS (Microsoft Media Server) protocol over TCP.

mmsh
MMS (Microsoft Media Server) protocol over HTTP.

The required syntax is:

mmsh://<server>[:<port>][/<app>][/<playpath>]

md5
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.

pipe
UNIX pipe access protocol.

Read and write from UNIX pipes.

The accepted syntax is:

pipe:[<number>]

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

This protocol accepts the following options:

Note that some formats (typically MOV), require the output protocol to
be seekable, so they will fail with the pipe output protocol.

rtmp
Real-Time Messaging Protocol.

The Real-Time Messaging Protocol (RTMP) is used for streaming
multimedia content across a TCP/IP network.

The required syntax is:

rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]

The accepted parameters are:

username
An optional username (mostly for publishing).

password
An optional password (mostly for publishing).

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.). You can override the value
parsed from the URI through the "rtmp_app" option, too.

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:". You
can override the value parsed from the URI through the
"rtmp_playpath" option, too.

listen
Act as a server, listening for an incoming connection.

timeout
Maximum time to wait for the incoming connection. Implies listen.

Additionally, the following parameters can be set via command line
options (or in code via "AVOption"s):

rtmp_app
Name of application to connect on the RTMP server. This option
overrides the parameter specified in the URI.

rtmp_buffer
Set the client buffer time in milliseconds. The default is 3000.

rtmp_conn
Extra arbitrary AMF connection parameters, parsed from a string,
e.g. like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0". Each
value is prefixed by a single character denoting the type, B for
Boolean, N for number, S for string, O for object, or Z for null,
followed by a colon. For Booleans the data must be either 0 or 1
for FALSE or TRUE, respectively. Likewise for Objects the data
must be 0 or 1 to end or begin an object, respectively. Data items
in subobjects may be named, by prefixing the type with 'N' and
specifying the name before the value (i.e. "NB:myFlag:1"). This
option may be used multiple times to construct arbitrary AMF
sequences.

rtmp_flashver
Version of the Flash plugin used to run the SWF player. The default
is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0
(compatible; <libavformat version>).)

rtmp_flush_interval
Number of packets flushed in the same request (RTMPT only). The
default is 10.

rtmp_live
Specify that the media is a live stream. No resuming or seeking in
live streams is possible. The default value is "any", which means
the subscriber first tries to play the live stream specified in the
playpath. If a live stream of that name is not found, it plays the
recorded stream. The other possible values are "live" and
"recorded".

rtmp_pageurl
URL of the web page in which the media was embedded. By default no
value will be sent.

rtmp_playpath
Stream identifier to play or to publish. This option overrides the
parameter specified in the URI.

rtmp_subscribe
Name of live stream to subscribe to. By default no value will be
sent. It is only sent if the option is specified or if rtmp_live
is set to live.

rtmp_swfhash
SHA256 hash of the decompressed SWF file (32 bytes).

rtmp_swfsize
Size of the decompressed SWF file, required for SWFVerification.

rtmp_swfurl
URL of the SWF player for the media. By default no value will be
sent.

rtmp_swfverify
URL to player swf file, compute hash/size automatically.

rtmp_tcurl
URL of the target stream. Defaults to proto://host[:port]/app.

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

To publish to a password protected server, passing the playpath and app
names separately:

ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/

rtmpe
Encrypted Real-Time Messaging Protocol.

The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
streaming multimedia content within standard cryptographic primitives,
consisting of Diffie-Hellman key exchange and HMACSHA256, generating a
pair of RC4 keys.

rtmps
Real-Time Messaging Protocol over a secure SSL connection.

The Real-Time Messaging Protocol (RTMPS) is used for streaming
multimedia content across an encrypted connection.

rtmpt
Real-Time Messaging Protocol tunneled through HTTP.

The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
for streaming multimedia content within HTTP requests to traverse
firewalls.

rtmpte
Encrypted Real-Time Messaging Protocol tunneled through HTTP.

The Encrypted Real-Time Messaging Protocol tunneled through HTTP
(RTMPTE) is used for streaming multimedia content within HTTP requests
to traverse firewalls.

rtmpts
Real-Time Messaging Protocol tunneled through HTTPS.

The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is
used for streaming multimedia content within HTTPS requests to traverse
firewalls.

libsmbclient
libsmbclient permits one to manipulate CIFS/SMB network resources.

Following syntax is required.

smb://[[domain:]user[:password@]]server[/share[/path[/file]]]

This protocol accepts the following options.

timeout
Set timeout in miliseconds of socket I/O operations used by the
underlying low level operation. By default it is set to -1, which
means that the timeout is not specified.

truncate
Truncate existing files on write, if set to 1. A value of 0
prevents truncating. Default value is 1.

workgroup
Set the workgroup used for making connections. By default workgroup
is not specified.

For more information see: <http://www.samba.org/>.

libssh
Secure File Transfer Protocol via libssh

Read from or write to remote resources using SFTP protocol.

Following syntax is required.

sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

This protocol accepts the following options.

timeout
Set timeout of socket I/O operations used by the underlying low
level operation. By default it is set to -1, which means that the
timeout is not specified.

truncate
Truncate existing files on write, if set to 1. A value of 0
prevents truncating. Default value is 1.

private_key
Specify the path of the file containing private key to use during
authorization. By default libssh searches for keys in the ~/.ssh/
directory.

Example: Play a file stored on remote server.

ffplay sftp://user:password@server_address:22/home/user/resource.mpeg

librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
Real-Time Messaging Protocol and its variants supported through
librtmp.

Requires the presence of the librtmp headers and library during
configuration. You need to explicitly 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"

rtp
Real-time Transport Protocol.

The required syntax for an RTP URL is:
rtp://hostname[:port][?option=val...]

port specifies the RTP port to use.

The following URL options are supported:

ttl=n
Set the TTL (Time-To-Live) value (for multicast only).

rtcpport=n
Set the remote RTCP port to n.

localrtpport=n
Set the local RTP port to n.

localrtcpport=n'
Set the local RTCP port to n.

pkt_size=n
Set max packet size (in bytes) to n.

connect=0|1
Do a "connect()" on the UDP socket (if set to 1) or not (if set to
0).

sources=ip[,ip]
List allowed source IP addresses.

block=ip[,ip]
List disallowed (blocked) source IP addresses.

write_to_source=0|1
Send packets to the source address of the latest received packet
(if set to 1) or to a default remote address (if set to 0).

localport=n
Set the local RTP port to n.

This is a deprecated option. Instead, localrtpport should be used.

Important notes:

1. If rtcpport is not set the RTCP port will be set to the RTP port
value plus 1.

2. If localrtpport (the local RTP port) is not set any available port
will be used for the local RTP and RTCP ports.

3. If localrtcpport (the local RTCP port) is not set it will be set to
the local RTP port value plus 1.

rtsp
Real-Time Streaming 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 <https://github.com/revmischa/rtsp-server>).

The required syntax for a RTSP url is:

rtsp://<hostname>[:<port>]/<path>

Options can be set on the ffmpeg/ffplay command line, or set in code
via "AVOption"s or in "avformat_open_input".

The following options are supported.

initial_pause
Do not start playing the stream immediately if set to 1. Default
value is 0.

rtsp_transport
Set RTSP transport protocols.

It accepts the following values:

udp Use UDP as lower transport protocol.

tcp Use TCP (interleaving within the RTSP control channel) as lower
transport protocol.

udp_multicast
Use UDP multicast as lower transport protocol.

http
Use HTTP tunneling as lower transport protocol, which is useful
for passing proxies.

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.

rtsp_flags
Set RTSP flags.

The following values are accepted:

filter_src
Accept packets only from negotiated peer address and port.

listen
Act as a server, listening for an incoming connection.

prefer_tcp
Try TCP for RTP transport first, if TCP is available as RTSP
RTP transport.

Default value is none.

allowed_media_types
Set media types to accept from the server.

The following flags are accepted:

video
audio
data

By default it accepts all media types.

min_port
Set minimum local UDP port. Default value is 5000.

max_port
Set maximum local UDP port. Default value is 65000.

timeout
Set maximum timeout (in seconds) to wait for incoming connections.

A value of -1 means infinite (default). This option implies the
rtsp_flags set to listen.

reorder_queue_size
Set number of packets to buffer for handling of reordered packets.

stimeout
Set socket TCP I/O timeout in microseconds.

user-agent
Override User-Agent header. If not specified, it defaults to the
libavformat identifier string.

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). This can be disabled by setting the maximum demuxing delay to
zero (via 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".

Examples

The following examples all make use of the ffplay and ffmpeg tools.

· Watch a stream over UDP, with a max reordering delay of 0.5
seconds:

ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4

· Watch a stream tunneled over HTTP:

ffplay -rtsp_transport http rtsp://server/video.mp4

· 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

· Receive a stream in realtime:

ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>

sap
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.

Muxer

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]

Demuxer

The syntax for a SAP url given to the demuxer is:

sap://[<address>][:<port>]

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:

ffplay sap://

To play back the first stream announced on one the default IPv6 SAP
multicast address:

ffplay sap://[ff0e::2:7ffe]

sctp
Stream Control Transmission Protocol.

The accepted URL syntax is:

sctp://<host>:<port>[?<options>]

The protocol accepts the following options:

listen
If set to any value, listen for an incoming connection. Outgoing
connection is done by default.

max_streams
Set the maximum number of streams. By default no limit is set.

srtp
Secure Real-time Transport Protocol.

The accepted options are:

srtp_in_suite
srtp_out_suite
Select input and output encoding suites.

Supported values:

AES_CM_128_HMAC_SHA1_80
SRTP_AES128_CM_HMAC_SHA1_80
AES_CM_128_HMAC_SHA1_32
SRTP_AES128_CM_HMAC_SHA1_32
srtp_in_params
srtp_out_params
Set input and output encoding parameters, which are expressed by a
base64-encoded representation of a binary block. The first 16 bytes
of this binary block are used as master key, the following 14 bytes
are used as master salt.

subfile
Virtually extract a segment of a file or another stream. The
underlying stream must be seekable.

Accepted options:

start
Start offset of the extracted segment, in bytes.

end End offset of the extracted segment, in bytes.

Examples:

Extract a chapter from a DVD VOB file (start and end sectors obtained
externally and multiplied by 2048):

subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB

Play an AVI file directly from a TAR archive:

subfile,,start,183241728,end,366490624,,:archive.tar

tcp
Transmission Control Protocol.

The required syntax for a TCP url is:

tcp://<hostname>:<port>[?<options>]

options contains a list of &-separated options of the form key=val.

The list of supported options follows.

listen=1|0
Listen for an incoming connection. Default value is 0.

timeout=microseconds
Set raise error timeout, expressed in microseconds.

This option is only relevant in read mode: if no data arrived in
more than this time interval, raise error.

listen_timeout=milliseconds
Set listen timeout, expressed in milliseconds.

The following example shows how to setup a listening TCP connection
with ffmpeg, which is then accessed with ffplay:

ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
ffplay tcp://<hostname>:<port>

tls
Transport Layer Security (TLS) / Secure Sockets Layer (SSL)

The required syntax for a TLS/SSL url is:

tls://<hostname>:<port>[?<options>]

The following parameters can be set via command line options (or in
code via "AVOption"s):

ca_file, cafile=filename
A file containing certificate authority (CA) root certificates to
treat as trusted. If the linked TLS library contains a default this
might not need to be specified for verification to work, but not
all libraries and setups have defaults built in. The file must be
in OpenSSL PEM format.

tls_verify=1|0
If enabled, try to verify the peer that we are communicating with.
Note, if using OpenSSL, this currently only makes sure that the
peer certificate is signed by one of the root certificates in the
CA database, but it does not validate that the certificate actually
matches the host name we are trying to connect to. (With GnuTLS,
the host name is validated as well.)

This is disabled by default since it requires a CA database to be
provided by the caller in many cases.

cert_file, cert=filename
A file containing a certificate to use in the handshake with the
peer. (When operating as server, in listen mode, this is more
often required by the peer, while client certificates only are
mandated in certain setups.)

key_file, key=filename
A file containing the private key for the certificate.

listen=1|0
If enabled, listen for connections on the provided port, and assume
the server role in the handshake instead of the client role.

Example command lines:

To create a TLS/SSL server that serves an input stream.

ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>

To play back a stream from the TLS/SSL server using ffplay:

ffplay tls://<hostname>:<port>

udp
User Datagram Protocol.

The required syntax for an UDP URL is:

udp://<hostname>:<port>[?<options>]

options contains a list of &-separated options of the form key=val.

In case threading is enabled on the system, a circular buffer is used
to store the incoming data, which allows one to reduce loss of data due
to UDP socket buffer overruns. The fifo_size and overrun_nonfatal
options are related to this buffer.

The list of supported options follows.

buffer_size=size
Set the UDP maximum socket buffer size in bytes. This is used to
set either the receive or send buffer size, depending on what the
socket is used for. Default is 64KB. See also fifo_size.

localport=port
Override the local UDP port to bind with.

localaddr=addr
Choose the local IP address. This is useful e.g. if sending
multicast and the host has multiple interfaces, where the user can
choose which interface to send on by specifying the IP address of
that interface.

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.

sources=address[,address]
Only receive packets sent to the multicast group from one of the
specified sender IP addresses.

block=address[,address]
Ignore packets sent to the multicast group from the specified
sender IP addresses.

fifo_size=units
Set the UDP receiving circular buffer size, expressed as a number
of packets with size of 188 bytes. If not specified defaults to
7*4096.

overrun_nonfatal=1|0
Survive in case of UDP receiving circular buffer overrun. Default
value is 0.

timeout=microseconds
Set raise error timeout, expressed in microseconds.

This option is only relevant in read mode: if no data arrived in
more than this time interval, raise error.

broadcast=1|0
Explicitly allow or disallow UDP broadcasting.

Note that broadcasting may not work properly on networks having a
broadcast storm protection.

Examples

· Use ffmpeg to stream over UDP to a remote endpoint:

ffmpeg -i <input> -f <format> udp://<hostname>:<port>

· Use ffmpeg 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

· Use ffmpeg to receive over UDP from a remote endpoint:

ffmpeg -i udp://[<multicast-address>]:<port> ...

unix
Unix local socket

The required syntax for a Unix socket URL is:

unix://<filepath>

The following parameters can be set via command line options (or in
code via "AVOption"s):

timeout
Timeout in ms.

listen
Create the Unix socket in listening mode.

DEVICE OPTIONS
The libavdevice library provides the same interface as libavformat.
Namely, an input device is considered like a demuxer, and an output
device like a muxer, and the interface and generic device options are
the same provided by libavformat (see the ffmpeg-formats manual).

In addition each input or output device may support so-called private
options, which are specific for that component.

Options may be set by specifying -option value in the FFmpeg tools, or
by setting the value explicitly in the device "AVFormatContext" options
or using the libavutil/opt.h API for programmatic use.

INPUT DEVICES
Input devices are configured elements in FFmpeg which enable accessing
the data coming from a multimedia device attached to your system.

When you configure your FFmpeg 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 "-devices" of the ff* tools will display the list of
supported input devices.

A description of the currently available input devices follows.

alsa
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:

hw:<CARD>[,<DEV>[,<SUBDEV>]]

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>

Options

sample_rate
Set the sample rate in Hz. Default is 48000.

channels
Set the number of channels. Default is 2.

avfoundation
AVFoundation input device.

AVFoundation is the currently recommended framework by Apple for
streamgrabbing on OSX >= 10.7 as well as on iOS. The older QTKit
framework has been marked deprecated since OSX version 10.7.

The input filename has to be given in the following syntax:

-i "[[VIDEO]:[AUDIO]]"

The first entry selects the video input while the latter selects the
audio input. The stream has to be specified by the device name or the
device index as shown by the device list. Alternatively, the video
and/or audio input device can be chosen by index using the

B<-video_device_index E<lt>INDEXE<gt>>

and/or

B<-audio_device_index E<lt>INDEXE<gt>>

, overriding any device name or index given in the input filename.

All available devices can be enumerated by using -list_devices true,
listing all device names and corresponding indices.

There are two device name aliases:

"default"
Select the AVFoundation default device of the corresponding type.

"none"
Do not record the corresponding media type. This is equivalent to
specifying an empty device name or index.

Options

AVFoundation supports the following options:

-list_devices <TRUE|FALSE>
If set to true, a list of all available input devices is given
showing all device names and indices.

-video_device_index <INDEX>
Specify the video device by its index. Overrides anything given in
the input filename.

-audio_device_index <INDEX>
Specify the audio device by its index. Overrides anything given in
the input filename.

-pixel_format <FORMAT>
Request the video device to use a specific pixel format. If the
specified format is not supported, a list of available formats is
given und the first one in this list is used instead. Available
pixel formats are: "monob, rgb555be, rgb555le, rgb565be, rgb565le,
rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16,
yuv422p10, yuv444p10,
yuv420p, nv12, yuyv422, gray"

-framerate
Set the grabbing frame rate. Default is "ntsc", corresponding to a
frame rate of "30000/1001".

-video_size
Set the video frame size.

-capture_cursor
Capture the mouse pointer. Default is 0.

-capture_mouse_clicks
Capture the screen mouse clicks. Default is 0.

Examples

· Print the list of AVFoundation supported devices and exit:

$ ffmpeg -f avfoundation -list_devices true -i ""

· Record video from video device 0 and audio from audio device 0 into
out.avi:

$ ffmpeg -f avfoundation -i "0:0" out.avi

· Record video from video device 2 and audio from audio device 1 into
out.avi:

$ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi

· Record video from the system default video device using the pixel
format bgr0 and do not record any audio into out.avi:

$ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi

bktr
BSD video input device.

Options

framerate
Set the frame rate.

video_size
Set the video frame size. Default is "vga".

standard
Available values are:

pal
ntsc
secam
paln
palm
ntscj

decklink
The decklink input device provides capture capabilities for Blackmagic
DeckLink devices.

To enable this input device, you need the Blackmagic DeckLink SDK and
you need to configure with the appropriate "--extra-cflags" and
"--extra-ldflags". On Windows, you need to run the IDL files through
widl.

DeckLink is very picky about the formats it supports. Pixel format is
uyvy422 or v210, framerate and video size must be determined for your
device with -list_formats 1. Audio sample rate is always 48 kHz and the
number of channels can be 2, 8 or 16.

Options

list_devices
If set to true, print a list of devices and exit. Defaults to
false.

list_formats
If set to true, print a list of supported formats and exit.
Defaults to false.

bm_v210
If set to 1, video is captured in 10 bit v210 instead of uyvy422.
Not all Blackmagic devices support this option.

Examples

· List input devices:

ffmpeg -f decklink -list_devices 1 -i dummy

· List supported formats:

ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'

· Capture video clip at 1080i50 (format 11):

ffmpeg -f decklink -i 'Intensity Pro@11' -acodec copy -vcodec copy output.avi

· Capture video clip at 1080i50 10 bit:

ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@11' -acodec copy -vcodec copy output.avi

dshow
Windows DirectShow input device.

DirectShow support is enabled when FFmpeg is built with the mingw-w64
project. Currently only audio and video devices are supported.

Multiple devices may be opened as separate inputs, but they may also be
opened on the same input, which should improve synchronism between
them.

The input name should be in the format:

<TYPE>=<NAME>[:<TYPE>=<NAME>]

where TYPE can be either audio or video, and NAME is the device's name
or alternative name..

Options

If no options are specified, the device's defaults are used. If the
device does not support the requested options, it will fail to open.

video_size
Set the video size in the captured video.

framerate
Set the frame rate in the captured video.

sample_rate
Set the sample rate (in Hz) of the captured audio.

sample_size
Set the sample size (in bits) of the captured audio.

channels
Set the number of channels in the captured audio.

list_devices
If set to true, print a list of devices and exit.

list_options
If set to true, print a list of selected device's options and exit.

video_device_number
Set video device number for devices with the same name (starts at
0, defaults to 0).

audio_device_number
Set audio device number for devices with the same name (starts at
0, defaults to 0).

pixel_format
Select pixel format to be used by DirectShow. This may only be set
when the video codec is not set or set to rawvideo.

audio_buffer_size
Set audio device buffer size in milliseconds (which can directly
impact latency, depending on the device). Defaults to using the
audio device's default buffer size (typically some multiple of
500ms). Setting this value too low can degrade performance. See
also
<http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx>

video_pin_name
Select video capture pin to use by name or alternative name.

audio_pin_name
Select audio capture pin to use by name or alternative name.

crossbar_video_input_pin_number
Select video input pin number for crossbar device. This will be
routed to the crossbar device's Video Decoder output pin. Note
that changing this value can affect future invocations (sets a new
default) until system reboot occurs.

crossbar_audio_input_pin_number
Select audio input pin number for crossbar device. This will be
routed to the crossbar device's Audio Decoder output pin. Note
that changing this value can affect future invocations (sets a new
default) until system reboot occurs.

show_video_device_dialog
If set to true, before capture starts, popup a display dialog to
the end user, allowing them to change video filter properties and
configurations manually. Note that for crossbar devices, adjusting
values in this dialog may be needed at times to toggle between PAL
(25 fps) and NTSC (29.97) input frame rates, sizes, interlacing,
etc. Changing these values can enable different scan rates/frame
rates and avoiding green bars at the bottom, flickering scan lines,
etc. Note that with some devices, changing these properties can
also affect future invocations (sets new defaults) until system
reboot occurs.

show_audio_device_dialog
If set to true, before capture starts, popup a display dialog to
the end user, allowing them to change audio filter properties and
configurations manually.

show_video_crossbar_connection_dialog
If set to true, before capture starts, popup a display dialog to
the end user, allowing them to manually modify crossbar pin
routings, when it opens a video device.

show_audio_crossbar_connection_dialog
If set to true, before capture starts, popup a display dialog to
the end user, allowing them to manually modify crossbar pin
routings, when it opens an audio device.

show_analog_tv_tuner_dialog
If set to true, before capture starts, popup a display dialog to
the end user, allowing them to manually modify TV channels and
frequencies.

show_analog_tv_tuner_audio_dialog
If set to true, before capture starts, popup a display dialog to
the end user, allowing them to manually modify TV audio (like mono
vs. stereo, Language A,B or C).

audio_device_load
Load an audio capture filter device from file instead of searching
it by name. It may load additional parameters too, if the filter
supports the serialization of its properties to. To use this an
audio capture source has to be specified, but it can be anything
even fake one.

audio_device_save
Save the currently used audio capture filter device and its
parameters (if the filter supports it) to a file. If a file with
the same name exists it will be overwritten.

video_device_load
Load a video capture filter device from file instead of searching
it by name. It may load additional parameters too, if the filter
supports the serialization of its properties to. To use this a
video capture source has to be specified, but it can be anything
even fake one.

video_device_save
Save the currently used video capture filter device and its
parameters (if the filter supports it) to a file. If a file with
the same name exists it will be overwritten.

Examples

· Print the list of DirectShow supported devices and exit:

$ ffmpeg -list_devices true -f dshow -i dummy

· Open video device Camera:

$ ffmpeg -f dshow -i video="Camera"

· Open second video device with name Camera:

$ ffmpeg -f dshow -video_device_number 1 -i video="Camera"

· Open video device Camera and audio device Microphone:

$ ffmpeg -f dshow -i video="Camera":audio="Microphone"

· Print the list of supported options in selected device and exit:

$ ffmpeg -list_options true -f dshow -i video="Camera"

· Specify pin names to capture by name or alternative name, specify
alternative device name:

$ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{ca465100-deb0-4d59-818f-8c477184adf6}":audio="Microphone"

· Configure a crossbar device, specifying crossbar pins, allow user
to adjust video capture properties at startup:

$ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
-crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"

dv1394
Linux DV 1394 input device.

Options

framerate
Set the frame rate. Default is 25.

standard
Available values are:

pal
ntsc

Default value is "ntsc".

fbdev
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.

See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).

To record from the framebuffer device /dev/fb0 with ffmpeg:

ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi

You can take a single screenshot image with the command:

ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg

Options

framerate
Set the frame rate. Default is 25.

gdigrab
Win32 GDI-based screen capture device.

This device allows you to capture a region of the display on Windows.

There are two options for the input filename:

desktop

title=<window_title>

The first option will capture the entire desktop, or a fixed region of
the desktop. The second option will instead capture the contents of a
single window, regardless of its position on the screen.

For example, to grab the entire desktop using ffmpeg:

ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg

Grab a 640x480 region at position "10,20":

ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg

Grab the contents of the window named "Calculator"

ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg

Options

draw_mouse
Specify whether to draw the mouse pointer. Use the value 0 to not
draw the pointer. Default value is 1.

framerate
Set the grabbing frame rate. Default value is "ntsc", corresponding
to a frame rate of "30000/1001".

show_region
Show grabbed region on screen.

If show_region is specified with 1, then the grabbing region will
be indicated on screen. With this option, it is easy to know what
is being grabbed if only a portion of the screen is grabbed.

Note that show_region is incompatible with grabbing the contents of
a single window.

For example:

ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg

video_size
Set the video frame size. The default is to capture the full screen
if desktop is selected, or the full window size if
title=window_title is selected.

offset_x
When capturing a region with video_size, set the distance from the
left edge of the screen or desktop.

Note that the offset calculation is from the top left corner of the
primary monitor on Windows. If you have a monitor positioned to the
left of your primary monitor, you will need to use a negative
offset_x value to move the region to that monitor.

offset_y
When capturing a region with video_size, set the distance from the
top edge of the screen or desktop.

Note that the offset calculation is from the top left corner of the
primary monitor on Windows. If you have a monitor positioned above
your primary monitor, you will need to use a negative offset_y
value to move the region to that monitor.

iec61883
FireWire DV/HDV input device using libiec61883.

To enable this input device, you need libiec61883, libraw1394 and
libavc1394 installed on your system. Use the configure option
"--enable-libiec61883" to compile with the device enabled.

The iec61883 capture device supports capturing from a video device
connected via IEEE1394 (FireWire), using libiec61883 and the new Linux
FireWire stack (juju). This is the default DV/HDV input method in Linux
Kernel 2.6.37 and later, since the old FireWire stack was removed.

Specify the FireWire port to be used as input file, or "auto" to choose
the first port connected.

Options

dvtype
Override autodetection of DV/HDV. This should only be used if auto
detection does not work, or if usage of a different device type
should be prohibited. Treating a DV device as HDV (or vice versa)
will not work and result in undefined behavior. The values auto,
dv and hdv are supported.

dvbuffer
Set maximum size of buffer for incoming data, in frames. For DV,
this is an exact value. For HDV, it is not frame exact, since HDV
does not have a fixed frame size.

dvguid
Select the capture device by specifying it's GUID. Capturing will
only be performed from the specified device and fails if no device
with the given GUID is found. This is useful to select the input if
multiple devices are connected at the same time. Look at
/sys/bus/firewire/devices to find out the GUIDs.

Examples

· Grab and show the input of a FireWire DV/HDV device.

ffplay -f iec61883 -i auto

· Grab and record the input of a FireWire DV/HDV device, using a
packet buffer of 100000 packets if the source is HDV.

ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg

jack
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 FFmpeg 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/>

Options

channels
Set the number of channels. Default is 2.

lavfi
Libavfilter input virtual device.

This input device reads data from the open output pads of a libavfilter
filtergraph.

For each filtergraph open output, the input device will create a
corresponding stream which is mapped to the generated output. Currently
only video data is supported. The filtergraph is specified through the
option graph.

Options

graph
Specify the filtergraph to use as input. Each video open output
must be labelled by a unique string of the form "outN", where N is
a number starting from 0 corresponding to the mapped input stream
generated by the device. The first unlabelled output is
automatically assigned to the "out0" label, but all the others need
to be specified explicitly.

The suffix "+subcc" can be appended to the output label to create
an extra stream with the closed captions packets attached to that
output (experimental; only for EIA-608 / CEA-708 for now). The
subcc streams are created after all the normal streams, in the
order of the corresponding stream. For example, if there is
"out19+subcc", "out7+subcc" and up to "out42", the stream #43 is
subcc for stream #7 and stream #44 is subcc for stream #19.

If not specified defaults to the filename specified for the input
device.

graph_file
Set the filename of the filtergraph to be read and sent to the
other filters. Syntax of the filtergraph is the same as the one
specified by the option graph.

dumpgraph
Dump graph to stderr.

Examples

· Create a color video stream and play it back with ffplay:

ffplay -f lavfi -graph "color=c=pink [out0]" dummy

· As the previous example, but use filename for specifying the graph
description, and omit the "out0" label:

ffplay -f lavfi color=c=pink

· Create three different video test filtered sources and play them:

ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3

· Read an audio stream from a file using the amovie source and play
it back with ffplay:

ffplay -f lavfi "amovie=test.wav"

· Read an audio stream and a video stream and play it back with
ffplay:

ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"

· Dump decoded frames to images and closed captions to a file
(experimental):

ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin

libcdio
Audio-CD input device based on libcdio.

To enable this input device during configuration you need libcdio
installed on your system. It requires the configure option
"--enable-libcdio".

This device allows playing and grabbing from an Audio-CD.

For example to copy with ffmpeg the entire Audio-CD in /dev/sr0, you
may run the command:

ffmpeg -f libcdio -i /dev/sr0 cd.wav

Options

speed
Set drive reading speed. Default value is 0.

The speed is specified CD-ROM speed units. The speed is set through
the libcdio "cdio_cddap_speed_set" function. On many CD-ROM drives,
specifying a value too large will result in using the fastest
speed.

paranoia_mode
Set paranoia recovery mode flags. It accepts one of the following
values:

disable
verify
overlap
neverskip
full

Default value is disable.

For more information about the available recovery modes, consult
the paranoia project documentation.

libdc1394
IIDC1394 input device, based on libdc1394 and libraw1394.

Requires the configure option "--enable-libdc1394".

openal
The OpenAL input device provides audio capture on all systems with a
working OpenAL 1.1 implementation.

To enable this input device during configuration, you need OpenAL
headers and libraries installed on your system, and need to configure
FFmpeg with "--enable-openal".

OpenAL headers and libraries should be provided as part of your OpenAL
implementation, or as an additional download (an SDK). Depending on
your installation you may need to specify additional flags via the
"--extra-cflags" and "--extra-ldflags" for allowing the build system to
locate the OpenAL headers and libraries.

An incomplete list of OpenAL implementations follows:

Creative
The official Windows implementation, providing hardware
acceleration with supported devices and software fallback. See
<http://openal.org/>.

OpenAL Soft
Portable, open source (LGPL) software implementation. Includes
backends for the most common sound APIs on the Windows, Linux,
Solaris, and BSD operating systems. See
<http://kcat.strangesoft.net/openal.html>.

Apple
OpenAL is part of Core Audio, the official Mac OS X Audio
interface. See
<http://developer.apple.com/technologies/mac/audio-and-video.html>

This device allows one to capture from an audio input device handled
through OpenAL.

You need to specify the name of the device to capture in the provided
filename. If the empty string is provided, the device will
automatically select the default device. You can get the list of the
supported devices by using the option list_devices.

Options

channels
Set the number of channels in the captured audio. Only the values 1
(monaural) and 2 (stereo) are currently supported. Defaults to 2.

sample_size
Set the sample size (in bits) of the captured audio. Only the
values 8 and 16 are currently supported. Defaults to 16.

sample_rate
Set the sample rate (in Hz) of the captured audio. Defaults to
44.1k.

list_devices
If set to true, print a list of devices and exit. Defaults to
false.

Examples

Print the list of OpenAL supported devices and exit:

$ ffmpeg -list_devices true -f openal -i dummy out.ogg

Capture from the OpenAL device DR-BT101 via PulseAudio:

$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg

Capture from the default device (note the empty string '' as filename):

$ ffmpeg -f openal -i '' out.ogg

Capture from two devices simultaneously, writing to two different
files, within the same ffmpeg command:

$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg

Note: not all OpenAL implementations support multiple simultaneous
capture - try the latest OpenAL Soft if the above does not work.

oss
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>

Options

sample_rate
Set the sample rate in Hz. Default is 48000.

channels
Set the number of channels. Default is 2.

pulse
PulseAudio input device.

To enable this output device you need to configure FFmpeg with
"--enable-libpulse".

The filename to provide to the input device is a source device or the
string "default"

To list the PulseAudio source devices and their properties you can
invoke the command pactl list sources.

More information about PulseAudio can be found on
<http://www.pulseaudio.org>.

Options

server
Connect to a specific PulseAudio server, specified by an IP
address. Default server is used when not provided.

name
Specify the application name PulseAudio will use when showing
active clients, by default it is the "LIBAVFORMAT_IDENT" string.

stream_name
Specify the stream name PulseAudio will use when showing active
streams, by default it is "record".

sample_rate
Specify the samplerate in Hz, by default 48kHz is used.

channels
Specify the channels in use, by default 2 (stereo) is set.

frame_size
Specify the number of bytes per frame, by default it is set to
1024.

fragment_size
Specify the minimal buffering fragment in PulseAudio, it will
affect the audio latency. By default it is unset.

wallclock
Set the initial PTS using the current time. Default is 1.

Examples

Record a stream from default device:

ffmpeg -f pulse -i default /tmp/pulse.wav

qtkit
QTKit input device.

The filename passed as input is parsed to contain either a device name
or index. The device index can also be given by using
-video_device_index. A given device index will override any given
device name. If the desired device consists of numbers only, use
-video_device_index to identify it. The default device will be chosen
if an empty string or the device name "default" is given. The
available devices can be enumerated by using -list_devices.

ffmpeg -f qtkit -i "0" out.mpg

ffmpeg -f qtkit -video_device_index 0 -i "" out.mpg

ffmpeg -f qtkit -i "default" out.mpg

ffmpeg -f qtkit -list_devices true -i ""

Options

frame_rate
Set frame rate. Default is 30.

list_devices
If set to "true", print a list of devices and exit. Default is
"false".

video_device_index
Select the video device by index for devices with the same name
(starts at 0).

sndio
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

Options

sample_rate
Set the sample rate in Hz. Default is 48000.

channels
Set the number of channels. Default is 2.

video4linux2, v4l2
Video4Linux2 input video device.

"v4l2" can be used as alias for "video4linux2".

If FFmpeg is built with v4l-utils support (by using the
"--enable-libv4l2" configure option), it is possible to use it with the
"-use_libv4l2" input device option.

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.

Video4Linux2 devices usually support a limited set of widthxheight
sizes and frame rates. You can check which are supported using
-list_formats all for Video4Linux2 devices. Some devices, like TV
cards, support one or more standards. It is possible to list all the
supported standards using -list_standards all.

The time base for the timestamps is 1 microsecond. Depending on the
kernel version and configuration, the timestamps may be derived from
the real time clock (origin at the Unix Epoch) or the monotonic clock
(origin usually at boot time, unaffected by NTP or manual changes to
the clock). The -timestamps abs or -ts abs option can be used to force
conversion into the real time clock.

Some usage examples of the video4linux2 device with ffmpeg and ffplay:

· List supported formats for a video4linux2 device:

ffplay -f video4linux2 -list_formats all /dev/video0

· Grab and show the input of a video4linux2 device:

ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0

· Grab and record the input of a video4linux2 device, leave the frame
rate and size as previously set:

ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg

For more information about Video4Linux, check <http://linuxtv.org/>.

Options

standard
Set the standard. Must be the name of a supported standard. To get
a list of the supported standards, use the list_standards option.

channel
Set the input channel number. Default to -1, which means using the
previously selected channel.

video_size
Set the video frame size. The argument must be a string in the form
WIDTHxHEIGHT or a valid size abbreviation.

pixel_format
Select the pixel format (only valid for raw video input).

input_format
Set the preferred pixel format (for raw video) or a codec name.
This option allows one to select the input format, when several are
available.

framerate
Set the preferred video frame rate.

list_formats
List available formats (supported pixel formats, codecs, and frame
sizes) and exit.

Available values are:

all Show all available (compressed and non-compressed) formats.

raw Show only raw video (non-compressed) formats.

compressed
Show only compressed formats.

list_standards
List supported standards and exit.

Available values are:

all Show all supported standards.

timestamps, ts
Set type of timestamps for grabbed frames.

Available values are:

default
Use timestamps from the kernel.

abs Use absolute timestamps (wall clock).

mono2abs
Force conversion from monotonic to absolute timestamps.

Default value is "default".

use_libv4l2
Use libv4l2 (v4l-utils) conversion functions. Default is 0.

vfwcap
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.

Options

video_size
Set the video frame size.

framerate
Set the grabbing frame rate. Default value is "ntsc", corresponding
to a frame rate of "30000/1001".

x11grab
X11 video input device.

To enable this input device during configuration you need libxcb
installed on your system. It will be automatically detected during
configuration.

Alternatively, the configure option --enable-x11grab exists for legacy
Xlib users.

This device allows one 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 omitted, 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 xdpyinfo 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 -framerate 25 -video_size cif -i :0.0 out.mpg

Grab at position "10,20":

ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg

Options

draw_mouse
Specify whether to draw the mouse pointer. A value of 0 specify not
to draw the pointer. Default value is 1.

follow_mouse
Make the grabbed area follow the mouse. The argument can be
"centered" or a number of pixels PIXELS.

When it is specified with "centered", the grabbing region follows
the mouse pointer and keeps the pointer at the center of region;
otherwise, the region follows only when the mouse pointer reaches
within PIXELS (greater than zero) to the edge of region.

For example:

ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg

To follow only when the mouse pointer reaches within 100 pixels to
edge:

ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg

framerate
Set the grabbing frame rate. Default value is "ntsc", corresponding
to a frame rate of "30000/1001".

show_region
Show grabbed region on screen.

If show_region is specified with 1, then the grabbing region will
be indicated on screen. With this option, it is easy to know what
is being grabbed if only a portion of the screen is grabbed.

region_border
Set the region border thickness if -show_region 1 is used. Range
is 1 to 128 and default is 3 (XCB-based x11grab only).

For example:

ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg

With follow_mouse:

ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg

video_size
Set the video frame size. Default value is "vga".

use_shm
Use the MIT-SHM extension for shared memory. Default value is 1.
It may be necessary to disable it for remote displays (legacy
x11grab only).

grab_x grab_y AVOption

The syntax is:

-grab_x <x_offset> -grab_y <y_offset>

Set the grabbing region coordinates. They are expressed as offset from
the top left corner of the X11 window. The default value is 0.

OUTPUT DEVICES
Output devices are configured elements in FFmpeg that can write
multimedia data to an output device attached to your system.

When you configure your FFmpeg 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 "-devices" of the ff* tools will display the list of enabled
output devices.

A description of the currently available output devices follows.

alsa
ALSA (Advanced Linux Sound Architecture) output device.

Examples

· Play a file on default ALSA device:

ffmpeg -i INPUT -f alsa default

· Play a file on soundcard 1, audio device 7:

ffmpeg -i INPUT -f alsa hw:1,7

caca
CACA output device.

This output device allows one to show a video stream in CACA window.
Only one CACA window is allowed per application, so you can have only
one instance of this output device in an application.

To enable this output device you need to configure FFmpeg with
"--enable-libcaca". libcaca is a graphics library that outputs text
instead of pixels.

For more information about libcaca, check:
<http://caca.zoy.org/wiki/libcaca>

Options

window_title
Set the CACA window title, if not specified default to the filename
specified for the output device.

window_size
Set the CACA window size, can be a string of the form widthxheight
or a video size abbreviation. If not specified it defaults to the
size of the input video.

driver
Set display driver.

algorithm
Set dithering algorithm. Dithering is necessary because the picture
being rendered has usually far more colours than the available
palette. The accepted values are listed with "-list_dither
algorithms".

antialias
Set antialias method. Antialiasing smoothens the rendered image and
avoids the commonly seen staircase effect. The accepted values are
listed with "-list_dither antialiases".

charset
Set which characters are going to be used when rendering text. The
accepted values are listed with "-list_dither charsets".

color
Set color to be used when rendering text. The accepted values are
listed with "-list_dither colors".

list_drivers
If set to true, print a list of available drivers and exit.

list_dither
List available dither options related to the argument. The
argument must be one of "algorithms", "antialiases", "charsets",
"colors".

Examples

· The following command shows the ffmpeg output is an CACA window,
forcing its size to 80x25:

ffmpeg -i INPUT -vcodec rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -

· Show the list of available drivers and exit:

ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -

· Show the list of available dither colors and exit:

ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -

decklink
The decklink output device provides playback capabilities for
Blackmagic DeckLink devices.

To enable this output device, you need the Blackmagic DeckLink SDK and
you need to configure with the appropriate "--extra-cflags" and
"--extra-ldflags". On Windows, you need to run the IDL files through
widl.

DeckLink is very picky about the formats it supports. Pixel format is
always uyvy422, framerate and video size must be determined for your
device with -list_formats 1. Audio sample rate is always 48 kHz.

Options

list_devices
If set to true, print a list of devices and exit. Defaults to
false.

list_formats
If set to true, print a list of supported formats and exit.
Defaults to false.

preroll
Amount of time to preroll video in seconds. Defaults to 0.5.

Examples

· List output devices:

ffmpeg -i test.avi -f decklink -list_devices 1 dummy

· List supported formats:

ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'

· Play video clip:

ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'

· Play video clip with non-standard framerate or video size:

ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'

fbdev
Linux framebuffer output device.

For more detailed information read the file
Documentation/fb/framebuffer.txt included in the Linux source tree.

Options

xoffset
yoffset
Set x/y coordinate of top left corner. Default is 0.

Examples

Play a file on framebuffer device /dev/fb0. Required pixel format
depends on current framebuffer settings.

ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0

See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).

opengl
OpenGL output device.

To enable this output device you need to configure FFmpeg with
"--enable-opengl".

This output device allows one to render to OpenGL context. Context may
be provided by application or default SDL window is created.

When device renders to external context, application must implement
handlers for following messages: "AV_DEV_TO_APP_CREATE_WINDOW_BUFFER" -
create OpenGL context on current thread.
"AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER" - make OpenGL context current.
"AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER" - swap buffers.
"AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER" - destroy OpenGL context.
Application is also required to inform a device about current
resolution by sending "AV_APP_TO_DEV_WINDOW_SIZE" message.

Options

background
Set background color. Black is a default.

no_window
Disables default SDL window when set to non-zero value.
Application must provide OpenGL context and both "window_size_cb"
and "window_swap_buffers_cb" callbacks when set.

window_title
Set the SDL window title, if not specified default to the filename
specified for the output device. Ignored when no_window is set.

window_size
Set preferred window size, can be a string of the form widthxheight
or a video size abbreviation. If not specified it defaults to the
size of the input video, downscaled according to the aspect ratio.
Mostly usable when no_window is not set.

Examples

Play a file on SDL window using OpenGL rendering:

ffmpeg -i INPUT -f opengl "window title"

oss
OSS (Open Sound System) output device.

pulse
PulseAudio output device.

To enable this output device you need to configure FFmpeg with
"--enable-libpulse".

More information about PulseAudio can be found on
<http://www.pulseaudio.org>

Options

server
Connect to a specific PulseAudio server, specified by an IP
address. Default server is used when not provided.

name
Specify the application name PulseAudio will use when showing
active clients, by default it is the "LIBAVFORMAT_IDENT" string.

stream_name
Specify the stream name PulseAudio will use when showing active
streams, by default it is set to the specified output name.

device
Specify the device to use. Default device is used when not
provided. List of output devices can be obtained with command
pactl list sinks.

buffer_size
buffer_duration
Control the size and duration of the PulseAudio buffer. A small
buffer gives more control, but requires more frequent updates.

buffer_size specifies size in bytes while buffer_duration specifies
duration in milliseconds.

When both options are provided then the highest value is used
(duration is recalculated to bytes using stream parameters). If
they are set to 0 (which is default), the device will use the
default PulseAudio duration value. By default PulseAudio set buffer
duration to around 2 seconds.

prebuf
Specify pre-buffering size in bytes. The server does not start with
playback before at least prebuf bytes are available in the buffer.
By default this option is initialized to the same value as
buffer_size or buffer_duration (whichever is bigger).

minreq
Specify minimum request size in bytes. The server does not request
less than minreq bytes from the client, instead waits until the
buffer is free enough to request more bytes at once. It is
recommended to not set this option, which will initialize this to a
value that is deemed sensible by the server.

Examples

Play a file on default device on default server:

ffmpeg -i INPUT -f pulse "stream name"

sdl
SDL (Simple DirectMedia Layer) output device.

This output device allows one to show a video stream in an SDL window.
Only one SDL window is allowed per application, so you can have only
one instance of this output device in an application.

To enable this output device you need libsdl installed on your system
when configuring your build.

For more information about SDL, check: <http://www.libsdl.org/>

Options

window_title
Set the SDL window title, if not specified default to the filename
specified for the output device.

icon_title
Set the name of the iconified SDL window, if not specified it is
set to the same value of window_title.

window_size
Set the SDL window size, can be a string of the form widthxheight
or a video size abbreviation. If not specified it defaults to the
size of the input video, downscaled according to the aspect ratio.

window_fullscreen
Set fullscreen mode when non-zero value is provided. Default value
is zero.

Interactive commands

The window created by the device can be controlled through the
following interactive commands.

q, ESC
Quit the device immediately.

Examples

The following command shows the ffmpeg output is an SDL window, forcing
its size to the qcif format:

ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"

sndio
sndio audio output device.

xv
XV (XVideo) output device.

This output device allows one to show a video stream in a X Window
System window.

Options

display_name
Specify the hardware display name, which determines the display and
communications domain to be used.

The display name or DISPLAY environment variable can be a string in
the format hostname[:number[.screen_number]].

hostname specifies the name of the host machine on which the
display is physically attached. number specifies the number of the
display server on that host machine. screen_number specifies the
screen to be used on that server.

If unspecified, it defaults to the value of the DISPLAY environment
variable.

For example, "dual-headed:0.1" would specify screen 1 of display 0
on the machine named ``dual-headed''.

Check the X11 specification for more detailed information about the
display name format.

window_id
When set to non-zero value then device doesn't create new window,
but uses existing one with provided window_id. By default this
options is set to zero and device creates its own window.

window_size
Set the created window size, can be a string of the form
widthxheight or a video size abbreviation. If not specified it
defaults to the size of the input video. Ignored when window_id is
set.

window_x
window_y
Set the X and Y window offsets for the created window. They are
both set to 0 by default. The values may be ignored by the window
manager. Ignored when window_id is set.

window_title
Set the window title, if not specified default to the filename
specified for the output device. Ignored when window_id is set.

For more information about XVideo see <http://www.x.org/>.

Examples

· Decode, display and encode video input with ffmpeg at the same
time:

ffmpeg -i INPUT OUTPUT -f xv display

· Decode and display the input video to multiple X11 windows:

ffmpeg -i INPUT -f xv normal -vf negate -f xv negated

RESAMPLER OPTIONS
The audio resampler supports the following named options.

Options may be set by specifying -option value in the FFmpeg tools,
option=value for the aresample filter, by setting the value explicitly
in the "SwrContext" options or using the libavutil/opt.h API for
programmatic use.

ich, in_channel_count
Set the number of input channels. Default value is 0. Setting this
value is not mandatory if the corresponding channel layout
in_channel_layout is set.

och, out_channel_count
Set the number of output channels. Default value is 0. Setting this
value is not mandatory if the corresponding channel layout
out_channel_layout is set.

uch, used_channel_count
Set the number of used input channels. Default value is 0. This
option is only used for special remapping.

isr, in_sample_rate
Set the input sample rate. Default value is 0.

osr, out_sample_rate
Set the output sample rate. Default value is 0.

isf, in_sample_fmt
Specify the input sample format. It is set by default to "none".

osf, out_sample_fmt
Specify the output sample format. It is set by default to "none".

tsf, internal_sample_fmt
Set the internal sample format. Default value is "none". This will
automatically be chosen when it is not explicitly set.

icl, in_channel_layout
ocl, out_channel_layout
Set the input/output channel layout.

See the Channel Layout section in the ffmpeg-utils(1) manual for
the required syntax.

clev, center_mix_level
Set the center mix level. It is a value expressed in deciBel, and
must be in the interval [-32,32].

slev, surround_mix_level
Set the surround mix level. It is a value expressed in deciBel, and
must be in the interval [-32,32].

lfe_mix_level
Set LFE mix into non LFE level. It is used when there is a LFE
input but no LFE output. It is a value expressed in deciBel, and
must be in the interval [-32,32].

rmvol, rematrix_volume
Set rematrix volume. Default value is 1.0.

rematrix_maxval
Set maximum output value for rematrixing. This can be used to
prevent clipping vs. preventing volumn reduction A value of 1.0
prevents cliping.

flags, swr_flags
Set flags used by the converter. Default value is 0.

It supports the following individual flags:

res force resampling, this flag forces resampling to be used even
when the input and output sample rates match.

dither_scale
Set the dither scale. Default value is 1.

dither_method
Set dither method. Default value is 0.

Supported values:

rectangular
select rectangular dither

triangular
select triangular dither

triangular_hp
select triangular dither with high pass

lipshitz
select lipshitz noise shaping dither

shibata
select shibata noise shaping dither

low_shibata
select low shibata noise shaping dither

high_shibata
select high shibata noise shaping dither

f_weighted
select f-weighted noise shaping dither

modified_e_weighted
select modified-e-weighted noise shaping dither

improved_e_weighted
select improved-e-weighted noise shaping dither

resampler
Set resampling engine. Default value is swr.

Supported values:

swr select the native SW Resampler; filter options precision and
cheby are not applicable in this case.

soxr
select the SoX Resampler (where available); compensation, and
filter options filter_size, phase_shift, filter_type &
kaiser_beta, are not applicable in this case.

filter_size
For swr only, set resampling filter size, default value is 32.

phase_shift
For swr only, set resampling phase shift, default value is 10, and
must be in the interval [0,30].

linear_interp
Use Linear Interpolation if set to 1, default value is 0.

cutoff
Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must
be a float value between 0 and 1. Default value is 0.97 with swr,
and 0.91 with soxr (which, with a sample-rate of 44100, preserves
the entire audio band to 20kHz).

precision
For soxr only, the precision in bits to which the resampled signal
will be calculated. The default value of 20 (which, with suitable
dithering, is appropriate for a destination bit-depth of 16) gives
SoX's 'High Quality'; a value of 28 gives SoX's 'Very High
Quality'.

cheby
For soxr only, selects passband rolloff none (Chebyshev) & higher-
precision approximation for 'irrational' ratios. Default value is
0.

async
For swr only, simple 1 parameter audio sync to timestamps using
stretching, squeezing, filling and trimming. Setting this to 1 will
enable filling and trimming, larger values represent the maximum
amount in samples that the data may be stretched or squeezed for
each second. Default value is 0, thus no compensation is applied
to make the samples match the audio timestamps.

first_pts
For swr only, assume the first pts should be this value. The time
unit is 1 / sample rate. This allows for padding/trimming at the
start of stream. By default, no assumption is made about the first
frame's expected pts, so no padding or trimming is done. For
example, this could be set to 0 to pad the beginning with silence
if an audio stream starts after the video stream or to trim any
samples with a negative pts due to encoder delay.

min_comp
For swr only, set the minimum difference between timestamps and
audio data (in seconds) to trigger stretching/squeezing/filling or
trimming of the data to make it match the timestamps. The default
is that stretching/squeezing/filling and trimming is disabled
(min_comp = "FLT_MAX").

min_hard_comp
For swr only, set the minimum difference between timestamps and
audio data (in seconds) to trigger adding/dropping samples to make
it match the timestamps. This option effectively is a threshold to
select between hard (trim/fill) and soft (squeeze/stretch)
compensation. Note that all compensation is by default disabled
through min_comp. The default is 0.1.

comp_duration
For swr only, set duration (in seconds) over which data is
stretched/squeezed to make it match the timestamps. Must be a non-
negative double float value, default value is 1.0.

max_soft_comp
For swr only, set maximum factor by which data is
stretched/squeezed to make it match the timestamps. Must be a non-
negative double float value, default value is 0.

matrix_encoding
Select matrixed stereo encoding.

It accepts the following values:

none
select none

dolby
select Dolby

dplii
select Dolby Pro Logic II

Default value is "none".

filter_type
For swr only, select resampling filter type. This only affects
resampling operations.

It accepts the following values:

cubic
select cubic

blackman_nuttall
select Blackman Nuttall Windowed Sinc

kaiser
select Kaiser Windowed Sinc

kaiser_beta
For swr only, set Kaiser Window Beta value. Must be an integer in
the interval [2,16], default value is 9.

output_sample_bits
For swr only, set number of used output sample bits for dithering.
Must be an integer in the interval [0,64], default value is 0,
which means it's not used.

SCALER OPTIONS
The video scaler supports the following named options.

Options may be set by specifying -option value in the FFmpeg tools. For
programmatic use, they can be set explicitly in the "SwsContext"
options or through the libavutil/opt.h API.

sws_flags
Set the scaler flags. This is also used to set the scaling
algorithm. Only a single algorithm should be selected.

It accepts the following values:

fast_bilinear
Select fast bilinear scaling algorithm.

bilinear
Select bilinear scaling algorithm.

bicubic
Select bicubic scaling algorithm.

experimental
Select experimental scaling algorithm.

neighbor
Select nearest neighbor rescaling algorithm.

area
Select averaging area rescaling algorithm.

bicublin
Select bicubic scaling algorithm for the luma component,
bilinear for chroma components.

gauss
Select Gaussian rescaling algorithm.

sinc
Select sinc rescaling algorithm.

lanczos
Select lanczos rescaling algorithm.

spline
Select natural bicubic spline rescaling algorithm.

print_info
Enable printing/debug logging.

accurate_rnd
Enable accurate rounding.

full_chroma_int
Enable full chroma interpolation.

full_chroma_inp
Select full chroma input.

bitexact
Enable bitexact output.

srcw
Set source width.

srch
Set source height.

dstw
Set destination width.

dsth
Set destination height.

src_format
Set source pixel format (must be expressed as an integer).

dst_format
Set destination pixel format (must be expressed as an integer).

src_range
Select source range.

dst_range
Select destination range.

param0, param1
Set scaling algorithm parameters. The specified values are specific
of some scaling algorithms and ignored by others. The specified
values are floating point number values.

sws_dither
Set the dithering algorithm. Accepts one of the following values.
Default value is auto.

auto
automatic choice

none
no dithering

bayer
bayer dither

ed error diffusion dither

a_dither
arithmetic dither, based using addition

x_dither
arithmetic dither, based using xor (more random/less apparent
patterning that a_dither).

alphablend
Set the alpha blending to use when the input has alpha but the
output does not. Default value is none.

uniform_color
Blend onto a uniform background color

checkerboard
Blend onto a checkerboard

none
No blending

FILTERING INTRODUCTION
Filtering in FFmpeg is enabled through the libavfilter library.

In libavfilter, a filter can have multiple inputs and multiple outputs.
To illustrate the sorts of things that are possible, we consider the
following filtergraph.

[main]
input --> split ---------------------> overlay --> output
| ^
|[tmp] [flip]|
+-----> crop --> vflip -------+

This filtergraph splits the input stream in two streams, then sends one
stream through the crop filter and the vflip filter, before merging it
back with the other stream by overlaying it on top. You can use the
following command to achieve this:

ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT

The result will be that the top half of the video is mirrored onto the
bottom half of the output video.

Filters in the same linear chain are separated by commas, and distinct
linear chains of filters are separated by semicolons. In our example,
crop,vflip are in one linear chain, split and overlay are separately in
another. The points where the linear chains join are labelled by names
enclosed in square brackets. In the example, the split filter generates
two outputs that are associated to the labels [main] and [tmp].

The stream sent to the second output of split, labelled as [tmp], is
processed through the crop filter, which crops away the lower half part
of the video, and then vertically flipped. The overlay filter takes in
input the first unchanged output of the split filter (which was
labelled as [main]), and overlay on its lower half the output generated
by the crop,vflip filterchain.

Some filters take in input a list of parameters: they are specified
after the filter name and an equal sign, and are separated from each
other by a colon.

There exist so-called source filters that do not have an audio/video
input, and sink filters that will not have audio/video output.

GRAPH
The graph2dot program included in the FFmpeg tools directory can be
used to parse a filtergraph description and issue a corresponding
textual representation in the dot language.

Invoke the command:

graph2dot -h

to see how to use graph2dot.

You can then pass the dot description to the dot program (from the
graphviz suite of programs) and obtain a graphical representation of
the filtergraph.

For example the sequence of commands:

echo <GRAPH_DESCRIPTION> | \
tools/graph2dot -o graph.tmp && \
dot -Tpng graph.tmp -o graph.png && \
display graph.png

can be used to create and display an image representing the graph
described by the GRAPH_DESCRIPTION string. Note that this string must
be a complete self-contained graph, with its inputs and outputs
explicitly defined. For example if your command line is of the form:

ffmpeg -i infile -vf scale=640:360 outfile

your GRAPH_DESCRIPTION string will need to be of the form:

nullsrc,scale=640:360,nullsink

you may also need to set the nullsrc parameters and add a format filter
in order to simulate a specific input file.

FILTERGRAPH DESCRIPTION
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 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", and a filter with no
output pads is called a "sink".

Filtergraph syntax
A filtergraph has a textual representation, which is recognized by the
-filter/-vf/-af and -filter_complex options in ffmpeg and -vf/-af in
ffplay, and by the "avfilter_graph_parse_ptr()" function defined in
libavfilter/avfilter.h.

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. It may have one of two forms:

· A ':'-separated list of key=value pairs.

· A ':'-separated list of value. In this case, the keys are assumed
to be the option names in the order they are declared. E.g. the
"fade" filter declares three options in this order -- type,
start_frame and nb_frames. Then the parameter list in:0:30 means
that the value in is assigned to the option type, 0 to start_frame
and 30 to nb_frames.

· A ':'-separated list of mixed direct value and long key=value
pairs. The direct value must precede the key=value pairs, and
follow the same constraints order of the previous point. The
following key=value pairs can be set in any preferred order.

If the option value itself is a list of items (e.g. the "format" filter
takes a list of pixel formats), the items in the list are usually
separated by |.

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 one 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 filter description, if the input label of the first filter is not
specified, "in" is assumed; if the output label of the last filter is
not specified, "out" is assumed.

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.

Libavfilter will automatically insert scale filters where format
conversion is required. It is possible to specify swscale flags for
those automatically inserted scalers by prepending "sws_flags=flags;"
to the filtergraph description.

Here is a BNF description of the filtergraph syntax:

<NAME> ::= sequence of alphanumeric characters and '_'
<LINKLABEL> ::= "[" <NAME> "]"
<LINKLABELS> ::= <LINKLABEL> [<LINKLABELS>]
<FILTER_ARGUMENTS> ::= sequence of chars (possibly quoted)
<FILTER> ::= [<LINKLABELS>] <NAME> ["=" <FILTER_ARGUMENTS>] [<LINKLABELS>]
<FILTERCHAIN> ::= <FILTER> [,<FILTERCHAIN>]
<FILTERGRAPH> ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]

Notes on filtergraph escaping
Filtergraph description composition entails several levels of escaping.
See the "Quoting and escaping" section in the ffmpeg-utils(1) manual
for more information about the employed escaping procedure.

A first level escaping affects the content of each filter option value,
which may contain the special character ":" used to separate values, or
one of the escaping characters "\'".

A second level escaping affects the whole filter description, which may
contain the escaping characters "\'" or the special characters "[],;"
used by the filtergraph description.

Finally, when you specify a filtergraph on a shell commandline, you
need to perform a third level escaping for the shell special characters
contained within it.

For example, consider the following string to be embedded in the
drawtext filter description text value:

this is a 'string': may contain one, or more, special characters

This string contains the "'" special escaping character, and the ":"
special character, so it needs to be escaped in this way:

text=this is a \'string\'\: may contain one, or more, special characters

A second level of escaping is required when embedding the filter
description in a filtergraph description, in order to escape all the
filtergraph special characters. Thus the example above becomes:

drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters

(note that in addition to the "\'" escaping special characters, also
"," needs to be escaped).

Finally an additional level of escaping is needed when writing the
filtergraph description in a shell command, which depends on the
escaping rules of the adopted shell. For example, assuming that "\" is
special and needs to be escaped with another "\", the previous string
will finally result in:

-vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"

TIMELINE EDITING
Some filters support a generic enable option. For the filters
supporting timeline editing, this option can be set to an expression
which is evaluated before sending a frame to the filter. If the
evaluation is non-zero, the filter will be enabled, otherwise the frame
will be sent unchanged to the next filter in the filtergraph.

The expression accepts the following values:

t timestamp expressed in seconds, NAN if the input timestamp is
unknown

n sequential number of the input frame, starting from 0

pos the position in the file of the input frame, NAN if unknown

w
h width and height of the input frame if video

Additionally, these filters support an enable command that can be used
to re-define the expression.

Like any other filtering option, the enable option follows the same
rules.

For example, to enable a blur filter (smartblur) from 10 seconds to 3
minutes, and a curves filter starting at 3 seconds:

smartblur = enable='between(t,10,3*60)',
curves = enable='gte(t,3)' : preset=cross_process

AUDIO FILTERS
When you configure your FFmpeg 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.

acrossfade
Apply cross fade from one input audio stream to another input audio
stream. The cross fade is applied for specified duration near the end
of first stream.

The filter accepts the following options:

nb_samples, ns
Specify the number of samples for which the cross fade effect has
to last. At the end of the cross fade effect the first input audio
will be completely silent. Default is 44100.

duration, d
Specify the duration of the cross fade effect. See the Time
duration section in the ffmpeg-utils(1) manual for the accepted
syntax. By default the duration is determined by nb_samples. If
set this option is used instead of nb_samples.

overlap, o
Should first stream end overlap with second stream start. Default
is enabled.

curve1
Set curve for cross fade transition for first stream.

curve2
Set curve for cross fade transition for second stream.

For description of available curve types see afade filter
description.

Examples

· Cross fade from one input to another:

ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac

· Cross fade from one input to another but without overlapping:

ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac

adelay
Delay one or more audio channels.

Samples in delayed channel are filled with silence.

The filter accepts the following option:

delays
Set list of delays in milliseconds for each channel separated by
'|'. At least one delay greater than 0 should be provided. Unused
delays will be silently ignored. If number of given delays is
smaller than number of channels all remaining channels will not be
delayed.

Examples

· Delay first channel by 1.5 seconds, the third channel by 0.5
seconds and leave the second channel (and any other channels that
may be present) unchanged.

adelay=1500|0|500

aecho
Apply echoing to the input audio.

Echoes are reflected sound and can occur naturally amongst mountains
(and sometimes large buildings) when talking or shouting; digital echo
effects emulate this behaviour and are often used to help fill out the
sound of a single instrument or vocal. The time difference between the
original signal and the reflection is the "delay", and the loudness of
the reflected signal is the "decay". Multiple echoes can have
different delays and decays.

A description of the accepted parameters follows.

in_gain
Set input gain of reflected signal. Default is 0.6.

out_gain
Set output gain of reflected signal. Default is 0.3.

delays
Set list of time intervals in milliseconds between original signal
and reflections separated by '|'. Allowed range for each "delay" is
"(0 - 90000.0]". Default is 1000.

decays
Set list of loudnesses of reflected signals separated by '|'.
Allowed range for each "decay" is "(0 - 1.0]". Default is 0.5.

Examples

· Make it sound as if there are twice as many instruments as are
actually playing:

aecho=0.8:0.88:60:0.4

· If delay is very short, then it sound like a (metallic) robot
playing music:

aecho=0.8:0.88:6:0.4

· A longer delay will sound like an open air concert in the
mountains:

aecho=0.8:0.9:1000:0.3

· Same as above but with one more mountain:

aecho=0.8:0.9:1000|1800:0.3|0.25

aeval
Modify an audio signal according to the specified expressions.

This filter accepts one or more expressions (one for each channel),
which are evaluated and used to modify a corresponding audio signal.

It accepts the following parameters:

exprs
Set the '|'-separated expressions list for each separate channel.
If the number of input channels is greater than the number of
expressions, the last specified expression is used for the
remaining output channels.

channel_layout, c
Set output channel layout. If not specified, the channel layout is
specified by the number of expressions. If set to same, it will use
by default the same input channel layout.

Each expression in exprs can contain the following constants and
functions:

ch channel number of the current expression

n number of the evaluated sample, starting from 0

s sample rate

t time of the evaluated sample expressed in seconds

nb_in_channels
nb_out_channels
input and output number of channels

val(CH)
the value of input channel with number CH

Note: this filter is slow. For faster processing you should use a
dedicated filter.

Examples

· Half volume:

aeval=val(ch)/2:c=same

· Invert phase of the second channel:

aeval=val(0)|-val(1)

afade
Apply fade-in/out effect to input audio.

A description of the accepted parameters follows.

type, t
Specify the effect type, can be either "in" for fade-in, or "out"
for a fade-out effect. Default is "in".

start_sample, ss
Specify the number of the start sample for starting to apply the
fade effect. Default is 0.

nb_samples, ns
Specify the number of samples for which the fade effect has to
last. At the end of the fade-in effect the output audio will have
the same volume as the input audio, at the end of the fade-out
transition the output audio will be silence. Default is 44100.

start_time, st
Specify the start time of the fade effect. Default is 0. The value
must be specified as a time duration; see the Time duration section
in the ffmpeg-utils(1) manual for the accepted syntax. If set this
option is used instead of start_sample.

duration, d
Specify the duration of the fade effect. See the Time duration
section in the ffmpeg-utils(1) manual for the accepted syntax. At
the end of the fade-in effect the output audio will have the same
volume as the input audio, at the end of the fade-out transition
the output audio will be silence. By default the duration is
determined by nb_samples. If set this option is used instead of
nb_samples.

curve
Set curve for fade transition.

It accepts the following values:

tri select triangular, linear slope (default)

qsin
select quarter of sine wave

hsin
select half of sine wave

esin
select exponential sine wave

log select logarithmic

ipar
select inverted parabola

qua select quadratic

cub select cubic

squ select square root

cbr select cubic root

par select parabola

exp select exponential

iqsin
select inverted quarter of sine wave

ihsin
select inverted half of sine wave

dese
select double-exponential seat

desi
select double-exponential sigmoid

Examples

· Fade in first 15 seconds of audio:

afade=t=in:ss=0:d=15

· Fade out last 25 seconds of a 900 seconds audio:

afade=t=out:st=875:d=25

aformat
Set output format constraints for the input audio. The framework will
negotiate the most appropriate format to minimize conversions.

It accepts the following parameters:

sample_fmts
A '|'-separated list of requested sample formats.

sample_rates
A '|'-separated list of requested sample rates.

channel_layouts
A '|'-separated list of requested channel layouts.

See the Channel Layout section in the ffmpeg-utils(1) manual for
the required syntax.

If a parameter is omitted, all values are allowed.

Force the output to either unsigned 8-bit or signed 16-bit stereo

aformat=sample_fmts=u8|s16:channel_layouts=stereo

allpass
Apply a two-pole all-pass filter with central frequency (in Hz)
frequency, and filter-width width. An all-pass filter changes the
audio's frequency to phase relationship without changing its frequency
to amplitude relationship.

The filter accepts the following options:

frequency, f
Set frequency in Hz.

width_type
Set method to specify band-width of filter.

h Hz

q Q-Factor

o octave

s slope

width, w
Specify the band-width of a filter in width_type units.

amerge
Merge two or more audio streams into a single multi-channel stream.

The filter accepts the following options:

inputs
Set the number of inputs. Default is 2.

If the channel layouts of the inputs are disjoint, and therefore
compatible, the channel layout of the output will be set accordingly
and the channels will be reordered as necessary. If the channel layouts
of the inputs are not disjoint, the output will have all the channels
of the first input then all the channels of the second input, in that
order, and the channel layout of the output will be the default value
corresponding to the total number of channels.

For example, if the first input is in 2.1 (FL+FR+LF) and the second
input is FC+BL+BR, then the output will be in 5.1, with the channels in
the following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of
the first input, b1 is the first channel of the second input).

On the other hand, if both input are in stereo, the output channels
will be in the default order: a1, a2, b1, b2, and the channel layout
will be arbitrarily set to 4.0, which may or may not be the expected
value.

All inputs must have the same sample rate, and format.

If inputs do not have the same duration, the output will stop with the
shortest.

Examples

· Merge two mono files into a stereo stream:

amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge

· Multiple merges assuming 1 video stream and 6 audio streams in
input.mkv:

ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv

amix
Mixes multiple audio inputs into a single output.

Note that this filter only supports float samples (the amerge and pan
audio filters support many formats). If the amix input has integer
samples then aresample will be automatically inserted to perform the
conversion to float samples.

For example

ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT

will mix 3 input audio streams to a single output with the same
duration as the first input and a dropout transition time of 3 seconds.

It accepts the following parameters:

inputs
The number of inputs. If unspecified, it defaults to 2.

duration
How to determine the end-of-stream.

longest
The duration of the longest input. (default)

shortest
The duration of the shortest input.

first
The duration of the first input.

dropout_transition
The transition time, in seconds, for volume renormalization when an
input stream ends. The default value is 2 seconds.

anull
Pass the audio source unchanged to the output.

apad
Pad the end of an audio stream with silence.

This can be used together with ffmpeg -shortest to extend audio streams
to the same length as the video stream.

A description of the accepted options follows.

packet_size
Set silence packet size. Default value is 4096.

pad_len
Set the number of samples of silence to add to the end. After the
value is reached, the stream is terminated. This option is mutually
exclusive with whole_len.

whole_len
Set the minimum total number of samples in the output audio stream.
If the value is longer than the input audio length, silence is
added to the end, until the value is reached. This option is
mutually exclusive with pad_len.

If neither the pad_len nor the whole_len option is set, the filter will
add silence to the end of the input stream indefinitely.

Examples

· Add 1024 samples of silence to the end of the input:

apad=pad_len=1024

· Make sure the audio output will contain at least 10000 samples, pad
the input with silence if required:

apad=whole_len=10000

· Use ffmpeg to pad the audio input with silence, so that the video
stream will always result the shortest and will be converted until
the end in the output file when using the shortest option:

ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT

aphaser
Add a phasing effect to the input audio.

A phaser filter creates series of peaks and troughs in the frequency
spectrum. The position of the peaks and troughs are modulated so that
they vary over time, creating a sweeping effect.

A description of the accepted parameters follows.

in_gain
Set input gain. Default is 0.4.

out_gain
Set output gain. Default is 0.74

delay
Set delay in milliseconds. Default is 3.0.

decay
Set decay. Default is 0.4.

speed
Set modulation speed in Hz. Default is 0.5.

type
Set modulation type. Default is triangular.

It accepts the following values:

triangular, t
sinusoidal, s

aresample
Resample the input audio to the specified parameters, using the
libswresample library. If none are specified then the filter will
automatically convert between its input and output.

This filter is also able to stretch/squeeze the audio data to make it
match the timestamps or to inject silence / cut out audio to make it
match the timestamps, do a combination of both or do neither.

The filter accepts the syntax [sample_rate:]resampler_options, where
sample_rate expresses a sample rate and resampler_options is a list of
key=value pairs, separated by ":". See the ffmpeg-resampler manual for
the complete list of supported options.

Examples

· Resample the input audio to 44100Hz:

aresample=44100

· Stretch/squeeze samples to the given timestamps, with a maximum of
1000 samples per second compensation:

aresample=async=1000

asetnsamples
Set the number of samples per each output audio frame.

The last output packet may contain a different number of samples, as
the filter will flush all the remaining samples when the input audio
signal its end.

The filter accepts the following options:

nb_out_samples, n
Set the number of frames per each output audio frame. The number is
intended as the number of samples per each channel. Default value
is 1024.

pad, p
If set to 1, the filter will pad the last audio frame with zeroes,
so that the last frame will contain the same number of samples as
the previous ones. Default value is 1.

For example, to set the number of per-frame samples to 1234 and disable
padding for the last frame, use:

asetnsamples=n=1234:p=0

asetrate
Set the sample rate without altering the PCM data. This will result in
a change of speed and pitch.

The filter accepts the following options:

sample_rate, r
Set the output sample rate. Default is 44100 Hz.

ashowinfo
Show a line containing various information for each input audio frame.
The input audio is not modified.

The shown line contains a sequence of key/value pairs of the form
key:value.

The following values are shown in the output:

n The (sequential) number of the input frame, starting from 0.

pts The presentation timestamp of the input frame, in time base units;
the time base depends on the filter input pad, and is usually
1/sample_rate.

pts_time
The presentation timestamp of the input frame in seconds.

pos position of the frame in the input stream, -1 if this information
in unavailable and/or meaningless (for example in case of synthetic
audio)

fmt The sample format.

chlayout
The channel layout.

rate
The sample rate for the audio frame.

nb_samples
The number of samples (per channel) in the frame.

checksum
The Adler-32 checksum (printed in hexadecimal) of the audio data.
For planar audio, the data is treated as if all the planes were
concatenated.

plane_checksums
A list of Adler-32 checksums for each data plane.

astats
Display time domain statistical information about the audio channels.
Statistics are calculated and displayed for each audio channel and,
where applicable, an overall figure is also given.

It accepts the following option:

length
Short window length in seconds, used for peak and trough RMS
measurement. Default is 0.05 (50 milliseconds). Allowed range is
"[0.1 - 10]".

metadata
Set metadata injection. All the metadata keys are prefixed with
"lavfi.astats.X", where "X" is channel number starting from 1 or
string "Overall". Default is disabled.

Available keys for each channel are: DC_offset Min_level Max_level
Min_difference Max_difference Mean_difference Peak_level RMS_peak
RMS_trough Crest_factor Flat_factor Peak_count Bit_depth

and for Overall: DC_offset Min_level Max_level Min_difference
Max_difference Mean_difference Peak_level RMS_level RMS_peak
RMS_trough Flat_factor Peak_count Bit_depth Number_of_samples

For example full key look like this "lavfi.astats.1.DC_offset" or
this "lavfi.astats.Overall.Peak_count".

For description what each key means read below.

reset
Set number of frame after which stats are going to be recalculated.
Default is disabled.

A description of each shown parameter follows:

DC offset
Mean amplitude displacement from zero.

Min level
Minimal sample level.

Max level
Maximal sample level.

Min difference
Minimal difference between two consecutive samples.

Max difference
Maximal difference between two consecutive samples.

Mean difference
Mean difference between two consecutive samples. The average of
each difference between two consecutive samples.

Peak level dB
RMS level dB
Standard peak and RMS level measured in dBFS.

RMS peak dB
RMS trough dB
Peak and trough values for RMS level measured over a short window.

Crest factor
Standard ratio of peak to RMS level (note: not in dB).

Flat factor
Flatness (i.e. consecutive samples with the same value) of the
signal at its peak levels (i.e. either Min level or Max level).

Peak count
Number of occasions (not the number of samples) that the signal
attained either Min level or Max level.

Bit depth
Overall bit depth of audio. Number of bits used for each sample.

astreamsync
Forward two audio streams and control the order the buffers are
forwarded.

The filter accepts the following options:

expr, e
Set the expression deciding which stream should be forwarded next:
if the result is negative, the first stream is forwarded; if the
result is positive or zero, the second stream is forwarded. It can
use the following variables:

b1 b2
number of buffers forwarded so far on each stream

s1 s2
number of samples forwarded so far on each stream

t1 t2
current timestamp of each stream

The default value is "t1-t2", which means to always forward the
stream that has a smaller timestamp.

Examples

Stress-test "amerge" by randomly sending buffers on the wrong input,
while avoiding too much of a desynchronization:

amovie=file.ogg [a] ; amovie=file.mp3 [b] ;
[a] [b] astreamsync=(2*random(1))-1+tanh(5*(t1-t2)) [a2] [b2] ;
[a2] [b2] amerge

asyncts
Synchronize audio data with timestamps by squeezing/stretching it
and/or dropping samples/adding silence when needed.

This filter is not built by default, please use aresample to do
squeezing/stretching.

It accepts the following parameters:

compensate
Enable stretching/squeezing the data to make it match the
timestamps. Disabled by default. When disabled, time gaps are
covered with silence.

min_delta
The minimum difference between timestamps and audio data (in
seconds) to trigger adding/dropping samples. The default value is
0.1. If you get an imperfect sync with this filter, try setting
this parameter to 0.

max_comp
The maximum compensation in samples per second. Only relevant with
compensate=1. The default value is 500.

first_pts
Assume that the first PTS should be this value. The time base is 1
/ sample rate. This allows for padding/trimming at the start of the
stream. By default, no assumption is made about the first frame's
expected PTS, so no padding or trimming is done. For example, this
could be set to 0 to pad the beginning with silence if an audio
stream starts after the video stream or to trim any samples with a
negative PTS due to encoder delay.

atempo
Adjust audio tempo.

The filter accepts exactly one parameter, the audio tempo. If not
specified then the filter will assume nominal 1.0 tempo. Tempo must be
in the [0.5, 2.0] range.

Examples

· Slow down audio to 80% tempo:

atempo=0.8

· To speed up audio to 125% tempo:

atempo=1.25

atrim
Trim the input so that the output contains one continuous subpart of
the input.

It accepts the following parameters:

start
Timestamp (in seconds) of the start of the section to keep. I.e.
the audio sample with the timestamp start will be the first sample
in the output.

end Specify time of the first audio sample that will be dropped, i.e.
the audio sample immediately preceding the one with the timestamp
end will be the last sample in the output.

start_pts
Same as start, except this option sets the start timestamp in
samples instead of seconds.

end_pts
Same as end, except this option sets the end timestamp in samples
instead of seconds.

duration
The maximum duration of the output in seconds.

start_sample
The number of the first sample that should be output.

end_sample
The number of the first sample that should be dropped.

start, end, and duration are expressed as time duration specifications;
see the Time duration section in the ffmpeg-utils(1) manual.

Note that the first two sets of the start/end options and the duration
option look at the frame timestamp, while the _sample options simply
count the samples that pass through the filter. So start/end_pts and
start/end_sample will give different results when the timestamps are
wrong, inexact or do not start at zero. Also note that this filter does
not modify the timestamps. If you wish to have the output timestamps
start at zero, insert the asetpts filter after the atrim filter.

If multiple start or end options are set, this filter tries to be
greedy and keep all samples that match at least one of the specified
constraints. To keep only the part that matches all the constraints at
once, chain multiple atrim filters.

The defaults are such that all the input is kept. So it is possible to
set e.g. just the end values to keep everything before the specified
time.

Examples:

· Drop everything except the second minute of input:

ffmpeg -i INPUT -af atrim=60:120

· Keep only the first 1000 samples:

ffmpeg -i INPUT -af atrim=end_sample=1000

bandpass
Apply a two-pole Butterworth band-pass filter with central frequency
frequency, and (3dB-point) band-width width. The csg option selects a
constant skirt gain (peak gain = Q) instead of the default: constant
0dB peak gain. The filter roll off at 6dB per octave (20dB per
decade).

The filter accepts the following options:

frequency, f
Set the filter's central frequency. Default is 3000.

csg Constant skirt gain if set to 1. Defaults to 0.

width_type
Set method to specify band-width of filter.

h Hz

q Q-Factor

o octave

s slope

width, w
Specify the band-width of a filter in width_type units.

bandreject
Apply a two-pole Butterworth band-reject filter with central frequency
frequency, and (3dB-point) band-width width. The filter roll off at
6dB per octave (20dB per decade).

The filter accepts the following options:

frequency, f
Set the filter's central frequency. Default is 3000.

width_type
Set method to specify band-width of filter.

h Hz

q Q-Factor

o octave

s slope

width, w
Specify the band-width of a filter in width_type units.

bass
Boost or cut the bass (lower) frequencies of the audio using a two-pole
shelving filter with a response similar to that of a standard hi-fi's
tone-controls. This is also known as shelving equalisation (EQ).

The filter accepts the following options:

gain, g
Give the gain at 0 Hz. Its useful range is about -20 (for a large
cut) to +20 (for a large boost). Beware of clipping when using a
positive gain.

frequency, f
Set the filter's central frequency and so can be used to extend or
reduce the frequency range to be boosted or cut. The default value
is 100 Hz.

width_type
Set method to specify band-width of filter.

h Hz

q Q-Factor

o octave

s slope

width, w
Determine how steep is the filter's shelf transition.

biquad
Apply a biquad IIR filter with the given coefficients. Where b0, b1,
b2 and a0, a1, a2 are the numerator and denominator coefficients
respectively.

bs2b
Bauer stereo to binaural transformation, which improves headphone
listening of stereo audio records.

It accepts the following parameters:

profile
Pre-defined crossfeed level.

default
Default level (fcut=700, feed=50).

cmoy
Chu Moy circuit (fcut=700, feed=60).

jmeier
Jan Meier circuit (fcut=650, feed=95).

fcut
Cut frequency (in Hz).

feed
Feed level (in Hz).

channelmap
Remap input channels to new locations.

It accepts the following parameters:

channel_layout
The channel layout of the output stream.

map Map channels from input to output. The argument is a '|'-separated
list of mappings, each in the "in_channel-out_channel" or
in_channel form. in_channel can be either the name of the input
channel (e.g. FL for front left) or its index in the input channel
layout. out_channel is the name of the output channel or its index
in the output channel layout. If out_channel is not given then it
is implicitly an index, starting with zero and increasing by one
for each mapping.

If no mapping is present, the filter will implicitly map input channels
to output channels, preserving indices.

For example, assuming a 5.1+downmix input MOV file,

ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav

will create an output WAV file tagged as stereo from the downmix
channels of the input.

To fix a 5.1 WAV improperly encoded in AAC's native channel order

ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav

channelsplit
Split each channel from an input audio stream into a separate output
stream.

It accepts the following parameters:

channel_layout
The channel layout of the input stream. The default is "stereo".

For example, assuming a stereo input MP3 file,

ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv

will create an output Matroska file with two audio streams, one
containing only the left channel and the other the right channel.

Split a 5.1 WAV file into per-channel files:

ffmpeg -i in.wav -filter_complex
'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
-map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
side_right.wav

chorus
Add a chorus effect to the audio.

Can make a single vocal sound like a chorus, but can also be applied to
instrumentation.

Chorus resembles an echo effect with a short delay, but whereas with
echo the delay is constant, with chorus, it is varied using using
sinusoidal or triangular modulation. The modulation depth defines the
range the modulated delay is played before or after the delay. Hence
the delayed sound will sound slower or faster, that is the delayed
sound tuned around the original one, like in a chorus where some vocals
are slightly off key.

It accepts the following parameters:

in_gain
Set input gain. Default is 0.4.

out_gain
Set output gain. Default is 0.4.

delays
Set delays. A typical delay is around 40ms to 60ms.

decays
Set decays.

speeds
Set speeds.

depths
Set depths.

Examples

· A single delay:

chorus=0.7:0.9:55:0.4:0.25:2

· Two delays:

chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3

· Fuller sounding chorus with three delays:

chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3

compand
Compress or expand the audio's dynamic range.

It accepts the following parameters:

attacks
decays
A list of times in seconds for each channel over which the
instantaneous level of the input signal is averaged to determine
its volume. attacks refers to increase of volume and decays refers
to decrease of volume. For most situations, the attack time
(response to the audio getting louder) should be shorter than the
decay time, because the human ear is more sensitive to sudden loud
audio than sudden soft audio. A typical value for attack is 0.3
seconds and a typical value for decay is 0.8 seconds. If specified
number of attacks & decays is lower than number of channels, the
last set attack/decay will be used for all remaining channels.

points
A list of points for the transfer function, specified in dB
relative to the maximum possible signal amplitude. Each key points
list must be defined using the following syntax:
"x0/y0|x1/y1|x2/y2|...." or "x0/y0 x1/y1 x2/y2 ...."

The input values must be in strictly increasing order but the
transfer function does not have to be monotonically rising. The
point "0/0" is assumed but may be overridden (by "0/out-dBn").
Typical values for the transfer function are "-70/-70|-60/-20".

soft-knee
Set the curve radius in dB for all joints. It defaults to 0.01.

gain
Set the additional gain in dB to be applied at all points on the
transfer function. This allows for easy adjustment of the overall
gain. It defaults to 0.

volume
Set an initial volume, in dB, to be assumed for each channel when
filtering starts. This permits the user to supply a nominal level
initially, so that, for example, a very large gain is not applied
to initial signal levels before the companding has begun to
operate. A typical value for audio which is initially quiet is -90
dB. It defaults to 0.

delay
Set a delay, in seconds. The input audio is analyzed immediately,
but audio is delayed before being fed to the volume adjuster.
Specifying a delay approximately equal to the attack/decay times
allows the filter to effectively operate in predictive rather than
reactive mode. It defaults to 0.

Examples

· Make music with both quiet and loud passages suitable for listening
to in a noisy environment:

compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2

Another example for audio with whisper and explosion parts:

compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0

· A noise gate for when the noise is at a lower level than the
signal:

compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1

· Here is another noise gate, this time for when the noise is at a
higher level than the signal (making it, in some ways, similar to
squelch):

compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1

dcshift
Apply a DC shift to the audio.

This can be useful to remove a DC offset (caused perhaps by a hardware
problem in the recording chain) from the audio. The effect of a DC
offset is reduced headroom and hence volume. The astats filter can be
used to determine if a signal has a DC offset.

shift
Set the DC shift, allowed range is [-1, 1]. It indicates the amount
to shift the audio.

limitergain
Optional. It should have a value much less than 1 (e.g. 0.05 or
0.02) and is used to prevent clipping.

dynaudnorm
Dynamic Audio Normalizer.

This filter applies a certain amount of gain to the input audio in
order to bring its peak magnitude to a target level (e.g. 0 dBFS).
However, in contrast to more "simple" normalization algorithms, the
Dynamic Audio Normalizer *dynamically* re-adjusts the gain factor to
the input audio. This allows for applying extra gain to the "quiet"
sections of the audio while avoiding distortions or clipping the "loud"
sections. In other words: The Dynamic Audio Normalizer will "even out"
the volume of quiet and loud sections, in the sense that the volume of
each section is brought to the same target level. Note, however, that
the Dynamic Audio Normalizer achieves this goal *without* applying
"dynamic range compressing". It will retain 100% of the dynamic range
*within* each section of the audio file.

f Set the frame length in milliseconds. In range from 10 to 8000
milliseconds. Default is 500 milliseconds. The Dynamic Audio
Normalizer processes the input audio in small chunks, referred to
as frames. This is required, because a peak magnitude has no
meaning for just a single sample value. Instead, we need to
determine the peak magnitude for a contiguous sequence of sample
values. While a "standard" normalizer would simply use the peak
magnitude of the complete file, the Dynamic Audio Normalizer
determines the peak magnitude individually for each frame. The
length of a frame is specified in milliseconds. By default, the
Dynamic Audio Normalizer uses a frame length of 500 milliseconds,
which has been found to give good results with most files. Note
that the exact frame length, in number of samples, will be
determined automatically, based on the sampling rate of the
individual input audio file.

g Set the Gaussian filter window size. In range from 3 to 301, must
be odd number. Default is 31. Probably the most important
parameter of the Dynamic Audio Normalizer is the "window size" of
the Gaussian smoothing filter. The filter's window size is
specified in frames, centered around the current frame. For the
sake of simplicity, this must be an odd number. Consequently, the
default value of 31 takes into account the current frame, as well
as the 15 preceding frames and the 15 subsequent frames. Using a
larger window results in a stronger smoothing effect and thus in
less gain variation, i.e. slower gain adaptation. Conversely, using
a smaller window results in a weaker smoothing effect and thus in
more gain variation, i.e. faster gain adaptation. In other words,
the more you increase this value, the more the Dynamic Audio
Normalizer will behave like a "traditional" normalization filter.
On the contrary, the more you decrease this value, the more the
Dynamic Audio Normalizer will behave like a dynamic range
compressor.

p Set the target peak value. This specifies the highest permissible
magnitude level for the normalized audio input. This filter will
try to approach the target peak magnitude as closely as possible,
but at the same time it also makes sure that the normalized signal
will never exceed the peak magnitude. A frame's maximum local gain
factor is imposed directly by the target peak magnitude. The
default value is 0.95 and thus leaves a headroom of 5%*. It is not
recommended to go above this value.

m Set the maximum gain factor. In range from 1.0 to 100.0. Default is
10.0. The Dynamic Audio Normalizer determines the maximum possible
(local) gain factor for each input frame, i.e. the maximum gain
factor that does not result in clipping or distortion. The maximum
gain factor is determined by the frame's highest magnitude sample.
However, the Dynamic Audio Normalizer additionally bounds the
frame's maximum gain factor by a predetermined (global) maximum
gain factor. This is done in order to avoid excessive gain factors
in "silent" or almost silent frames. By default, the maximum gain
factor is 10.0, For most inputs the default value should be
sufficient and it usually is not recommended to increase this
value. Though, for input with an extremely low overall volume
level, it may be necessary to allow even higher gain factors. Note,
however, that the Dynamic Audio Normalizer does not simply apply a
"hard" threshold (i.e. cut off values above the threshold).
Instead, a "sigmoid" threshold function will be applied. This way,
the gain factors will smoothly approach the threshold value, but
never exceed that value.

r Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 -
disabled. By default, the Dynamic Audio Normalizer performs "peak"
normalization. This means that the maximum local gain factor for
each frame is defined (only) by the frame's highest magnitude
sample. This way, the samples can be amplified as much as possible
without exceeding the maximum signal level, i.e. without clipping.
Optionally, however, the Dynamic Audio Normalizer can also take
into account the frame's root mean square, abbreviated RMS. In
electrical engineering, the RMS is commonly used to determine the
power of a time-varying signal. It is therefore considered that the
RMS is a better approximation of the "perceived loudness" than just
looking at the signal's peak magnitude. Consequently, by adjusting
all frames to a constant RMS value, a uniform "perceived loudness"
can be established. If a target RMS value has been specified, a
frame's local gain factor is defined as the factor that would
result in exactly that RMS value. Note, however, that the maximum
local gain factor is still restricted by the frame's highest
magnitude sample, in order to prevent clipping.

n Enable channels coupling. By default is enabled. By default, the
Dynamic Audio Normalizer will amplify all channels by the same
amount. This means the same gain factor will be applied to all
channels, i.e. the maximum possible gain factor is determined by
the "loudest" channel. However, in some recordings, it may happen
that the volume of the different channels is uneven, e.g. one
channel may be "quieter" than the other one(s). In this case, this
option can be used to disable the channel coupling. This way, the
gain factor will be determined independently for each channel,
depending only on the individual channel's highest magnitude
sample. This allows for harmonizing the volume of the different
channels.

c Enable DC bias correction. By default is disabled. An audio signal
(in the time domain) is a sequence of sample values. In the
Dynamic Audio Normalizer these sample values are represented in the
-1.0 to 1.0 range, regardless of the original input format.
Normally, the audio signal, or "waveform", should be centered
around the zero point. That means if we calculate the mean value
of all samples in a file, or in a single frame, then the result
should be 0.0 or at least very close to that value. If, however,
there is a significant deviation of the mean value from 0.0, in
either positive or negative direction, this is referred to as a DC
bias or DC offset. Since a DC bias is clearly undesirable, the
Dynamic Audio Normalizer provides optional DC bias correction.
With DC bias correction enabled, the Dynamic Audio Normalizer will
determine the mean value, or "DC correction" offset, of each input
frame and subtract that value from all of the frame's sample values
which ensures those samples are centered around 0.0 again. Also, in
order to avoid "gaps" at the frame boundaries, the DC correction
offset values will be interpolated smoothly between neighbouring
frames.

b Enable alternative boundary mode. By default is disabled. The
Dynamic Audio Normalizer takes into account a certain neighbourhood
around each frame. This includes the preceding frames as well as
the subsequent frames. However, for the "boundary" frames, located
at the very beginning and at the very end of the audio file, not
all neighbouring frames are available. In particular, for the first
few frames in the audio file, the preceding frames are not known.
And, similarly, for the last few frames in the audio file, the
subsequent frames are not known. Thus, the question arises which
gain factors should be assumed for the missing frames in the
"boundary" region. The Dynamic Audio Normalizer implements two
modes to deal with this situation. The default boundary mode
assumes a gain factor of exactly 1.0 for the missing frames,
resulting in a smooth "fade in" and "fade out" at the beginning and
at the end of the input, respectively.

s Set the compress factor. In range from 0.0 to 30.0. Default is 0.0.
By default, the Dynamic Audio Normalizer does not apply
"traditional" compression. This means that signal peaks will not be
pruned and thus the full dynamic range will be retained within each
local neighbourhood. However, in some cases it may be desirable to
combine the Dynamic Audio Normalizer's normalization algorithm with
a more "traditional" compression. For this purpose, the Dynamic
Audio Normalizer provides an optional compression (thresholding)
function. If (and only if) the compression feature is enabled, all
input frames will be processed by a soft knee thresholding function
prior to the actual normalization process. Put simply, the
thresholding function is going to prune all samples whose magnitude
exceeds a certain threshold value. However, the Dynamic Audio
Normalizer does not simply apply a fixed threshold value. Instead,
the threshold value will be adjusted for each individual frame. In
general, smaller parameters result in stronger compression, and
vice versa. Values below 3.0 are not recommended, because audible
distortion may appear.

earwax
Make audio easier to listen to on headphones.

This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio
so that when listened to on headphones the stereo image is moved from
inside your head (standard for headphones) to outside and in front of
the listener (standard for speakers).

Ported from SoX.

equalizer
Apply a two-pole peaking equalisation (EQ) filter. With this filter,
the signal-level at and around a selected frequency can be increased or
decreased, whilst (unlike bandpass and bandreject filters) that at all
other frequencies is unchanged.

In order to produce complex equalisation curves, this filter can be
given several times, each with a different central frequency.

The filter accepts the following options:

frequency, f
Set the filter's central frequency in Hz.

width_type
Set method to specify band-width of filter.

h Hz

q Q-Factor

o octave

s slope

width, w
Specify the band-width of a filter in width_type units.

gain, g
Set the required gain or attenuation in dB. Beware of clipping
when using a positive gain.

Examples

· Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:

equalizer=f=1000:width_type=h:width=200:g=-10

· Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz
with Q 2:

equalizer=f=1000:width_type=q:width=1:g=2,equalizer=f=100:width_type=q:width=2:g=-5

flanger
Apply a flanging effect to the audio.

The filter accepts the following options:

delay
Set base delay in milliseconds. Range from 0 to 30. Default value
is 0.

depth
Set added swep delay in milliseconds. Range from 0 to 10. Default
value is 2.

regen
Set percentage regeneration (delayed signal feedback). Range from
-95 to 95. Default value is 0.

width
Set percentage of delayed signal mixed with original. Range from 0
to 100. Default value is 71.

speed
Set sweeps per second (Hz). Range from 0.1 to 10. Default value is
0.5.

shape
Set swept wave shape, can be triangular or sinusoidal. Default
value is sinusoidal.

phase
Set swept wave percentage-shift for multi channel. Range from 0 to
100. Default value is 25.

interp
Set delay-line interpolation, linear or quadratic. Default is
linear.

highpass
Apply a high-pass filter with 3dB point frequency. The filter can be
either single-pole, or double-pole (the default). The filter roll off
at 6dB per pole per octave (20dB per pole per decade).

The filter accepts the following options:

frequency, f
Set frequency in Hz. Default is 3000.

poles, p
Set number of poles. Default is 2.

width_type
Set method to specify band-width of filter.

h Hz

q Q-Factor

o octave

s slope

width, w
Specify the band-width of a filter in width_type units. Applies
only to double-pole filter. The default is 0.707q and gives a
Butterworth response.

join
Join multiple input streams into one multi-channel stream.

It accepts the following parameters:

inputs
The number of input streams. It defaults to 2.

channel_layout
The desired output channel layout. It defaults to stereo.

map Map channels from inputs to output. The argument is a '|'-separated
list of mappings, each in the "input_idx.in_channel-out_channel"
form. input_idx is the 0-based index of the input stream.
in_channel can be either the name of the input channel (e.g. FL for
front left) or its index in the specified input stream. out_channel
is the name of the output channel.

The filter will attempt to guess the mappings when they are not
specified explicitly. It does so by first trying to find an unused
matching input channel and if that fails it picks the first unused
input channel.

Join 3 inputs (with properly set channel layouts):

ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT

Build a 5.1 output from 6 single-channel streams:

ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE'
out

ladspa
Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.

To enable compilation of this filter you need to configure FFmpeg with
"--enable-ladspa".

file, f
Specifies the name of LADSPA plugin library to load. If the
environment variable LADSPA_PATH is defined, the LADSPA plugin is
searched in each one of the directories specified by the colon
separated list in LADSPA_PATH, otherwise in the standard LADSPA
paths, which are in this order: HOME/.ladspa/lib/,
/usr/local/lib/ladspa/, /usr/lib/ladspa/.

plugin, p
Specifies the plugin within the library. Some libraries contain
only one plugin, but others contain many of them. If this is not
set filter will list all available plugins within the specified
library.

controls, c
Set the '|' separated list of controls which are zero or more
floating point values that determine the behavior of the loaded
plugin (for example delay, threshold or gain). Controls need to be
defined using the following syntax:
c0=value0|c1=value1|c2=value2|..., where valuei is the value set on
the i-th control. If controls is set to "help", all available
controls and their valid ranges are printed.

sample_rate, s
Specify the sample rate, default to 44100. Only used if plugin have
zero inputs.

nb_samples, n
Set the number of samples per channel per each output frame,
default is 1024. Only used if plugin have zero inputs.

duration, d
Set the minimum duration of the sourced audio. See the Time
duration section in the ffmpeg-utils(1) manual for the accepted
syntax. Note that the resulting duration may be greater than the
specified duration, as the generated audio is always cut at the end
of a complete frame. If not specified, or the expressed duration
is negative, the audio is supposed to be generated forever. Only
used if plugin have zero inputs.

Examples

· List all available plugins within amp (LADSPA example plugin)
library:

ladspa=file=amp

· List all available controls and their valid ranges for "vcf_notch"
plugin from "VCF" library:

ladspa=f=vcf:p=vcf_notch:c=help

· Simulate low quality audio equipment using "Computer Music Toolkit"
(CMT) plugin library:

ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12

· Add reverberation to the audio using TAP-plugins (Tom's Audio
Processing plugins):

ladspa=file=tap_reverb:tap_reverb

· Generate white noise, with 0.2 amplitude:

ladspa=file=cmt:noise_source_white:c=c0=.2

· Generate 20 bpm clicks using plugin "C* Click - Metronome" from the
"C* Audio Plugin Suite" (CAPS) library:

ladspa=file=caps:Click:c=c1=20'

· Apply "C* Eq10X2 - Stereo 10-band equaliser" effect:

ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2

Commands

This filter supports the following commands:

cN Modify the N-th control value.

If the specified value is not valid, it is ignored and prior one is
kept.

lowpass
Apply a low-pass filter with 3dB point frequency. The filter can be
either single-pole or double-pole (the default). The filter roll off
at 6dB per pole per octave (20dB per pole per decade).

The filter accepts the following options:

frequency, f
Set frequency in Hz. Default is 500.

poles, p
Set number of poles. Default is 2.

width_type
Set method to specify band-width of filter.

h Hz

q Q-Factor

o octave

s slope

width, w
Specify the band-width of a filter in width_type units. Applies
only to double-pole filter. The default is 0.707q and gives a
Butterworth response.

pan
Mix channels with specific gain levels. The filter accepts the output
channel layout followed by a set of channels definitions.

This filter is also designed to efficiently remap the channels of an
audio stream.

The filter accepts parameters of the form: "l|outdef|outdef|..."

l output channel layout or number of channels

outdef
output channel specification, of the form:
"out_name=[gain*]in_name[+[gain*]in_name...]"

out_name
output channel to define, either a channel name (FL, FR, etc.) or a
channel number (c0, c1, etc.)

gain
multiplicative coefficient for the channel, 1 leaving the volume
unchanged

in_name
input channel to use, see out_name for details; it is not possible
to mix named and numbered input channels

If the `=' in a channel specification is replaced by `<', then the
gains for that specification will be renormalized so that the total is
1, thus avoiding clipping noise.

Mixing examples

For example, if you want to down-mix from stereo to mono, but with a
bigger factor for the left channel:

pan=1c|c0=0.9*c0+0.1*c1

A customized down-mix to stereo that works automatically for 3-, 4-, 5-
and 7-channels surround:

pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR

Note that ffmpeg integrates a default down-mix (and up-mix) system that
should be preferred (see "-ac" option) unless you have very specific
needs.

Remapping examples

The channel remapping will be effective if, and only if:

*<gain coefficients are zeroes or ones,>
*<only one input per channel output,>

If all these conditions are satisfied, the filter will notify the user
("Pure channel mapping detected"), and use an optimized and lossless
method to do the remapping.

For example, if you have a 5.1 source and want a stereo audio stream by
dropping the extra channels:

pan="stereo| c0=FL | c1=FR"

Given the same source, you can also switch front left and front right
channels and keep the input channel layout:

pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"

If the input is a stereo audio stream, you can mute the front left
channel (and still keep the stereo channel layout) with:

pan="stereo|c1=c1"

Still with a stereo audio stream input, you can copy the right channel
in both front left and right:

pan="stereo| c0=FR | c1=FR"

replaygain
ReplayGain scanner filter. This filter takes an audio stream as an
input and outputs it unchanged. At end of filtering it displays
"track_gain" and "track_peak".

resample
Convert the audio sample format, sample rate and channel layout. It is
not meant to be used directly.

sidechaincompress
This filter acts like normal compressor but has the ability to compress
detected signal using second input signal. It needs two input streams
and returns one output stream. First input stream will be processed
depending on second stream signal. The filtered signal then can be
filtered with other filters in later stages of processing. See pan and
amerge filter.

The filter accepts the following options:

threshold
If a signal of second stream raises above this level it will affect
the gain reduction of first stream. By default is 0.125. Range is
between 0.00097563 and 1.

ratio
Set a ratio about which the signal is reduced. 1:2 means that if
the level raised 4dB above the threshold, it will be only 2dB above
after the reduction. Default is 2. Range is between 1 and 20.

attack
Amount of milliseconds the signal has to rise above the threshold
before gain reduction starts. Default is 20. Range is between 0.01
and 2000.

release
Amount of milliseconds the signal has to fall below the threshold
before reduction is decreased again. Default is 250. Range is
between 0.01 and 9000.

makeup
Set the amount by how much signal will be amplified after
processing. Default is 2. Range is from 1 and 64.

knee
Curve the sharp knee around the threshold to enter gain reduction
more softly. Default is 2.82843. Range is between 1 and 8.

link
Choose if the "average" level between all channels of side-chain
stream or the louder("maximum") channel of side-chain stream
affects the reduction. Default is "average".

detection
Should the exact signal be taken in case of "peak" or an RMS one in
case of "rms". Default is "rms" which is mainly smoother.

Examples

· Full ffmpeg example taking 2 audio inputs, 1st input to be
compressed depending on the signal of 2nd input and later
compressed signal to be merged with 2nd input:

ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"

silencedetect
Detect silence in an audio stream.

This filter logs a message when it detects that the input audio volume
is less or equal to a noise tolerance value for a duration greater or
equal to the minimum detected noise duration.

The printed times and duration are expressed in seconds.

The filter accepts the following options:

duration, d
Set silence duration until notification (default is 2 seconds).

noise, n
Set noise tolerance. Can be specified in dB (in case "dB" is
appended to the specified value) or amplitude ratio. Default is
-60dB, or 0.001.

Examples

· Detect 5 seconds of silence with -50dB noise tolerance:

silencedetect=n=-50dB:d=5

· Complete example with ffmpeg to detect silence with 0.0001 noise
tolerance in silence.mp3:

ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -

silenceremove
Remove silence from the beginning, middle or end of the audio.

The filter accepts the following options:

start_periods
This value is used to indicate if audio should be trimmed at
beginning of the audio. A value of zero indicates no silence should
be trimmed from the beginning. When specifying a non-zero value, it
trims audio up until it finds non-silence. Normally, when trimming
silence from beginning of audio the start_periods will be 1 but it
can be increased to higher values to trim all audio up to specific
count of non-silence periods. Default value is 0.

start_duration
Specify the amount of time that non-silence must be detected before
it stops trimming audio. By increasing the duration, bursts of
noises can be treated as silence and trimmed off. Default value is
0.

start_threshold
This indicates what sample value should be treated as silence. For
digital audio, a value of 0 may be fine but for audio recorded from
analog, you may wish to increase the value to account for
background noise. Can be specified in dB (in case "dB" is appended
to the specified value) or amplitude ratio. Default value is 0.

stop_periods
Set the count for trimming silence from the end of audio. To
remove silence from the middle of a file, specify a stop_periods
that is negative. This value is then treated as a positive value
and is used to indicate the effect should restart processing as
specified by start_periods, making it suitable for removing periods
of silence in the middle of the audio. Default value is 0.

stop_duration
Specify a duration of silence that must exist before audio is not
copied any more. By specifying a higher duration, silence that is
wanted can be left in the audio. Default value is 0.

stop_threshold
This is the same as start_threshold but for trimming silence from
the end of audio. Can be specified in dB (in case "dB" is appended
to the specified value) or amplitude ratio. Default value is 0.

leave_silence
This indicate that stop_duration length of audio should be left
intact at the beginning of each period of silence. For example, if
you want to remove long pauses between words but do not want to
remove the pauses completely. Default value is 0.

Examples

· The following example shows how this filter can be used to start a
recording that does not contain the delay at the start which
usually occurs between pressing the record button and the start of
the performance:

silenceremove=1:5:0.02

treble
Boost or cut treble (upper) frequencies of the audio using a two-pole
shelving filter with a response similar to that of a standard hi-fi's
tone-controls. This is also known as shelving equalisation (EQ).

The filter accepts the following options:

gain, g
Give the gain at whichever is the lower of ~22 kHz and the Nyquist
frequency. Its useful range is about -20 (for a large cut) to +20
(for a large boost). Beware of clipping when using a positive gain.

frequency, f
Set the filter's central frequency and so can be used to extend or
reduce the frequency range to be boosted or cut. The default value
is 3000 Hz.

width_type
Set method to specify band-width of filter.

h Hz

q Q-Factor

o octave

s slope

width, w
Determine how steep is the filter's shelf transition.

volume
Adjust the input audio volume.

It accepts the following parameters:

volume
Set audio volume expression.

Output values are clipped to the maximum value.

The output audio volume is given by the relation:

<output_volume> = <volume> * <input_volume>

The default value for volume is "1.0".

precision
This parameter represents the mathematical precision.

It determines which input sample formats will be allowed, which
affects the precision of the volume scaling.

fixed
8-bit fixed-point; this limits input sample format to U8, S16,
and S32.

float
32-bit floating-point; this limits input sample format to FLT.
(default)

double
64-bit floating-point; this limits input sample format to DBL.

replaygain
Choose the behaviour on encountering ReplayGain side data in input
frames.

drop
Remove ReplayGain side data, ignoring its contents (the
default).

ignore
Ignore ReplayGain side data, but leave it in the frame.

track
Prefer the track gain, if present.

album
Prefer the album gain, if present.

replaygain_preamp
Pre-amplification gain in dB to apply to the selected replaygain
gain.

Default value for replaygain_preamp is 0.0.

eval
Set when the volume expression is evaluated.

It accepts the following values:

once
only evaluate expression once during the filter initialization,
or when the volume command is sent

frame
evaluate expression for each incoming frame

Default value is once.

The volume expression can contain the following parameters.

n frame number (starting at zero)

nb_channels
number of channels

nb_consumed_samples
number of samples consumed by the filter

nb_samples
number of samples in the current frame

pos original frame position in the file

pts frame PTS

sample_rate
sample rate

startpts
PTS at start of stream

startt
time at start of stream

t frame time

tb timestamp timebase

volume
last set volume value

Note that when eval is set to once only the sample_rate and tb
variables are available, all other variables will evaluate to NAN.

Commands

This filter supports the following commands:

volume
Modify the volume expression. The command accepts the same syntax
of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

replaygain_noclip
Prevent clipping by limiting the gain applied.

Default value for replaygain_noclip is 1.

Examples

· Halve the input audio volume:

volume=volume=0.5
volume=volume=1/2
volume=volume=-6.0206dB

In all the above example the named key for volume can be omitted,
for example like in:

volume=0.5

· Increase input audio power by 6 decibels using fixed-point
precision:

volume=volume=6dB:precision=fixed

· Fade volume after time 10 with an annihilation period of 5 seconds:

volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame

volumedetect
Detect the volume of the input video.

The filter has no parameters. The input is not modified. Statistics
about the volume will be printed in the log when the input stream end
is reached.

In particular it will show the mean volume (root mean square), maximum
volume (on a per-sample basis), and the beginning of a histogram of the
registered volume values (from the maximum value to a cumulated 1/1000
of the samples).

All volumes are in decibels relative to the maximum PCM value.

Examples

Here is an excerpt of the output:

[Parsed_volumedetect_0 0xa23120] mean_volume: -27 dB
[Parsed_volumedetect_0 0xa23120] max_volume: -4 dB
[Parsed_volumedetect_0 0xa23120] histogram_4db: 6
[Parsed_volumedetect_0 0xa23120] histogram_5db: 62
[Parsed_volumedetect_0 0xa23120] histogram_6db: 286
[Parsed_volumedetect_0 0xa23120] histogram_7db: 1042
[Parsed_volumedetect_0 0xa23120] histogram_8db: 2551
[Parsed_volumedetect_0 0xa23120] histogram_9db: 4609
[Parsed_volumedetect_0 0xa23120] histogram_10db: 8409

It means that:

· The mean square energy is approximately -27 dB, or 10^-2.7.

· The largest sample is at -4 dB, or more precisely between -4 dB and
-5 dB.

· There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

In other words, raising the volume by +4 dB does not cause any
clipping, raising it by +5 dB causes clipping for 6 samples, etc.

AUDIO SOURCES
Below is a description of the currently available audio sources.

abuffer
Buffer audio 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/asrc_abuffer.h.

It accepts the following parameters:

time_base
The timebase which will be used for timestamps of submitted frames.
It must be either a floating-point number or in
numerator/denominator form.

sample_rate
The sample rate of the incoming audio buffers.

sample_fmt
The sample format of the incoming audio buffers. Either a sample
format name or its corresponding integer representation from the
enum AVSampleFormat in libavutil/samplefmt.h

channel_layout
The channel layout of the incoming audio buffers. Either a channel
layout name from channel_layout_map in libavutil/channel_layout.c
or its corresponding integer representation from the AV_CH_LAYOUT_*
macros in libavutil/channel_layout.h

channels
The number of channels of the incoming audio buffers. If both
channels and channel_layout are specified, then they must be
consistent.

Examples

abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo

will instruct the source to accept planar 16bit signed stereo at
44100Hz. Since the sample format with name "s16p" corresponds to the
number 6 and the "stereo" channel layout corresponds to the value 0x3,
this is equivalent to:

abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3

aevalsrc
Generate an audio signal specified by an expression.

This source accepts in input one or more expressions (one for each
channel), which are evaluated and used to generate a corresponding
audio signal.

This source accepts the following options:

exprs
Set the '|'-separated expressions list for each separate channel.
In case the channel_layout option is not specified, the selected
channel layout depends on the number of provided expressions.
Otherwise the last specified expression is applied to the remaining
output channels.

channel_layout, c
Set the channel layout. The number of channels in the specified
layout must be equal to the number of specified expressions.

If not specified, or the expressed duration is negative, the audio
is supposed to be generated forever.

nb_samples, n
Set the number of samples per channel per each output frame,
default to 1024.

sample_rate, s
Specify the sample rate, default to 44100.

Each expression in exprs can contain the following constants:

n number of the evaluated sample, starting from 0

t time of the evaluated sample expressed in seconds, starting from 0

s sample rate

Examples

· Generate silence:

aevalsrc=0

· Generate a sin signal with frequency of 440 Hz, set sample rate to
8000 Hz:

aevalsrc="sin(440*2*PI*t):s=8000"

· Generate a two channels signal, specify the channel layout (Front
Center + Back Center) explicitly:

aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"

· Generate white noise:

aevalsrc="-2+random(0)"

· Generate an amplitude modulated signal:

aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"

· Generate 2.5 Hz binaural beats on a 360 Hz carrier:

aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"

anullsrc
The null audio source, return unprocessed audio frames. It is mainly
useful as a template and to be employed in analysis / debugging tools,
or as the source for filters which ignore the input data (for example
the sox synth filter).

This source accepts the following options:

channel_layout, cl
Specifies the channel layout, and can be either an integer or a
string representing a channel layout. The default value of
channel_layout is "stereo".

Check the channel_layout_map definition in
libavutil/channel_layout.c for the mapping between strings and
channel layout values.

sample_rate, r
Specifies the sample rate, and defaults to 44100.

nb_samples, n
Set the number of samples per requested frames.

Examples

· Set the sample rate to 48000 Hz and the channel layout to
AV_CH_LAYOUT_MONO.

anullsrc=r=48000:cl=4

· Do the same operation with a more obvious syntax:

anullsrc=r=48000:cl=mono

All the parameters need to be explicitly defined.

flite
Synthesize a voice utterance using the libflite library.

To enable compilation of this filter you need to configure FFmpeg with
"--enable-libflite".

Note that the flite library is not thread-safe.

The filter accepts the following options:

list_voices
If set to 1, list the names of the available voices and exit
immediately. Default value is 0.

nb_samples, n
Set the maximum number of samples per frame. Default value is 512.

textfile
Set the filename containing the text to speak.

text
Set the text to speak.

voice, v
Set the voice to use for the speech synthesis. Default value is
"kal". See also the list_voices option.

Examples

· Read from file speech.txt, and synthesize the text using the
standard flite voice:

flite=textfile=speech.txt

· Read the specified text selecting the "slt" voice:

flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

· Input text to ffmpeg:

ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

· Make ffplay speak the specified text, using "flite" and the "lavfi"
device:

ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'

For more information about libflite, check:
<http://www.speech.cs.cmu.edu/flite/>

sine
Generate an audio signal made of a sine wave with amplitude 1/8.

The audio signal is bit-exact.

The filter accepts the following options:

frequency, f
Set the carrier frequency. Default is 440 Hz.

beep_factor, b
Enable a periodic beep every second with frequency beep_factor
times the carrier frequency. Default is 0, meaning the beep is
disabled.

sample_rate, r
Specify the sample rate, default is 44100.

duration, d
Specify the duration of the generated audio stream.

samples_per_frame
Set the number of samples per output frame, default is 1024.

Examples

· Generate a simple 440 Hz sine wave:

sine

· Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5
seconds:

sine=220:4:d=5
sine=f=220:b=4:d=5
sine=frequency=220:beep_factor=4:duration=5

AUDIO SINKS
Below is a description of the currently available audio sinks.

abuffersink
Buffer audio frames, and make them available to the end of filter
chain.

This sink is mainly intended for programmatic use, in particular
through the interface defined in libavfilter/buffersink.h or the
options system.

It accepts a pointer to an AVABufferSinkContext structure, which
defines the incoming buffers' formats, to be passed as the opaque
parameter to "avfilter_init_filter" for initialization.

anullsink
Null audio sink; do absolutely nothing with the input audio. It is
mainly useful as a template and for use in analysis / debugging tools.

VIDEO FILTERS
When you configure your FFmpeg 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.

alphaextract
Extract the alpha component from the input as a grayscale video. This
is especially useful with the alphamerge filter.

alphamerge
Add or replace the alpha component of the primary input with the
grayscale value of a second input. This is intended for use with
alphaextract to allow the transmission or storage of frame sequences
that have alpha in a format that doesn't support an alpha channel.

For example, to reconstruct full frames from a normal YUV-encoded video
and a separate video created with alphaextract, you might use:

movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out]

Since this filter is designed for reconstruction, it operates on frame
sequences without considering timestamps, and terminates when either
input reaches end of stream. This will cause problems if your encoding
pipeline drops frames. If you're trying to apply an image as an overlay
to a video stream, consider the overlay filter instead.

ass
Same as the subtitles filter, except that it doesn't require libavcodec
and libavformat to work. On the other hand, it is limited to ASS
(Advanced Substation Alpha) subtitles files.

This filter accepts the following option in addition to the common
options from the subtitles filter:

shaping
Set the shaping engine

Available values are:

auto
The default libass shaping engine, which is the best available.

simple
Fast, font-agnostic shaper that can do only substitutions

complex
Slower shaper using OpenType for substitutions and positioning

The default is "auto".

atadenoise
Apply an Adaptive Temporal Averaging Denoiser to the video input.

The filter accepts the following options:

0a Set threshold A for 1st plane. Default is 0.02. Valid range is 0
to 0.3.

0b Set threshold B for 1st plane. Default is 0.04. Valid range is 0
to 5.

1a Set threshold A for 2nd plane. Default is 0.02. Valid range is 0
to 0.3.

1b Set threshold B for 2nd plane. Default is 0.04. Valid range is 0
to 5.

2a Set threshold A for 3rd plane. Default is 0.02. Valid range is 0
to 0.3.

2b Set threshold B for 3rd plane. Default is 0.04. Valid range is 0
to 5.

Threshold A is designed to react on abrupt changes in the input
signal and threshold B is designed to react on continuous changes
in the input signal.

s Set number of frames filter will use for averaging. Default is 33.
Must be odd number in range [5, 129].

bbox
Compute the bounding box for the non-black pixels in the input frame
luminance plane.

This filter computes the bounding box containing all the pixels with a
luminance value greater than the minimum allowed value. The parameters
describing the bounding box are printed on the filter log.

The filter accepts the following option:

min_val
Set the minimal luminance value. Default is 16.

blackdetect
Detect video intervals that are (almost) completely black. Can be
useful to detect chapter transitions, commercials, or invalid
recordings. Output lines contains the time for the start, end and
duration of the detected black interval expressed 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 following options:

black_min_duration, d
Set the minimum detected black duration expressed in seconds. It
must be a non-negative floating point number.

Default value is 2.0.

picture_black_ratio_th, pic_th
Set the threshold for considering a picture "black". Express the
minimum value for the ratio:

<nb_black_pixels> / <nb_pixels>

for which a picture is considered black. Default value is 0.98.

pixel_black_th, pix_th
Set the threshold for considering a pixel "black".

The threshold expresses the maximum pixel luminance value for which
a pixel is considered "black". The provided value is scaled
according to the following equation:

<absolute_threshold> = <luminance_minimum_value> + <pixel_black_th> * <luminance_range_size>

luminance_range_size and luminance_minimum_value depend on the
input video format, the range is [0-255] for YUV full-range formats
and [16-235] for YUV non full-range formats.

Default value is 0.10.

The following example sets the maximum pixel threshold to the minimum
value, and detects only black intervals of 2 or more seconds:

blackdetect=d=2:pix_th=0.00

blackframe
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.

It accepts the following parameters:

amount
The percentage of the pixels that have to be below the threshold;
it defaults to 98.

threshold, thresh
The threshold below which a pixel value is considered black; it
defaults to 32.

blend, tblend
Blend two video frames into each other.

The "blend" filter takes two input streams and outputs one stream, the
first input is the "top" layer and second input is "bottom" layer.
Output terminates when shortest input terminates.

The "tblend" (time blend) filter takes two consecutive frames from one
single stream, and outputs the result obtained by blending the new
frame on top of the old frame.

A description of the accepted options follows.

c0_mode
c1_mode
c2_mode
c3_mode
all_mode
Set blend mode for specific pixel component or all pixel components
in case of all_mode. Default value is "normal".

Available values for component modes are:

addition
and
average
burn
darken
difference
difference128
divide
dodge
exclusion
glow
hardlight
hardmix
lighten
linearlight
multiply
negation
normal
or
overlay
phoenix
pinlight
reflect
screen
softlight
subtract
vividlight
xor
c0_opacity
c1_opacity
c2_opacity
c3_opacity
all_opacity
Set blend opacity for specific pixel component or all pixel
components in case of all_opacity. Only used in combination with
pixel component blend modes.

c0_expr
c1_expr
c2_expr
c3_expr
all_expr
Set blend expression for specific pixel component or all pixel
components in case of all_expr. Note that related mode options will
be ignored if those are set.

The expressions can use the following variables:

N The sequential number of the filtered frame, starting from 0.

X
Y the coordinates of the current sample

W
H the width and height of currently filtered plane

SW
SH Width and height scale depending on the currently filtered
plane. It is the ratio between the corresponding luma plane
number of pixels and the current plane ones. E.g. for YUV4:2:0
the values are "1,1" for the luma plane, and "0.5,0.5" for
chroma planes.

T Time of the current frame, expressed in seconds.

TOP, A
Value of pixel component at current location for first video
frame (top layer).

BOTTOM, B
Value of pixel component at current location for second video
frame (bottom layer).

shortest
Force termination when the shortest input terminates. Default is 0.
This option is only defined for the "blend" filter.

repeatlast
Continue applying the last bottom frame after the end of the
stream. A value of 0 disable the filter after the last frame of the
bottom layer is reached. Default is 1. This option is only defined
for the "blend" filter.

Examples

· Apply transition from bottom layer to top layer in first 10
seconds:

blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'

· Apply 1x1 checkerboard effect:

blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'

· Apply uncover left effect:

blend=all_expr='if(gte(N*SW+X,W),A,B)'

· Apply uncover down effect:

blend=all_expr='if(gte(Y-N*SH,0),A,B)'

· Apply uncover up-left effect:

blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'

· Display differences between the current and the previous frame:

tblend=all_mode=difference128

boxblur
Apply a boxblur algorithm to the input video.

It accepts the following parameters:

luma_radius, lr
luma_power, lp
chroma_radius, cr
chroma_power, cp
alpha_radius, ar
alpha_power, ap

A description of the accepted options follows.

luma_radius, lr
chroma_radius, cr
alpha_radius, ar
Set an expression for the box radius in pixels used for blurring
the corresponding input plane.

The radius value must be a non-negative number, and must not be
greater than the value of the expression "min(w,h)/2" for the luma
and alpha planes, and of "min(cw,ch)/2" for the chroma planes.

Default value for luma_radius is "2". If not specified,
chroma_radius and alpha_radius default to the corresponding value
set for luma_radius.

The expressions can contain the following constants:

w
h The input width and height in pixels.

cw
ch The input chroma image width and height in pixels.

hsub
vsub
The horizontal and vertical chroma subsample values. For
example, for the pixel format "yuv422p", hsub is 2 and vsub is
1.

luma_power, lp
chroma_power, cp
alpha_power, ap
Specify how many times the boxblur filter is applied to the
corresponding plane.

Default value for luma_power is 2. If not specified, chroma_power
and alpha_power default to the corresponding value set for
luma_power.

A value of 0 will disable the effect.

Examples

· Apply a boxblur filter with the luma, chroma, and alpha radii set
to 2:

boxblur=luma_radius=2:luma_power=1
boxblur=2:1

· Set the luma radius to 2, and alpha and chroma radius to 0:

boxblur=2:1:cr=0:ar=0

· Set the luma and chroma radii to a fraction of the video dimension:

boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1

codecview
Visualize information exported by some codecs.

Some codecs can export information through frames using side-data or
other means. For example, some MPEG based codecs export motion vectors
through the export_mvs flag in the codec flags2 option.

The filter accepts the following option:

mv Set motion vectors to visualize.

Available flags for mv are:

pf forward predicted MVs of P-frames

bf forward predicted MVs of B-frames

bb backward predicted MVs of B-frames

Examples

· Visualizes multi-directionals MVs from P and B-Frames using ffplay:

ffplay -flags2 +export_mvs input.mpg -vf codecview=mv=pf+bf+bb

colorbalance
Modify intensity of primary colors (red, green and blue) of input
frames.

The filter allows an input frame to be adjusted in the shadows,
midtones or highlights regions for the red-cyan, green-magenta or blue-
yellow balance.

A positive adjustment value shifts the balance towards the primary
color, a negative value towards the complementary color.

The filter accepts the following options:

rs
gs
bs Adjust red, green and blue shadows (darkest pixels).

rm
gm
bm Adjust red, green and blue midtones (medium pixels).

rh
gh
bh Adjust red, green and blue highlights (brightest pixels).

Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.

Examples

· Add red color cast to shadows:

colorbalance=rs=.3

colorkey
RGB colorspace color keying.

The filter accepts the following options:

color
The color which will be replaced with transparency.

similarity
Similarity percentage with the key color.

0.01 matches only the exact key color, while 1.0 matches
everything.

blend
Blend percentage.

0.0 makes pixels either fully transparent, or not transparent at
all.

Higher values result in semi-transparent pixels, with a higher
transparency the more similar the pixels color is to the key color.

Examples

· Make every green pixel in the input image transparent:

ffmpeg -i input.png -vf colorkey=green out.png

· Overlay a greenscreen-video on top of a static background image.

ffmpeg -i background.png -i video.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.flv

colorlevels
Adjust video input frames using levels.

The filter accepts the following options:

rimin
gimin
bimin
aimin
Adjust red, green, blue and alpha input black point. Allowed
ranges for options are "[-1.0, 1.0]". Defaults are 0.

rimax
gimax
bimax
aimax
Adjust red, green, blue and alpha input white point. Allowed
ranges for options are "[-1.0, 1.0]". Defaults are 1.

Input levels are used to lighten highlights (bright tones), darken
shadows (dark tones), change the balance of bright and dark tones.

romin
gomin
bomin
aomin
Adjust red, green, blue and alpha output black point. Allowed
ranges for options are "[0, 1.0]". Defaults are 0.

romax
gomax
bomax
aomax
Adjust red, green, blue and alpha output white point. Allowed
ranges for options are "[0, 1.0]". Defaults are 1.

Output levels allows manual selection of a constrained output level
range.

Examples

· Make video output darker:

colorlevels=rimin=0.058:gimin=0.058:bimin=0.058

· Increase contrast:

colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96

· Make video output lighter:

colorlevels=rimax=0.902:gimax=0.902:bimax=0.902

· Increase brightness:

colorlevels=romin=0.5:gomin=0.5:bomin=0.5

colorchannelmixer
Adjust video input frames by re-mixing color channels.

This filter modifies a color channel by adding the values associated to
the other channels of the same pixels. For example if the value to
modify is red, the output value will be:

The filter accepts the following options:

rr
rg
rb
ra Adjust contribution of input red, green, blue and alpha channels
for output red channel. Default is 1 for rr, and 0 for rg, rb and
ra.

gr
gg
gb
ga Adjust contribution of input red, green, blue and alpha channels
for output green channel. Default is 1 for gg, and 0 for gr, gb
and ga.

br
bg
bb
ba Adjust contribution of input red, green, blue and alpha channels
for output blue channel. Default is 1 for bb, and 0 for br, bg and
ba.

ar
ag
ab
aa Adjust contribution of input red, green, blue and alpha channels
for output alpha channel. Default is 1 for aa, and 0 for ar, ag
and ab.

Allowed ranges for options are "[-2.0, 2.0]".

Examples

· Convert source to grayscale:

colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3

· Simulate sepia tones:

colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131

colormatrix
Convert color matrix.

The filter accepts the following options:

src
dst Specify the source and destination color matrix. Both values must
be specified.

The accepted values are:

bt709
BT.709

bt601
BT.601

smpte240m
SMPTE-240M

fcc FCC

For example to convert from BT.601 to SMPTE-240M, use the command:

colormatrix=bt601:smpte240m

copy
Copy the input source unchanged to the output. This is mainly useful
for testing purposes.

crop
Crop the input video to given dimensions.

It accepts the following parameters:

w, out_w
The width of the output video. It defaults to "iw". This
expression is evaluated only once during the filter configuration,
or when the w or out_w command is sent.

h, out_h
The height of the output video. It defaults to "ih". This
expression is evaluated only once during the filter configuration,
or when the h or out_h command is sent.

x The horizontal position, in the input video, of the left edge of
the output video. It defaults to "(in_w-out_w)/2". This expression
is evaluated per-frame.

y The vertical position, in the input video, of the top edge of the
output video. It defaults to "(in_h-out_h)/2". This expression is
evaluated per-frame.

keep_aspect
If set to 1 will force the output display aspect ratio to be the
same of the input, by changing the output sample aspect ratio. It
defaults to 0.

The out_w, out_h, x, y parameters are expressions containing the
following constants:

x
y The computed values for x and y. They are evaluated for each new
frame.

in_w
in_h
The input width and height.

iw
ih These are the same as in_w and in_h.

out_w
out_h
The output (cropped) width and height.

ow
oh These are the same as out_w and out_h.

a same as iw / ih

sar input sample aspect ratio

dar input display aspect ratio, it is the same as (iw / ih) * sar

hsub
vsub
horizontal and vertical chroma subsample values. For example for
the pixel format "yuv422p" hsub is 2 and vsub is 1.

n The number of the input frame, starting from 0.

pos the position in the file of the input frame, NAN if unknown

t The timestamp expressed in seconds. It's NAN if the input timestamp
is unknown.

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 expression for x may depend on y, and the expression for y may
depend on x.

Examples

· Crop area with size 100x100 at position (12,34).

crop=100:100:12:34

Using named options, the example above becomes:

crop=w=100:h=100:x=12:y=34

· 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=out_w=in_h
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

· Apply 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)

· Apply 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)

Commands

This filter supports the following commands:

w, out_w
h, out_h
x
y Set width/height of the output video and the horizontal/vertical
position in the input video. The command accepts the same syntax
of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

cropdetect
Auto-detect the crop size.

It calculates the necessary cropping parameters and prints the
recommended parameters via the logging system. The detected dimensions
correspond to the non-black area of the input video.

It accepts the following parameters:

limit
Set higher black value threshold, which can be optionally specified
from nothing (0) to everything (255 for 8bit based formats). An
intensity value greater to the set value is considered non-black.
It defaults to 24. You can also specify a value between 0.0 and
1.0 which will be scaled depending on the bitdepth of the pixel
format.

round
The value which the width/height should be divisible by. It
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_count, reset
Set the 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. Default value is 0.

This can be useful when channel logos distort the video area. 0
indicates 'never reset', and returns the largest area encountered
during playback.

curves
Apply color adjustments using curves.

This filter is similar to the Adobe Photoshop and GIMP curves tools.
Each component (red, green and blue) has its values defined by N key
points tied from each other using a smooth curve. The x-axis represents
the pixel values from the input frame, and the y-axis the new pixel
values to be set for the output frame.

By default, a component curve is defined by the two points (0;0) and
(1;1). This creates a straight line where each original pixel value is
"adjusted" to its own value, which means no change to the image.

The filter allows you to redefine these two points and add some more. A
new curve (using a natural cubic spline interpolation) will be define
to pass smoothly through all these new coordinates. The new defined
points needs to be strictly increasing over the x-axis, and their x and
y values must be in the [0;1] interval. If the computed curves
happened to go outside the vector spaces, the values will be clipped
accordingly.

If there is no key point defined in "x=0", the filter will
automatically insert a (0;0) point. In the same way, if there is no key
point defined in "x=1", the filter will automatically insert a (1;1)
point.

The filter accepts the following options:

preset
Select one of the available color presets. This option can be used
in addition to the r, g, b parameters; in this case, the later
options takes priority on the preset values. Available presets
are:

none
color_negative
cross_process
darker
increase_contrast
lighter
linear_contrast
medium_contrast
negative
strong_contrast
vintage

Default is "none".

master, m
Set the master key points. These points will define a second pass
mapping. It is sometimes called a "luminance" or "value" mapping.
It can be used with r, g, b or all since it acts like a post-
processing LUT.

red, r
Set the key points for the red component.

green, g
Set the key points for the green component.

blue, b
Set the key points for the blue component.

all Set the key points for all components (not including master). Can
be used in addition to the other key points component options. In
this case, the unset component(s) will fallback on this all
setting.

psfile
Specify a Photoshop curves file (".asv") to import the settings
from.

To avoid some filtergraph syntax conflicts, each key points list need
to be defined using the following syntax: "x0/y0 x1/y1 x2/y2 ...".

Examples

· Increase slightly the middle level of blue:

curves=blue='0.5/0.58'

· Vintage effect:

curves=r='0/0.11 .42/.51 1/0.95':g='0.50/0.48':b='0/0.22 .49/.44 1/0.8'

Here we obtain the following coordinates for each components:

red "(0;0.11) (0.42;0.51) (1;0.95)"

green
"(0;0) (0.50;0.48) (1;1)"

blue
"(0;0.22) (0.49;0.44) (1;0.80)"

· The previous example can also be achieved with the associated
built-in preset:

curves=preset=vintage

· Or simply:

curves=vintage

· Use a Photoshop preset and redefine the points of the green
component:

curves=psfile='MyCurvesPresets/purple.asv':green='0.45/0.53'

dctdnoiz
Denoise frames using 2D DCT (frequency domain filtering).

This filter is not designed for real time.

The filter accepts the following options:

sigma, s
Set the noise sigma constant.

This sigma defines a hard threshold of "3 * sigma"; every DCT
coefficient (absolute value) below this threshold with be dropped.

If you need a more advanced filtering, see expr.

Default is 0.

overlap
Set number overlapping pixels for each block. Since the filter can
be slow, you may want to reduce this value, at the cost of a less
effective filter and the risk of various artefacts.

If the overlapping value doesn't permit processing the whole input
width or height, a warning will be displayed and according borders
won't be denoised.

Default value is blocksize-1, which is the best possible setting.

expr, e
Set the coefficient factor expression.

For each coefficient of a DCT block, this expression will be
evaluated as a multiplier value for the coefficient.

If this is option is set, the sigma option will be ignored.

The absolute value of the coefficient can be accessed through the c
variable.

n Set the blocksize using the number of bits. "1<<n" defines the
blocksize, which is the width and height of the processed blocks.

The default value is 3 (8x8) and can be raised to 4 for a blocksize
of 16x16. Note that changing this setting has huge consequences on
the speed processing. Also, a larger block size does not
necessarily means a better de-noising.

Examples

Apply a denoise with a sigma of 4.5:

dctdnoiz=4.5

The same operation can be achieved using the expression system:

dctdnoiz=e='gte(c, 4.5*3)'

Violent denoise using a block size of "16x16":

dctdnoiz=15:n=4

deband
Remove banding artifacts from input video. It works by replacing
banded pixels with average value of referenced pixels.

The filter accepts the following options:

1thr
2thr
3thr
4thr
Set banding detection threshold for each plane. Default is 0.02.
Valid range is 0.00003 to 0.5. If difference between current pixel
and reference pixel is less than threshold, it will be considered
as banded.

range, r
Banding detection range in pixels. Default is 16. If positive,
random number in range 0 to set value will be used. If negative,
exact absolute value will be used. The range defines square of
four pixels around current pixel.

direction, d
Set direction in radians from which four pixel will be compared. If
positive, random direction from 0 to set direction will be picked.
If negative, exact of absolute value will be picked. For example
direction 0, -PI or -2*PI radians will pick only pixels on same row
and -PI/2 will pick only pixels on same column.

blur
If enabled, current pixel is compared with average value of all
four surrounding pixels. The default is enabled. If disabled
current pixel is compared with all four surrounding pixels. The
pixel is considered banded if only all four differences with
surrounding pixels are less than threshold.

decimate
Drop duplicated frames at regular intervals.

The filter accepts the following options:

cycle
Set the number of frames from which one will be dropped. Setting
this to N means one frame in every batch of N frames will be
dropped. Default is 5.

dupthresh
Set the threshold for duplicate detection. If the difference metric
for a frame is less than or equal to this value, then it is
declared as duplicate. Default is 1.1

scthresh
Set scene change threshold. Default is 15.

blockx
blocky
Set the size of the x and y-axis blocks used during metric
calculations. Larger blocks give better noise suppression, but
also give worse detection of small movements. Must be a power of
two. Default is 32.

ppsrc
Mark main input as a pre-processed input and activate clean source
input stream. This allows the input to be pre-processed with
various filters to help the metrics calculation while keeping the
frame selection lossless. When set to 1, the first stream is for
the pre-processed input, and the second stream is the clean source
from where the kept frames are chosen. Default is 0.

chroma
Set whether or not chroma is considered in the metric calculations.
Default is 1.

deflate
Apply deflate effect to the video.

This filter replaces the pixel by the local(3x3) average by taking into
account only values lower than the pixel.

It accepts the following options:

threshold0
threshold1
threshold2
threshold3
Limit the maximum change for each plane, default is 65535. If 0,
plane will remain unchanged.

dejudder
Remove judder produced by partially interlaced telecined content.

Judder can be introduced, for instance, by pullup filter. If the
original source was partially telecined content then the output of
"pullup,dejudder" will have a variable frame rate. May change the
recorded frame rate of the container. Aside from that change, this
filter will not affect constant frame rate video.

The option available in this filter is:

cycle
Specify the length of the window over which the judder repeats.

Accepts any integer greater than 1. Useful values are:

4 If the original was telecined from 24 to 30 fps (Film to NTSC).

5 If the original was telecined from 25 to 30 fps (PAL to NTSC).

20 If a mixture of the two.

The default is 4.

delogo
Suppress a TV station logo by a simple interpolation of the surrounding
pixels. Just set a rectangle covering the logo and watch it disappear
(and sometimes something even uglier appear - your mileage may vary).

It accepts the following parameters:

x
y Specify the top left corner coordinates of the logo. They must be
specified.

w
h Specify the width and height of the logo to clear. They must be
specified.

band, t
Specify the thickness of the fuzzy edge of the rectangle (added to
w and h). The default value is 4.

show
When set to 1, a green rectangle is drawn on the screen to simplify
finding the right x, y, w, and h parameters. The default value is
0.

The rectangle is drawn on the outermost pixels which will be
(partly) replaced with interpolated values. The values of the next
pixels immediately outside this rectangle in each direction will be
used to compute the interpolated pixel values inside the rectangle.

Examples

· Set a rectangle covering the area with top left corner coordinates
0,0 and size 100x77, and a band of size 10:

delogo=x=0:y=0:w=100:h=77:band=10

deshake
Attempt to fix small changes in horizontal and/or vertical shift. This
filter helps remove camera shake from hand-holding a camera, bumping a
tripod, moving on a vehicle, etc.

The filter accepts the following options:

x
y
w
h Specify a rectangular area where to limit the search for motion
vectors. If desired the search for motion vectors can be limited
to a rectangular area of the frame defined by its top left corner,
width and height. These parameters have the same meaning as the
drawbox filter which can be used to visualise the position of the
bounding box.

This is useful when simultaneous movement of subjects within the
frame might be confused for camera motion by the motion vector
search.

If any or all of x, y, w and h are set to -1 then the full frame is
used. This allows later options to be set without specifying the
bounding box for the motion vector search.

Default - search the whole frame.

rx
ry Specify the maximum extent of movement in x and y directions in the
range 0-64 pixels. Default 16.

edge
Specify how to generate pixels to fill blanks at the edge of the
frame. Available values are:

blank, 0
Fill zeroes at blank locations

original, 1
Original image at blank locations

clamp, 2
Extruded edge value at blank locations

mirror, 3
Mirrored edge at blank locations

Default value is mirror.

blocksize
Specify the blocksize to use for motion search. Range 4-128 pixels,
default 8.

contrast
Specify the contrast threshold for blocks. Only blocks with more
than the specified contrast (difference between darkest and
lightest pixels) will be considered. Range 1-255, default 125.

search
Specify the search strategy. Available values are:

exhaustive, 0
Set exhaustive search

less, 1
Set less exhaustive search.

Default value is exhaustive.

filename
If set then a detailed log of the motion search is written to the
specified file.

opencl
If set to 1, specify using OpenCL capabilities, only available if
FFmpeg was configured with "--enable-opencl". Default value is 0.

detelecine
Apply an exact inverse of the telecine operation. It requires a
predefined pattern specified using the pattern option which must be the
same as that passed to the telecine filter.

This filter accepts the following options:

first_field
top, t
top field first

bottom, b
bottom field first The default value is "top".

pattern
A string of numbers representing the pulldown pattern you wish to
apply. The default value is 23.

start_frame
A number representing position of the first frame with respect to
the telecine pattern. This is to be used if the stream is cut. The
default value is 0.

dilation
Apply dilation effect to the video.

This filter replaces the pixel by the local(3x3) maximum.

It accepts the following options:

threshold0
threshold1
threshold2
threshold3
Limit the maximum change for each plane, default is 65535. If 0,
plane will remain unchanged.

coordinates
Flag which specifies the pixel to refer to. Default is 255 i.e. all
eight pixels are used.

Flags to local 3x3 coordinates maps like this:

1 2 3
4 5
6 7 8

drawbox
Draw a colored box on the input image.

It accepts the following parameters:

x
y The expressions which specify the top left corner coordinates of
the box. It defaults to 0.

width, w
height, h
The expressions which specify the width and height of the box; if 0
they are interpreted as the input width and height. It defaults to
0.

color, c
Specify the color of the box to write. For the general syntax of
this option, check the "Color" section in the ffmpeg-utils manual.
If the special value "invert" is used, the box edge color is the
same as the video with inverted luma.

thickness, t
The expression which sets the thickness of the box edge. Default
value is 3.

See below for the list of accepted constants.

The parameters for x, y, w and h and t are expressions containing the
following constants:

dar The input display aspect ratio, it is the same as (w / h) * sar.

hsub
vsub
horizontal and vertical chroma subsample values. For example for
the pixel format "yuv422p" hsub is 2 and vsub is 1.

in_h, ih
in_w, iw
The input width and height.

sar The input sample aspect ratio.

x
y The x and y offset coordinates where the box is drawn.

w
h The width and height of the drawn box.

t The thickness of the drawn box.

These constants allow the x, y, w, h and t expressions to refer to
each other, so you may for example specify "y=x/dar" or "h=w/dar".

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

The previous example can be specified as:

drawbox=x=10:y=20:w=200:h=60:color=red@0.5

· Fill the box with pink color:

drawbox=x=10:y=10:w=100:h=100:color=pink@0.5:t=max

· Draw a 2-pixel red 2.40:1 mask:

drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red

drawgraph, adrawgraph
Draw a graph using input video or audio metadata.

It accepts the following parameters:

m1 Set 1st frame metadata key from which metadata values will be used
to draw a graph.

fg1 Set 1st foreground color expression.

m2 Set 2nd frame metadata key from which metadata values will be used
to draw a graph.

fg2 Set 2nd foreground color expression.

m3 Set 3rd frame metadata key from which metadata values will be used
to draw a graph.

fg3 Set 3rd foreground color expression.

m4 Set 4th frame metadata key from which metadata values will be used
to draw a graph.

fg4 Set 4th foreground color expression.

min Set minimal value of metadata value.

max Set maximal value of metadata value.

bg Set graph background color. Default is white.

mode
Set graph mode.

Available values for mode is:

bar
dot
line

Default is "line".

slide
Set slide mode.

Available values for slide is:

frame
Draw new frame when right border is reached.

replace
Replace old columns with new ones.

scroll
Scroll from right to left.

rscroll
Scroll from left to right.

Default is "frame".

size
Set size of graph video. For the syntax of this option, check the
"Video size" section in the ffmpeg-utils manual. The default value
is "900x256".

The foreground color expressions can use the following variables:

MIN Minimal value of metadata value.

MAX Maximal value of metadata value.

VAL Current metadata key value.

The color is defined as 0xAABBGGRR.

Example using metadata from signalstats filter:

signalstats,drawgraph=lavfi.signalstats.YAVG:min=0:max=255

Example using metadata from ebur128 filter:

ebur128=metadata=1,adrawgraph=lavfi.r128.M:min=-120:max=5

drawgrid
Draw a grid on the input image.

It accepts the following parameters:

x
y The expressions which specify the coordinates of some point of grid
intersection (meant to configure offset). Both default to 0.

width, w
height, h
The expressions which specify the width and height of the grid
cell, if 0 they are interpreted as the input width and height,
respectively, minus "thickness", so image gets framed. Default to
0.

color, c
Specify the color of the grid. For the general syntax of this
option, check the "Color" section in the ffmpeg-utils manual. If
the special value "invert" is used, the grid color is the same as
the video with inverted luma.

thickness, t
The expression which sets the thickness of the grid line. Default
value is 1.

See below for the list of accepted constants.

The parameters for x, y, w and h and t are expressions containing the
following constants:

dar The input display aspect ratio, it is the same as (w / h) * sar.

hsub
vsub
horizontal and vertical chroma subsample values. For example for
the pixel format "yuv422p" hsub is 2 and vsub is 1.

in_h, ih
in_w, iw
The input grid cell width and height.

sar The input sample aspect ratio.

x
y The x and y coordinates of some point of grid intersection (meant
to configure offset).

w
h The width and height of the drawn cell.

t The thickness of the drawn cell.

These constants allow the x, y, w, h and t expressions to refer to
each other, so you may for example specify "y=x/dar" or "h=w/dar".

Examples

· Draw a grid with cell 100x100 pixels, thickness 2 pixels, with
color red and an opacity of 50%:

drawgrid=width=100:height=100:thickness=2:color=red@0.5

· Draw a white 3x3 grid with an opacity of 50%:

drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5

drawtext
Draw a text string or text from a specified file on top of a video,
using the libfreetype library.

To enable compilation of this filter, you need to configure FFmpeg with
"--enable-libfreetype". To enable default font fallback and the font
option you need to configure FFmpeg with "--enable-libfontconfig". To
enable the text_shaping option, you need to configure FFmpeg with
"--enable-libfribidi".

Syntax

It accepts the following parameters:

box Used to draw a box around text using the background color. The
value must be either 1 (enable) or 0 (disable). The default value
of box is 0.

boxborderw
Set the width of the border to be drawn around the box using
boxcolor. The default value of boxborderw is 0.

boxcolor
The color to be used for drawing box around text. For the syntax of
this option, check the "Color" section in the ffmpeg-utils manual.

The default value of boxcolor is "white".

borderw
Set the width of the border to be drawn around the text using
bordercolor. The default value of borderw is 0.

bordercolor
Set the color to be used for drawing border around text. For the
syntax of this option, check the "Color" section in the ffmpeg-
utils manual.

The default value of bordercolor is "black".

expansion
Select how the text is expanded. Can be either "none", "strftime"
(deprecated) or "normal" (default). See the drawtext_expansion,
Text expansion section below for details.

fix_bounds
If true, check and fix text coords to avoid clipping.

fontcolor
The color to be used for drawing fonts. For the syntax of this
option, check the "Color" section in the ffmpeg-utils manual.

The default value of fontcolor is "black".

fontcolor_expr
String which is expanded the same way as text to obtain dynamic
fontcolor value. By default this option has empty value and is not
processed. When this option is set, it overrides fontcolor option.

font
The font family to be used for drawing text. By default Sans.

fontfile
The font file to be used for drawing text. The path must be
included. This parameter is mandatory if the fontconfig support is
disabled.

draw
This option does not exist, please see the timeline system

alpha
Draw the text applying alpha blending. The value can be either a
number between 0.0 and 1.0 The expression accepts the same
variables x, y do. The default value is 1. Please see
fontcolor_expr

fontsize
The font size to be used for drawing text. The default value of
fontsize is 16.

text_shaping
If set to 1, attempt to shape the text (for example, reverse the
order of right-to-left text and join Arabic characters) before
drawing it. Otherwise, just draw the text exactly as given. By
default 1 (if supported).

ft_load_flags
The 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

Default value is "default".

For more information consult the documentation for the FT_LOAD_*
libfreetype flags.

shadowcolor
The color to be used for drawing a shadow behind the drawn text.
For the syntax of this option, check the "Color" section in the
ffmpeg-utils manual.

The default value of shadowcolor is "black".

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. The default value for both is "0".

start_number
The starting frame number for the n/frame_num variable. The default
value is "0".

tabsize
The size in number of spaces to use for rendering the tab. Default
value is 4.

timecode
Set the initial timecode representation in "hh:mm:ss[:;.]ff"
format. It can be used with or without text parameter.
timecode_rate option must be specified.

timecode_rate, rate, r
Set the timecode frame rate (timecode only).

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.

reload
If set to 1, the textfile will be reloaded before each frame. Be
sure to update it atomically, or it may be read partially, or even
fail.

x
y The expressions which specify the offsets where text will be drawn
within the video frame. They are relative to the top/left border of
the output image.

The default value of x and y is "0".

See below for the list of accepted constants and functions.

The parameters for x and y are expressions containing the following
constants and functions:

dar input display aspect ratio, it is the same as (w / h) * sar

hsub
vsub
horizontal and vertical chroma subsample values. For example for
the pixel format "yuv422p" hsub is 2 and vsub is 1.

line_h, lh
the height of each text line

main_h, h, H
the input height

main_w, w, W
the input width

max_glyph_a, ascent
the maximum distance from the baseline to the highest/upper grid
coordinate used to place a glyph outline point, for all the
rendered glyphs. It is a positive value, due to the grid's
orientation with the Y axis upwards.

max_glyph_d, descent
the maximum distance from the baseline to the lowest grid
coordinate used to place a glyph outline point, for all the
rendered glyphs. This is a negative value, due to the grid's
orientation, with the Y axis upwards.

max_glyph_h
maximum glyph height, that is the maximum height for all the glyphs
contained in the rendered text, it is equivalent to ascent -
descent.

max_glyph_w
maximum glyph width, that is the maximum width for all the glyphs
contained in the rendered text

n the number of input frame, starting from 0

rand(min, max)
return a random number included between min and max

sar The input sample aspect ratio.

t timestamp expressed in seconds, NAN if the input timestamp is
unknown

text_h, th
the height of the rendered text

text_w, tw
the width of the rendered text

x
y the x and y offset coordinates where the text is drawn.

These parameters allow the x and y expressions to refer each other,
so you can for example specify "y=x/dar".

Text expansion

If expansion is set to "strftime", the filter recognizes strftime()
sequences in the provided text and expands them accordingly. Check the
documentation of strftime(). This feature is deprecated.

If expansion is set to "none", the text is printed verbatim.

If expansion is set to "normal" (which is the default), the following
expansion mechanism is used.

The backslash character \, followed by any character, always expands to
the second character.

Sequence of the form "%{...}" are expanded. The text between the braces
is a function name, possibly followed by arguments separated by ':'.
If the arguments contain special characters or delimiters (':' or '}'),
they should be escaped.

Note that they probably must also be escaped as the value for the text
option in the filter argument string and as the filter argument in the
filtergraph description, and possibly also for the shell, that makes up
to four levels of escaping; using a text file avoids these problems.

The following functions are available:

expr, e
The expression evaluation result.

It must take one argument specifying the expression to be
evaluated, which accepts the same constants and functions as the x
and y values. Note that not all constants should be used, for
example the text size is not known when evaluating the expression,
so the constants text_w and text_h will have an undefined value.

expr_int_format, eif
Evaluate the expression's value and output as formatted integer.

The first argument is the expression to be evaluated, just as for
the expr function. The second argument specifies the output
format. Allowed values are x, X, d and u. They are treated exactly
as in the "printf" function. The third parameter is optional and
sets the number of positions taken by the output. It can be used
to add padding with zeros from the left.

gmtime
The time at which the filter is running, expressed in UTC. It can
accept an argument: a strftime() format string.

localtime
The time at which the filter is running, expressed in the local
time zone. It can accept an argument: a strftime() format string.

metadata
Frame metadata. It must take one argument specifying metadata key.

n, frame_num
The frame number, starting from 0.

pict_type
A 1 character description of the current picture type.

pts The timestamp of the current frame. It can take up to two
arguments.

The first argument is the format of the timestamp; it defaults to
"flt" for seconds as a decimal number with microsecond accuracy;
"hms" stands for a formatted [-]HH:MM:SS.mmm timestamp with
millisecond accuracy.

The second argument is an offset added to the timestamp.

Examples

· Draw "Test Text" with font FreeSerif, using the default values for
the optional parameters.

drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'"

· 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%.

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"

Note that the double quotes are not necessary if spaces are not
used within the parameter list.

· Show the text at the center of the video frame:

drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2"

· Show a text line sliding from right to left in the last row of the
video frame. The file LONG_LINE is assumed to contain a single line
with no newlines.

drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t"

· Show the content of file CREDITS off the bottom of the frame and
scroll up.

drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t"

· Draw a single green letter "g", at the center of the input video.
The glyph baseline is placed at half screen height.

drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent"

· Show text for 1 second every 3 seconds:

drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'"

· Use fontconfig to set the font. Note that the colons need to be
escaped.

drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg'

· Print the date of a real-time encoding (see strftime(3)):

drawtext='fontfile=FreeSans.ttf:text=%{localtime\:%a %b %d %Y}'

· Show text fading in and out (appearing/disappearing):

#!/bin/sh
DS=1.0 # display start
DE=10.0 # display end
FID=1.5 # fade in duration
FOD=5 # fade out duration
ffplay -f lavfi "color,drawtext=text=TEST:fontsize=50:fontfile=FreeSerif.ttf:fontcolor_expr=ff0000%{eif\\\\: clip(255*(1*between(t\\, $DS + $FID\\, $DE - $FOD) + ((t - $DS)/$FID)*between(t\\, $DS\\, $DS + $FID) + (-(t - $DE)/$FOD)*between(t\\, $DE - $FOD\\, $DE) )\\, 0\\, 255) \\\\: x\\\\: 2 }"

For more information about libfreetype, check:
<http://www.freetype.org/>.

For more information about fontconfig, check:
<http://freedesktop.org/software/fontconfig/fontconfig-user.html>.

For more information about libfribidi, check: <http://fribidi.org/>.

edgedetect
Detect and draw edges. The filter uses the Canny Edge Detection
algorithm.

The filter accepts the following options:

low
high
Set low and high threshold values used by the Canny thresholding
algorithm.

The high threshold selects the "strong" edge pixels, which are then
connected through 8-connectivity with the "weak" edge pixels
selected by the low threshold.

low and high threshold values must be chosen in the range [0,1],
and low should be lesser or equal to high.

Default value for low is "20/255", and default value for high is
"50/255".

mode
Define the drawing mode.

wires
Draw white/gray wires on black background.

colormix
Mix the colors to create a paint/cartoon effect.

Default value is wires.

Examples

· Standard edge detection with custom values for the hysteresis
thresholding:

edgedetect=low=0.1:high=0.4

· Painting effect without thresholding:

edgedetect=mode=colormix:high=0

eq
Set brightness, contrast, saturation and approximate gamma adjustment.

The filter accepts the following options:

contrast
Set the contrast expression. The value must be a float value in
range "-2.0" to 2.0. The default value is "0".

brightness
Set the brightness expression. The value must be a float value in
range "-1.0" to 1.0. The default value is "0".

saturation
Set the saturation expression. The value must be a float in range
0.0 to 3.0. The default value is "1".

gamma
Set the gamma expression. The value must be a float in range 0.1 to
10.0. The default value is "1".

gamma_r
Set the gamma expression for red. The value must be a float in
range 0.1 to 10.0. The default value is "1".

gamma_g
Set the gamma expression for green. The value must be a float in
range 0.1 to 10.0. The default value is "1".

gamma_b
Set the gamma expression for blue. The value must be a float in
range 0.1 to 10.0. The default value is "1".

gamma_weight
Set the gamma weight expression. It can be used to reduce the
effect of a high gamma value on bright image areas, e.g. keep them
from getting overamplified and just plain white. The value must be
a float in range 0.0 to 1.0. A value of 0.0 turns the gamma
correction all the way down while 1.0 leaves it at its full
strength. Default is "1".

eval
Set when the expressions for brightness, contrast, saturation and
gamma expressions are evaluated.

It accepts the following values:

init
only evaluate expressions once during the filter initialization
or when a command is processed

frame
evaluate expressions for each incoming frame

Default value is init.

The expressions accept the following parameters:

n frame count of the input frame starting from 0

pos byte position of the corresponding packet in the input file, NAN if
unspecified

r frame rate of the input video, NAN if the input frame rate is
unknown

t timestamp expressed in seconds, NAN if the input timestamp is
unknown

Commands

The filter supports the following commands:

contrast
Set the contrast expression.

brightness
Set the brightness expression.

saturation
Set the saturation expression.

gamma
Set the gamma expression.

gamma_r
Set the gamma_r expression.

gamma_g
Set gamma_g expression.

gamma_b
Set gamma_b expression.

gamma_weight
Set gamma_weight expression.

The command accepts the same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

erosion
Apply erosion effect to the video.

This filter replaces the pixel by the local(3x3) minimum.

It accepts the following options:

threshold0
threshold1
threshold2
threshold3
Limit the maximum change for each plane, default is 65535. If 0,
plane will remain unchanged.

coordinates
Flag which specifies the pixel to refer to. Default is 255 i.e. all
eight pixels are used.

Flags to local 3x3 coordinates maps like this:

1 2 3
4 5
6 7 8

extractplanes
Extract color channel components from input video stream into separate
grayscale video streams.

The filter accepts the following option:

planes
Set plane(s) to extract.

Available values for planes are:

y
u
v
a
r
g
b

Choosing planes not available in the input will result in an error.
That means you cannot select "r", "g", "b" planes with "y", "u",
"v" planes at same time.

Examples

· Extract luma, u and v color channel component from input video
frame into 3 grayscale outputs:

ffmpeg -i video.avi -filter_complex 'extractplanes=y+u+v[y][u][v]' -map '[y]' y.avi -map '[u]' u.avi -map '[v]' v.avi

elbg
Apply a posterize effect using the ELBG (Enhanced LBG) algorithm.

For each input image, the filter will compute the optimal mapping from
the input to the output given the codebook length, that is the number
of distinct output colors.

This filter accepts the following options.

codebook_length, l
Set codebook length. The value must be a positive integer, and
represents the number of distinct output colors. Default value is
256.

nb_steps, n
Set the maximum number of iterations to apply for computing the
optimal mapping. The higher the value the better the result and the
higher the computation time. Default value is 1.

seed, s
Set a random seed, must be an integer included between 0 and
UINT32_MAX. If not specified, or if explicitly set to -1, the
filter will try to use a good random seed on a best effort basis.

pal8
Set pal8 output pixel format. This option does not work with
codebook length greater than 256.

fade
Apply a fade-in/out effect to the input video.

It accepts the following parameters:

type, t
The effect type can be either "in" for a fade-in, or "out" for a
fade-out effect. Default is "in".

start_frame, s
Specify the number of the frame to start applying the fade effect
at. Default is 0.

nb_frames, n
The number of frames that the fade effect lasts. 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 filled with the selected color. Default is 25.

alpha
If set to 1, fade only alpha channel, if one exists on the input.
Default value is 0.

start_time, st
Specify the timestamp (in seconds) of the frame to start to apply
the fade effect. If both start_frame and start_time are specified,
the fade will start at whichever comes last. Default is 0.

duration, d
The number of seconds 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 filled with the selected color. If both
duration and nb_frames are specified, duration is used. Default is
0 (nb_frames is used by default).

color, c
Specify the color of the fade. Default is "black".

Examples

· Fade in the first 30 frames of video:

fade=in:0:30

The command above is equivalent to:

fade=t=in:s=0:n=30

· Fade out the last 45 frames of a 200-frame video:

fade=out:155:45
fade=type=out:start_frame=155:nb_frames=45

· Fade in the first 25 frames and fade out the last 25 frames of a
1000-frame video:

fade=in:0:25, fade=out:975:25

· Make the first 5 frames yellow, then fade in from frame 5-24:

fade=in:5:20:color=yellow

· Fade in alpha over first 25 frames of video:

fade=in:0:25:alpha=1

· Make the first 5.5 seconds black, then fade in for 0.5 seconds:

fade=t=in:st=5.5:d=0.5

fftfilt
Apply arbitrary expressions to samples in frequency domain

dc_Y
Adjust the dc value (gain) of the luma plane of the image. The
filter accepts an integer value in range 0 to 1000. The default
value is set to 0.

dc_U
Adjust the dc value (gain) of the 1st chroma plane of the image.
The filter accepts an integer value in range 0 to 1000. The default
value is set to 0.

dc_V
Adjust the dc value (gain) of the 2nd chroma plane of the image.
The filter accepts an integer value in range 0 to 1000. The default
value is set to 0.

weight_Y
Set the frequency domain weight expression for the luma plane.

weight_U
Set the frequency domain weight expression for the 1st chroma
plane.

weight_V
Set the frequency domain weight expression for the 2nd chroma
plane.

The filter accepts the following variables:

X
Y The coordinates of the current sample.

W
H The width and height of the image.

Examples

· High-pass:

fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'

· Low-pass:

fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'

· Sharpen:

fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'

field
Extract a single field from an interlaced image using stride arithmetic
to avoid wasting CPU time. The output frames are marked as non-
interlaced.

The filter accepts the following options:

type
Specify whether to extract the top (if the value is 0 or "top") or
the bottom field (if the value is 1 or "bottom").

fieldmatch
Field matching filter for inverse telecine. It is meant to reconstruct
the progressive frames from a telecined stream. The filter does not
drop duplicated frames, so to achieve a complete inverse telecine
"fieldmatch" needs to be followed by a decimation filter such as
decimate in the filtergraph.

The separation of the field matching and the decimation is notably
motivated by the possibility of inserting a de-interlacing filter
fallback between the two. If the source has mixed telecined and real
interlaced content, "fieldmatch" will not be able to match fields for
the interlaced parts. But these remaining combed frames will be marked
as interlaced, and thus can be de-interlaced by a later filter such as
yadif before decimation.

In addition to the various configuration options, "fieldmatch" can take
an optional second stream, activated through the ppsrc option. If
enabled, the frames reconstruction will be based on the fields and
frames from this second stream. This allows the first input to be pre-
processed in order to help the various algorithms of the filter, while
keeping the output lossless (assuming the fields are matched properly).
Typically, a field-aware denoiser, or brightness/contrast adjustments
can help.

Note that this filter uses the same algorithms as TIVTC/TFM (AviSynth
project) and VIVTC/VFM (VapourSynth project). The later is a light
clone of TFM from which "fieldmatch" is based on. While the semantic
and usage are very close, some behaviour and options names can differ.

The decimate filter currently only works for constant frame rate input.
If your input has mixed telecined (30fps) and progressive content with
a lower framerate like 24fps use the following filterchain to produce
the necessary cfr stream:
"dejudder,fps=30000/1001,fieldmatch,decimate".

The filter accepts the following options:

order
Specify the assumed field order of the input stream. Available
values are:

auto
Auto detect parity (use FFmpeg's internal parity value).

bff Assume bottom field first.

tff Assume top field first.

Note that it is sometimes recommended not to trust the parity
announced by the stream.

Default value is auto.

mode
Set the matching mode or strategy to use. pc mode is the safest in
the sense that it won't risk creating jerkiness due to duplicate
frames when possible, but if there are bad edits or blended fields
it will end up outputting combed frames when a good match might
actually exist. On the other hand, pcn_ub mode is the most risky in
terms of creating jerkiness, but will almost always find a good
frame if there is one. The other values are all somewhere in
between pc and pcn_ub in terms of risking jerkiness and creating
duplicate frames versus finding good matches in sections with bad
edits, orphaned fields, blended fields, etc.

More details about p/c/n/u/b are available in p/c/n/u/b meaning
section.

Available values are:

pc 2-way matching (p/c)

pc_n
2-way matching, and trying 3rd match if still combed (p/c + n)

pc_u
2-way matching, and trying 3rd match (same order) if still
combed (p/c + u)

pc_n_ub
2-way matching, trying 3rd match if still combed, and trying
4th/5th matches if still combed (p/c + n + u/b)

pcn 3-way matching (p/c/n)

pcn_ub
3-way matching, and trying 4th/5th matches if all 3 of the
original matches are detected as combed (p/c/n + u/b)

The parenthesis at the end indicate the matches that would be used
for that mode assuming order=tff (and field on auto or top).

In terms of speed pc mode is by far the fastest and pcn_ub is the
slowest.

Default value is pc_n.

ppsrc
Mark the main input stream as a pre-processed input, and enable the
secondary input stream as the clean source to pick the fields from.
See the filter introduction for more details. It is similar to the
clip2 feature from VFM/TFM.

Default value is 0 (disabled).

field
Set the field to match from. It is recommended to set this to the
same value as order unless you experience matching failures with
that setting. In certain circumstances changing the field that is
used to match from can have a large impact on matching performance.
Available values are:

auto
Automatic (same value as order).

bottom
Match from the bottom field.

top Match from the top field.

Default value is auto.

mchroma
Set whether or not chroma is included during the match comparisons.
In most cases it is recommended to leave this enabled. You should
set this to 0 only if your clip has bad chroma problems such as
heavy rainbowing or other artifacts. Setting this to 0 could also
be used to speed things up at the cost of some accuracy.

Default value is 1.

y0
y1 These define an exclusion band which excludes the lines between y0
and y1 from being included in the field matching decision. An
exclusion band can be used to ignore subtitles, a logo, or other
things that may interfere with the matching. y0 sets the starting
scan line and y1 sets the ending line; all lines in between y0 and
y1 (including y0 and y1) will be ignored. Setting y0 and y1 to the
same value will disable the feature. y0 and y1 defaults to 0.

scthresh
Set the scene change detection threshold as a percentage of maximum
change on the luma plane. Good values are in the "[8.0, 14.0]"
range. Scene change detection is only relevant in case
combmatch=sc. The range for scthresh is "[0.0, 100.0]".

Default value is 12.0.

combmatch
When combatch is not none, "fieldmatch" will take into account the
combed scores of matches when deciding what match to use as the
final match. Available values are:

none
No final matching based on combed scores.

sc Combed scores are only used when a scene change is detected.

full
Use combed scores all the time.

Default is sc.

combdbg
Force "fieldmatch" to calculate the combed metrics for certain
matches and print them. This setting is known as micout in TFM/VFM
vocabulary. Available values are:

none
No forced calculation.

pcn Force p/c/n calculations.

pcnub
Force p/c/n/u/b calculations.

Default value is none.

cthresh
This is the area combing threshold used for combed frame detection.
This essentially controls how "strong" or "visible" combing must be
to be detected. Larger values mean combing must be more visible
and smaller values mean combing can be less visible or strong and
still be detected. Valid settings are from "-1" (every pixel will
be detected as combed) to 255 (no pixel will be detected as
combed). This is basically a pixel difference value. A good range
is "[8, 12]".

Default value is 9.

chroma
Sets whether or not chroma is considered in the combed frame
decision. Only disable this if your source has chroma problems
(rainbowing, etc.) that are causing problems for the combed frame
detection with chroma enabled. Actually, using chroma=0 is usually
more reliable, except for the case where there is chroma only
combing in the source.

Default value is 0.

blockx
blocky
Respectively set the x-axis and y-axis size of the window used
during combed frame detection. This has to do with the size of the
area in which combpel pixels are required to be detected as combed
for a frame to be declared combed. See the combpel parameter
description for more info. Possible values are any number that is
a power of 2 starting at 4 and going up to 512.

Default value is 16.

combpel
The number of combed pixels inside any of the blocky by blockx size
blocks on the frame for the frame to be detected as combed. While
cthresh controls how "visible" the combing must be, this setting
controls "how much" combing there must be in any localized area (a
window defined by the blockx and blocky settings) on the frame.
Minimum value is 0 and maximum is "blocky x blockx" (at which point
no frames will ever be detected as combed). This setting is known
as MI in TFM/VFM vocabulary.

Default value is 80.

p/c/n/u/b meaning

p/c/n

We assume the following telecined stream:

Top fields: 1 2 2 3 4
Bottom fields: 1 2 3 4 4

The numbers correspond to the progressive frame the fields relate to.
Here, the first two frames are progressive, the 3rd and 4th are combed,
and so on.

When "fieldmatch" is configured to run a matching from bottom
(field=bottom) this is how this input stream get transformed:

Input stream:
T 1 2 2 3 4
B 1 2 3 4 4 <-- matching reference

Matches: c c n n c

Output stream:
T 1 2 3 4 4
B 1 2 3 4 4

As a result of the field matching, we can see that some frames get
duplicated. To perform a complete inverse telecine, you need to rely
on a decimation filter after this operation. See for instance the
decimate filter.

The same operation now matching from top fields (field=top) looks like
this:

Input stream:
T 1 2 2 3 4 <-- matching reference
B 1 2 3 4 4

Matches: c c p p c

Output stream:
T 1 2 2 3 4
B 1 2 2 3 4

In these examples, we can see what p, c and n mean; basically, they
refer to the frame and field of the opposite parity:

*<p matches the field of the opposite parity in the previous frame>
*<c matches the field of the opposite parity in the current frame>
*<n matches the field of the opposite parity in the next frame>

u/b

The u and b matching are a bit special in the sense that they match
from the opposite parity flag. In the following examples, we assume
that we are currently matching the 2nd frame (Top:2, bottom:2).
According to the match, a 'x' is placed above and below each matched
fields.

With bottom matching (field=bottom):

Match: c p n b u

x x x x x
Top 1 2 2 1 2 2 1 2 2 1 2 2 1 2 2
Bottom 1 2 3 1 2 3 1 2 3 1 2 3 1 2 3
x x x x x

Output frames:
2 1 2 2 2
2 2 2 1 3

With top matching (field=top):

Match: c p n b u

x x x x x
Top 1 2 2 1 2 2 1 2 2 1 2 2 1 2 2
Bottom 1 2 3 1 2 3 1 2 3 1 2 3 1 2 3
x x x x x

Output frames:
2 2 2 1 2
2 1 3 2 2

Examples

Simple IVTC of a top field first telecined stream:

fieldmatch=order=tff:combmatch=none, decimate

Advanced IVTC, with fallback on yadif for still combed frames:

fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced, decimate

fieldorder
Transform the field order of the input video.

It accepts the following parameters:

order
The output field order. Valid values are tff for top field first or
bff for bottom field first.

The default value is tff.

The transformation is done 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.

It 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

fifo
Buffer input images and send them when they are requested.

It is mainly useful when auto-inserted by the libavfilter framework.

It does not take parameters.

find_rect
Find a rectangular object

It accepts the following options:

object
Filepath of the object image, needs to be in gray8.

threshold
Detection threshold, default is 0.5.

mipmaps
Number of mipmaps, default is 3.

xmin, ymin, xmax, ymax
Specifies the rectangle in which to search.

Examples

· Generate a representative palette of a given video using ffmpeg:

ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv

cover_rect
Cover a rectangular object

It accepts the following options:

cover
Filepath of the optional cover image, needs to be in yuv420.

mode
Set covering mode.

It accepts the following values:

cover
cover it by the supplied image

blur
cover it by interpolating the surrounding pixels

Default value is blur.

Examples

· Generate a representative palette of a given video using ffmpeg:

ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv

format
Convert the input video to one of the specified pixel formats.
Libavfilter will try to pick one that is suitable as input to the next
filter.

It accepts the following parameters:

pix_fmts
A '|'-separated list of pixel format names, such as
"pix_fmts=yuv420p|monow|rgb24".

Examples

· Convert the input video to the yuv420p format

format=pix_fmts=yuv420p

Convert the input video to any of the formats in the list

format=pix_fmts=yuv420p|yuv444p|yuv410p

fps
Convert the video to specified constant frame rate by duplicating or
dropping frames as necessary.

It accepts the following parameters:

fps The desired output frame rate. The default is 25.

round
Rounding method.

Possible values are:

zero
zero round towards 0

inf round away from 0

down
round towards -infinity

up round towards +infinity

near
round to nearest

The default is "near".

start_time
Assume the first PTS should be the given value, in seconds. This
allows for padding/trimming at the start of stream. By default, no
assumption is made about the first frame's expected PTS, so no
padding or trimming is done. For example, this could be set to 0
to pad the beginning with duplicates of the first frame if a video
stream starts after the audio stream or to trim any frames with a
negative PTS.

Alternatively, the options can be specified as a flat string:
fps[:round].

See also the setpts filter.

Examples

· A typical usage in order to set the fps to 25:

fps=fps=25

· Sets the fps to 24, using abbreviation and rounding method to round
to nearest:

fps=fps=film:round=near

framepack
Pack two different video streams into a stereoscopic video, setting
proper metadata on supported codecs. The two views should have the same
size and framerate and processing will stop when the shorter video
ends. Please note that you may conveniently adjust view properties with
the scale and fps filters.

It accepts the following parameters:

format
The desired packing format. Supported values are:

sbs The views are next to each other (default).

tab The views are on top of each other.

lines
The views are packed by line.

columns
The views are packed by column.

frameseq
The views are temporally interleaved.

Some examples:

# Convert left and right views into a frame-sequential video
ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT

# Convert views into a side-by-side video with the same output resolution as the input
ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT

framerate
Change the frame rate by interpolating new video output frames from the
source frames.

This filter is not designed to function correctly with interlaced
media. If you wish to change the frame rate of interlaced media then
you are required to deinterlace before this filter and re-interlace
after this filter.

A description of the accepted options follows.

fps Specify the output frames per second. This option can also be
specified as a value alone. The default is 50.

interp_start
Specify the start of a range where the output frame will be created
as a linear interpolation of two frames. The range is [0-255], the
default is 15.

interp_end
Specify the end of a range where the output frame will be created
as a linear interpolation of two frames. The range is [0-255], the
default is 240.

scene
Specify the level at which a scene change is detected as a value
between 0 and 100 to indicate a new scene; a low value reflects a
low probability for the current frame to introduce a new scene,
while a higher value means the current frame is more likely to be
one. The default is 7.

flags
Specify flags influencing the filter process.

Available value for flags is:

scene_change_detect, scd
Enable scene change detection using the value of the option
scene. This flag is enabled by default.

framestep
Select one frame every N-th frame.

This filter accepts the following option:

step
Select frame after every "step" frames. Allowed values are
positive integers higher than 0. Default value is 1.

frei0r
Apply a frei0r effect to the input video.

To enable the compilation of this filter, you need to install the
frei0r header and configure FFmpeg with "--enable-frei0r".

It accepts the following parameters:

filter_name
The name of the frei0r effect to load. If the environment variable
FREI0R_PATH is defined, the frei0r effect is searched for in each
of the directories specified by the colon-separated list in
FREIOR_PATH. Otherwise, the standard frei0r paths are searched, in
this order: HOME/.frei0r-1/lib/, /usr/local/lib/frei0r-1/,
/usr/lib/frei0r-1/.

filter_params
A '|'-separated list of parameters to pass to the frei0r effect.

A frei0r effect parameter can be a boolean (its value is either "y" or
"n"), a double, a color (specified as R/G/B, where R, G, and B are
floating point numbers between 0.0 and 1.0, inclusive) or by a color
description specified in the "Color" section in the ffmpeg-utils
manual), a position (specified as X/Y, where X and Y are floating point
numbers) and/or a string.

The number and types of parameters depend on the loaded effect. If an
effect parameter is not specified, the default value is set.

Examples

· Apply the distort0r effect, setting the first two double
parameters:

frei0r=filter_name=distort0r:filter_params=0.5|0.01

· Apply the colordistance effect, taking a color as the first
parameter:

frei0r=colordistance:0.2/0.3/0.4
frei0r=colordistance:violet
frei0r=colordistance:0x112233

· Apply the perspective effect, specifying the top left and top right
image positions:

frei0r=perspective:0.2/0.2|0.8/0.2

For more information, see <http://frei0r.dyne.org>

fspp
Apply fast and simple postprocessing. It is a faster version of spp.

It splits (I)DCT into horizontal/vertical passes. Unlike the simple
post- processing filter, one of them is performed once per block, not
per pixel. This allows for much higher speed.

The filter accepts the following options:

quality
Set quality. This option defines the number of levels for
averaging. It accepts an integer in the range 4-5. Default value is
4.

qp Force a constant quantization parameter. It accepts an integer in
range 0-63. If not set, the filter will use the QP from the video
stream (if available).

strength
Set filter strength. It accepts an integer in range -15 to 32.
Lower values mean more details but also more artifacts, while
higher values make the image smoother but also blurrier. Default
value is 0 X PSNR optimal.

use_bframe_qp
Enable the use of the QP from the B-Frames if set to 1. Using this
option may cause flicker since the B-Frames have often larger QP.
Default is 0 (not enabled).

geq
The filter accepts the following options:

lum_expr, lum
Set the luminance expression.

cb_expr, cb
Set the chrominance blue expression.

cr_expr, cr
Set the chrominance red expression.

alpha_expr, a
Set the alpha expression.

red_expr, r
Set the red expression.

green_expr, g
Set the green expression.

blue_expr, b
Set the blue expression.

The colorspace is selected according to the specified options. If one
of the lum_expr, cb_expr, or cr_expr options is specified, the filter
will automatically select a YCbCr colorspace. If one of the red_expr,
green_expr, or blue_expr options is specified, it will select an RGB
colorspace.

If one of the chrominance expression is not defined, it falls back on
the other one. If no alpha expression is specified it will evaluate to
opaque value. If none of chrominance expressions are specified, they
will evaluate to the luminance expression.

The expressions can use the following variables and functions:

N The sequential number of the filtered frame, starting from 0.

X
Y The coordinates of the current sample.

W
H The width and height of the image.

SW
SH Width and height scale depending on the currently filtered plane.
It is the ratio between the corresponding luma plane number of
pixels and the current plane ones. E.g. for YUV4:2:0 the values are
"1,1" for the luma plane, and "0.5,0.5" for chroma planes.

T Time of the current frame, expressed in seconds.

p(x, y)
Return the value of the pixel at location (x,y) of the current
plane.

lum(x, y)
Return the value of the pixel at location (x,y) of the luminance
plane.

cb(x, y)
Return the value of the pixel at location (x,y) of the blue-
difference chroma plane. Return 0 if there is no such plane.

cr(x, y)
Return the value of the pixel at location (x,y) of the red-
difference chroma plane. Return 0 if there is no such plane.

r(x, y)
g(x, y)
b(x, y)
Return the value of the pixel at location (x,y) of the
red/green/blue component. Return 0 if there is no such component.

alpha(x, y)
Return the value of the pixel at location (x,y) of the alpha plane.
Return 0 if there is no such plane.

For functions, if x and y are outside the area, the value will be
automatically clipped to the closer edge.

Examples

· Flip the image horizontally:

geq=p(W-X\,Y)

· Generate a bidimensional sine wave, with angle "PI/3" and a
wavelength of 100 pixels:

geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128

· Generate a fancy enigmatic moving light:

nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128

· Generate a quick emboss effect:

format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2'

· Modify RGB components depending on pixel position:

geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)'

· Create a radial gradient that is the same size as the input (also
see the vignette filter):

geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray

· Create a linear gradient to use as a mask for another filter, then
compose with overlay. In this example the video will gradually
become more blurry from the top to the bottom of the y-axis as
defined by the linear gradient:

ffmpeg -i input.mp4 -filter_complex "geq=lum=255*(Y/H),format=gray[grad];[0:v]boxblur=4[blur];[blur][grad]alphamerge[alpha];[0:v][alpha]overlay" output.mp4

gradfun
Fix the banding artifacts that are sometimes introduced into nearly
flat regions by truncation to 8bit color depth. Interpolate the
gradients that should go where the bands are, and dither them.

It 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.

It accepts the following parameters:

strength
The maximum amount by which the filter will change any one pixel.
This is also the threshold for detecting nearly flat regions.
Acceptable values range from .51 to 64; the default value is 1.2.
Out-of-range values will be clipped to the valid range.

radius
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; the
default value is 16. Out-of-range values will be clipped to the
valid range.

Alternatively, the options can be specified as a flat string:
strength[:radius]

Examples

· Apply the filter with a 3.5 strength and radius of 8:

gradfun=3.5:8

· Specify radius, omitting the strength (which will fall-back to the
default value):

gradfun=radius=8

haldclut
Apply a Hald CLUT to a video stream.

First input is the video stream to process, and second one is the Hald
CLUT. The Hald CLUT input can be a simple picture or a complete video
stream.

The filter accepts the following options:

shortest
Force termination when the shortest input terminates. Default is 0.

repeatlast
Continue applying the last CLUT after the end of the stream. A
value of 0 disable the filter after the last frame of the CLUT is
reached. Default is 1.

"haldclut" also has the same interpolation options as lut3d (both
filters share the same internals).

More information about the Hald CLUT can be found on Eskil Steenberg's
website (Hald CLUT author) at
<http://www.quelsolaar.com/technology/clut.html>.

Workflow examples

Hald CLUT video stream

Generate an identity Hald CLUT stream altered with various effects:

ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut

Note: make sure you use a lossless codec.

Then use it with "haldclut" to apply it on some random stream:

ffmpeg -f lavfi -i mandelbrot -i clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv

The Hald CLUT will be applied to the 10 first seconds (duration of
clut.nut), then the latest picture of that CLUT stream will be applied
to the remaining frames of the "mandelbrot" stream.

Hald CLUT with preview

A Hald CLUT is supposed to be a squared image of "Level*Level*Level" by
"Level*Level*Level" pixels. For a given Hald CLUT, FFmpeg will select
the biggest possible square starting at the top left of the picture.
The remaining padding pixels (bottom or right) will be ignored. This
area can be used to add a preview of the Hald CLUT.

Typically, the following generated Hald CLUT will be supported by the
"haldclut" filter:

ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "
pad=iw+320 [padded_clut];
smptebars=s=320x256, split [a][b];
[padded_clut][a] overlay=W-320:h, curves=color_negative [main];
[main][b] overlay=W-320" -frames:v 1 clut.png

It contains the original and a preview of the effect of the CLUT: SMPTE
color bars are displayed on the right-top, and below the same color
bars processed by the color changes.

Then, the effect of this Hald CLUT can be visualized with:

ffplay input.mkv -vf "movie=clut.png, [in] haldclut"

hflip
Flip the input video horizontally.

For example, to horizontally flip the input video with ffmpeg:

ffmpeg -i in.avi -vf "hflip" out.avi

histeq
This filter applies a global color histogram equalization on a per-
frame basis.

It can be used to correct video that has a compressed range of pixel
intensities. The filter redistributes the pixel intensities to
equalize their distribution across the intensity range. It may be
viewed as an "automatically adjusting contrast filter". This filter is
useful only for correcting degraded or poorly captured source video.

The filter accepts the following options:

strength
Determine the amount of equalization to be applied. As the
strength is reduced, the distribution of pixel intensities more-
and-more approaches that of the input frame. The value must be a
float number in the range [0,1] and defaults to 0.200.

intensity
Set the maximum intensity that can generated and scale the output
values appropriately. The strength should be set as desired and
then the intensity can be limited if needed to avoid washing-out.
The value must be a float number in the range [0,1] and defaults to
0.210.

antibanding
Set the antibanding level. If enabled the filter will randomly vary
the luminance of output pixels by a small amount to avoid banding
of the histogram. Possible values are "none", "weak" or "strong".
It defaults to "none".

histogram
Compute and draw a color distribution histogram for the input video.

The computed histogram is a representation of the color component
distribution in an image.

The filter accepts the following options:

mode
Set histogram mode.

It accepts the following values:

levels
Standard histogram that displays the color components
distribution in an image. Displays color graph for each color
component. Shows distribution of the Y, U, V, A or R, G, B
components, depending on input format, in the current frame.
Below each graph a color component scale meter is shown.

color
Displays chroma values (U/V color placement) in a two
dimensional graph (which is called a vectorscope). The brighter
a pixel in the vectorscope, the more pixels of the input frame
correspond to that pixel (i.e., more pixels have this chroma
value). The V component is displayed on the horizontal (X)
axis, with the leftmost side being V = 0 and the rightmost side
being V = 255. The U component is displayed on the vertical (Y)
axis, with the top representing U = 0 and the bottom
representing U = 255.

The position of a white pixel in the graph corresponds to the
chroma value of a pixel of the input clip. The graph can
therefore be used to read the hue (color flavor) and the
saturation (the dominance of the hue in the color). As the hue
of a color changes, it moves around the square. At the center
of the square the saturation is zero, which means that the
corresponding pixel has no color. If the amount of a specific
color is increased (while leaving the other colors unchanged)
the saturation increases, and the indicator moves towards the
edge of the square.

color2
Chroma values in vectorscope, similar as "color" but actual
chroma values are displayed.

waveform
Per row/column color component graph. In row mode, the graph on
the left side represents color component value 0 and the right
side represents value = 255. In column mode, the top side
represents color component value = 0 and bottom side represents
value = 255.

Default value is "levels".

level_height
Set height of level in "levels". Default value is 200. Allowed
range is [50, 2048].

scale_height
Set height of color scale in "levels". Default value is 12.
Allowed range is [0, 40].

step
Set step for "waveform" mode. Smaller values are useful to find out
how many values of the same luminance are distributed across input
rows/columns. Default value is 10. Allowed range is [1, 255].

waveform_mode
Set mode for "waveform". Can be either "row", or "column". Default
is "row".

waveform_mirror
Set mirroring mode for "waveform". 0 means unmirrored, 1 means
mirrored. In mirrored mode, higher values will be represented on
the left side for "row" mode and at the top for "column" mode.
Default is 0 (unmirrored).

display_mode
Set display mode for "waveform" and "levels". It accepts the
following values:

parade
Display separate graph for the color components side by side in
"row" waveform mode or one below the other in "column" waveform
mode for "waveform" histogram mode. For "levels" histogram
mode, per color component graphs are placed below each other.

Using this display mode in "waveform" histogram mode makes it
easy to spot color casts in the highlights and shadows of an
image, by comparing the contours of the top and the bottom
graphs of each waveform. Since whites, grays, and blacks are
characterized by exactly equal amounts of red, green, and blue,
neutral areas of the picture should display three waveforms of
roughly equal width/height. If not, the correction is easy to
perform by making level adjustments the three waveforms.

overlay
Presents information identical to that in the "parade", except
that the graphs representing color components are superimposed
directly over one another.

This display mode in "waveform" histogram mode makes it easier
to spot relative differences or similarities in overlapping
areas of the color components that are supposed to be
identical, such as neutral whites, grays, or blacks.

Default is "parade".

levels_mode
Set mode for "levels". Can be either "linear", or "logarithmic".
Default is "linear".

components
Set what color components to display for mode "levels". Default is
7.

Examples

· Calculate and draw histogram:

ffplay -i input -vf histogram

hqdn3d
This is a high precision/quality 3d denoise filter. It 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
A non-negative floating point number which specifies spatial luma
strength. It defaults to 4.0.

chroma_spatial
A non-negative floating point number which specifies spatial chroma
strength. It defaults to 3.0*luma_spatial/4.0.

luma_tmp
A floating point number which specifies luma temporal strength. It
defaults to 6.0*luma_spatial/4.0.

chroma_tmp
A floating point number which specifies chroma temporal strength.
It defaults to luma_tmp*chroma_spatial/luma_spatial.

hqx
Apply a high-quality magnification filter designed for pixel art. This
filter was originally created by Maxim Stepin.

It accepts the following option:

n Set the scaling dimension: 2 for "hq2x", 3 for "hq3x" and 4 for
"hq4x". Default is 3.

hstack
Stack input videos horizontally.

All streams must be of same pixel format and of same height.

Note that this filter is faster than using overlay and pad filter to
create same output.

The filter accept the following option:

nb_inputs
Set number of input streams. Default is 2.

hue
Modify the hue and/or the saturation of the input.

It accepts the following parameters:

h Specify the hue angle as a number of degrees. It accepts an
expression, and defaults to "0".

s Specify the saturation in the [-10,10] range. It accepts an
expression and defaults to "1".

H Specify the hue angle as a number of radians. It accepts an
expression, and defaults to "0".

b Specify the brightness in the [-10,10] range. It accepts an
expression and defaults to "0".

h and H are mutually exclusive, and can't be specified at the same
time.

The b, h, H and s option values are expressions containing the
following constants:

n frame count of the input frame starting from 0

pts presentation timestamp of the input frame expressed in time base
units

r frame rate of the input video, NAN if the input frame rate is
unknown

t timestamp expressed in seconds, NAN if the input timestamp is
unknown

tb time base of the input video

Examples

· Set the hue to 90 degrees and the saturation to 1.0:

hue=h=90:s=1

· Same command but expressing the hue in radians:

hue=H=PI/2:s=1

· Rotate hue and make the saturation swing between 0 and 2 over a
period of 1 second:

hue="H=2*PI*t: s=sin(2*PI*t)+1"

· Apply a 3 seconds saturation fade-in effect starting at 0:

hue="s=min(t/3\,1)"

The general fade-in expression can be written as:

hue="s=min(0\, max((t-START)/DURATION\, 1))"

· Apply a 3 seconds saturation fade-out effect starting at 5 seconds:

hue="s=max(0\, min(1\, (8-t)/3))"

The general fade-out expression can be written as:

hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))"

Commands

This filter supports the following commands:

b
s
h
H Modify the hue and/or the saturation and/or brightness of the input
video. The command accepts the same syntax of the corresponding
option.

If the specified expression is not valid, it is kept at its current
value.

idet
Detect video interlacing type.

This filter tries to detect if the input frames as interlaced,
progressive, top or bottom field first. It will also try and detect
fields that are repeated between adjacent frames (a sign of telecine).

Single frame detection considers only immediately adjacent frames when
classifying each frame. Multiple frame detection incorporates the
classification history of previous frames.

The filter will log these metadata values:

single.current_frame
Detected type of current frame using single-frame detection. One
of: ``tff'' (top field first), ``bff'' (bottom field first),
``progressive'', or ``undetermined''

single.tff
Cumulative number of frames detected as top field first using
single-frame detection.

multiple.tff
Cumulative number of frames detected as top field first using
multiple-frame detection.

single.bff
Cumulative number of frames detected as bottom field first using
single-frame detection.

multiple.current_frame
Detected type of current frame using multiple-frame detection. One
of: ``tff'' (top field first), ``bff'' (bottom field first),
``progressive'', or ``undetermined''

multiple.bff
Cumulative number of frames detected as bottom field first using
multiple-frame detection.

single.progressive
Cumulative number of frames detected as progressive using single-
frame detection.

multiple.progressive
Cumulative number of frames detected as progressive using multiple-
frame detection.

single.undetermined
Cumulative number of frames that could not be classified using
single-frame detection.

multiple.undetermined
Cumulative number of frames that could not be classified using
multiple-frame detection.

repeated.current_frame
Which field in the current frame is repeated from the last. One of
``neither'', ``top'', or ``bottom''.

repeated.neither
Cumulative number of frames with no repeated field.

repeated.top
Cumulative number of frames with the top field repeated from the
previous frame's top field.

repeated.bottom
Cumulative number of frames with the bottom field repeated from the
previous frame's bottom field.

The filter accepts the following options:

intl_thres
Set interlacing threshold.

prog_thres
Set progressive threshold.

repeat_thres
Threshold for repeated field detection.

half_life
Number of frames after which a given frame's contribution to the
statistics is halved (i.e., it contributes only 0.5 to it's
classification). The default of 0 means that all frames seen are
given full weight of 1.0 forever.

analyze_interlaced_flag
When this is not 0 then idet will use the specified number of
frames to determine if the interlaced flag is accurate, it will not
count undetermined frames. If the flag is found to be accurate it
will be used without any further computations, if it is found to be
inaccurate it will be cleared without any further computations.
This allows inserting the idet filter as a low computational method
to clean up the interlaced flag

il
Deinterleave or interleave fields.

This filter allows one to process interlaced images fields without
deinterlacing them. Deinterleaving splits the input frame into 2 fields
(so called half pictures). Odd lines are moved to the top half of the
output image, even lines to the bottom half. You can process (filter)
them independently and then re-interleave them.

The filter accepts the following options:

luma_mode, l
chroma_mode, c
alpha_mode, a
Available values for luma_mode, chroma_mode and alpha_mode are:

none
Do nothing.

deinterleave, d
Deinterleave fields, placing one above the other.

interleave, i
Interleave fields. Reverse the effect of deinterleaving.

Default value is "none".

luma_swap, ls
chroma_swap, cs
alpha_swap, as
Swap luma/chroma/alpha fields. Exchange even & odd lines. Default
value is 0.

inflate
Apply inflate effect to the video.

This filter replaces the pixel by the local(3x3) average by taking into
account only values higher than the pixel.

It accepts the following options:

threshold0
threshold1
threshold2
threshold3
Limit the maximum change for each plane, default is 65535. If 0,
plane will remain unchanged.

interlace
Simple interlacing filter from progressive contents. This interleaves
upper (or lower) lines from odd frames with lower (or upper) lines from
even frames, halving the frame rate and preserving image height.

Original Original New Frame
Frame 'j' Frame 'j+1' (tff)
========== =========== ==================
Line 0 --------------------> Frame 'j' Line 0
Line 1 Line 1 ----> Frame 'j+1' Line 1
Line 2 ---------------------> Frame 'j' Line 2
Line 3 Line 3 ----> Frame 'j+1' Line 3
... ... ...
New Frame + 1 will be generated by Frame 'j+2' and Frame 'j+3' and so on

It accepts the following optional parameters:

scan
This determines whether the interlaced frame is taken from the even
(tff - default) or odd (bff) lines of the progressive frame.

lowpass
Enable (default) or disable the vertical lowpass filter to avoid
twitter interlacing and reduce moire patterns.

kerndeint
Deinterlace input video by applying Donald Graft's adaptive kernel
deinterling. Work on interlaced parts of a video to produce progressive
frames.

The description of the accepted parameters follows.

thresh
Set the threshold which affects the filter's tolerance when
determining if a pixel line must be processed. It must be an
integer in the range [0,255] and defaults to 10. A value of 0 will
result in applying the process on every pixels.

map Paint pixels exceeding the threshold value to white if set to 1.
Default is 0.

order
Set the fields order. Swap fields if set to 1, leave fields alone
if 0. Default is 0.

sharp
Enable additional sharpening if set to 1. Default is 0.

twoway
Enable twoway sharpening if set to 1. Default is 0.

Examples

· Apply default values:

kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0

· Enable additional sharpening:

kerndeint=sharp=1

· Paint processed pixels in white:

kerndeint=map=1

lenscorrection
Correct radial lens distortion

This filter can be used to correct for radial distortion as can result
from the use of wide angle lenses, and thereby re-rectify the image. To
find the right parameters one can use tools available for example as
part of opencv or simply trial-and-error. To use opencv use the
calibration sample (under samples/cpp) from the opencv sources and
extract the k1 and k2 coefficients from the resulting matrix.

Note that effectively the same filter is available in the open-source
tools Krita and Digikam from the KDE project.

In contrast to the vignette filter, which can also be used to
compensate lens errors, this filter corrects the distortion of the
image, whereas vignette corrects the brightness distribution, so you
may want to use both filters together in certain cases, though you will
have to take care of ordering, i.e. whether vignetting should be
applied before or after lens correction.

Options

The filter accepts the following options:

cx Relative x-coordinate of the focal point of the image, and thereby
the center of the distortion. This value has a range [0,1] and is
expressed as fractions of the image width.

cy Relative y-coordinate of the focal point of the image, and thereby
the center of the distortion. This value has a range [0,1] and is
expressed as fractions of the image height.

k1 Coefficient of the quadratic correction term. 0.5 means no
correction.

k2 Coefficient of the double quadratic correction term. 0.5 means no
correction.

The formula that generates the correction is:

r_src = r_tgt * (1 + k1 * (r_tgt / r_0)^2 + k2 * (r_tgt / r_0)^4)

where r_0 is halve of the image diagonal and r_src and r_tgt are the
distances from the focal point in the source and target images,
respectively.

lut3d
Apply a 3D LUT to an input video.

The filter accepts the following options:

file
Set the 3D LUT file name.

Currently supported formats:

3dl AfterEffects

cube
Iridas

dat DaVinci

m3d Pandora

interp
Select interpolation mode.

Available values are:

nearest
Use values from the nearest defined point.

trilinear
Interpolate values using the 8 points defining a cube.

tetrahedral
Interpolate values using a tetrahedron.

lut, lutrgb, lutyuv
Compute a look-up table for binding each pixel component input value to
an output value, and apply it to the input video.

lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB
input video.

These filters accept the following parameters:

c0 set first pixel component expression

c1 set second pixel component expression

c2 set third pixel component expression

c3 set fourth pixel component expression, corresponds to the alpha
component

r set red component expression

g set green component expression

b set blue component expression

a alpha component expression

y set Y/luminance component expression

u set U/Cb component expression

v set V/Cr component expression

Each of them specifies the expression to use for computing the lookup
table for the corresponding pixel component values.

The exact component associated to each of the c* options depends on the
format in input.

The lut filter requires either YUV or RGB pixel formats in input,
lutrgb requires RGB pixel formats in input, and lutyuv requires YUV.

The expressions can contain the following constants and functions:

w
h The input width and height.

val The input value for the pixel component.

clipval
The input value, clipped to the minval-maxval range.

maxval
The maximum value for the pixel component.

minval
The minimum value for the pixel component.

negval
The negated value for the pixel component value, clipped to the
minval-maxval range; it corresponds to the expression
"maxval-clipval+minval".

clip(val)
The computed value in val, clipped to the minval-maxval range.

gammaval(gamma)
The computed gamma correction value of the pixel component value,
clipped to the minval-maxval range. It corresponds to the
expression
"pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"

All expressions default to "val".

Examples

· Negate input video:

lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val"
lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val"

The above is the same as:

lutrgb="r=negval:g=negval:b=negval"
lutyuv="y=negval:u=negval:v=negval"

· Negate luminance:

lutyuv=y=negval

· Remove chroma components, turning the video into a graytone image:

lutyuv="u=128:v=128"

· Apply a luma burning effect:

lutyuv="y=2*val"

· Remove green and blue components:

lutrgb="g=0:b=0"

· Set a constant alpha channel value on input:

format=rgba,lutrgb=a="maxval-minval/2"

· Correct luminance gamma by a factor of 0.5:

lutyuv=y=gammaval(0.5)

· Discard least significant bits of luma:

lutyuv=y='bitand(val, 128+64+32)'

mergeplanes
Merge color channel components from several video streams.

The filter accepts up to 4 input streams, and merge selected input
planes to the output video.

This filter accepts the following options:

mapping
Set input to output plane mapping. Default is 0.

The mappings is specified as a bitmap. It should be specified as a
hexadecimal number in the form 0xAa[Bb[Cc[Dd]]]. 'Aa' describes the
mapping for the first plane of the output stream. 'A' sets the
number of the input stream to use (from 0 to 3), and 'a' the plane
number of the corresponding input to use (from 0 to 3). The rest of
the mappings is similar, 'Bb' describes the mapping for the output
stream second plane, 'Cc' describes the mapping for the output
stream third plane and 'Dd' describes the mapping for the output
stream fourth plane.

format
Set output pixel format. Default is "yuva444p".

Examples

· Merge three gray video streams of same width and height into single
video stream:

[a0][a1][a2]mergeplanes=0x001020:yuv444p

· Merge 1st yuv444p stream and 2nd gray video stream into yuva444p
video stream:

[a0][a1]mergeplanes=0x00010210:yuva444p

· Swap Y and A plane in yuva444p stream:

format=yuva444p,mergeplanes=0x03010200:yuva444p

· Swap U and V plane in yuv420p stream:

format=yuv420p,mergeplanes=0x000201:yuv420p

· Cast a rgb24 clip to yuv444p:

format=rgb24,mergeplanes=0x000102:yuv444p

mcdeint
Apply motion-compensation deinterlacing.

It needs one field per frame as input and must thus be used together
with yadif=1/3 or equivalent.

This filter accepts the following options:

mode
Set the deinterlacing mode.

It accepts one of the following values:

fast
medium
slow
use iterative motion estimation

extra_slow
like slow, but use multiple reference frames.

Default value is fast.

parity
Set the picture field parity assumed for the input video. It must
be one of the following values:

0, tff
assume top field first

1, bff
assume bottom field first

Default value is bff.

qp Set per-block quantization parameter (QP) used by the internal
encoder.

Higher values should result in a smoother motion vector field but
less optimal individual vectors. Default value is 1.

mpdecimate
Drop frames that do not differ greatly from the previous frame in order
to reduce frame rate.

The main use of this filter is for very-low-bitrate encoding (e.g.
streaming over dialup modem), but it could in theory be used for fixing
movies that were inverse-telecined incorrectly.

A description of the accepted options follows.

max Set the maximum number of consecutive frames which can be dropped
(if positive), or the minimum interval between dropped frames (if
negative). If the value is 0, the frame is dropped unregarding the
number of previous sequentially dropped frames.

Default value is 0.

hi
lo
frac
Set the dropping threshold values.

Values for hi and lo are for 8x8 pixel blocks and represent actual
pixel value differences, so a threshold of 64 corresponds to 1 unit
of difference for each pixel, or the same spread out differently
over the block.

A frame is a candidate for dropping if no 8x8 blocks differ by more
than a threshold of hi, and if no more than frac blocks (1 meaning
the whole image) differ by more than a threshold of lo.

Default value for hi is 64*12, default value for lo is 64*5, and
default value for frac is 0.33.

negate
Negate input video.

It accepts an integer in input; if non-zero it negates the alpha
component (if available). The default value in input is 0.

noformat
Force libavfilter not to use any of the specified pixel formats for the
input to the next filter.

It accepts the following parameters:

pix_fmts
A '|'-separated list of pixel format names, such as
apix_fmts=yuv420p|monow|rgb24".

Examples

· Force libavfilter to use a format different from yuv420p for the
input to the vflip filter:

noformat=pix_fmts=yuv420p,vflip

· Convert the input video to any of the formats not contained in the
list:

noformat=yuv420p|yuv444p|yuv410p

noise
Add noise on video input frame.

The filter accepts the following options:

all_seed
c0_seed
c1_seed
c2_seed
c3_seed
Set noise seed for specific pixel component or all pixel components
in case of all_seed. Default value is 123457.

all_strength, alls
c0_strength, c0s
c1_strength, c1s
c2_strength, c2s
c3_strength, c3s
Set noise strength for specific pixel component or all pixel
components in case all_strength. Default value is 0. Allowed range
is [0, 100].

all_flags, allf
c0_flags, c0f
c1_flags, c1f
c2_flags, c2f
c3_flags, c3f
Set pixel component flags or set flags for all components if
all_flags. Available values for component flags are:

a averaged temporal noise (smoother)

p mix random noise with a (semi)regular pattern

t temporal noise (noise pattern changes between frames)

u uniform noise (gaussian otherwise)

Examples

Add temporal and uniform noise to input video:

noise=alls=20:allf=t+u

null
Pass the video source unchanged to the output.

ocv
Apply a video transform using libopencv.

To enable this filter, install the libopencv library and headers and
configure FFmpeg with "--enable-libopencv".

It accepts the following parameters:

filter_name
The name of the libopencv filter to apply.

filter_params
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
information:
<http://docs.opencv.org/master/modules/imgproc/doc/filtering.html>

Several libopencv filters are supported; see the following subsections.

dilate

Dilate an image by using a specific structuring element. It
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 columns and rows of the
structuring element, anchor_x and anchor_y the anchor point, and shape
the shape for the structuring element. shape must be "rect", "cross",
"ellipse", or "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.

Some examples:

# Use the default values
ocv=dilate

# Dilate using a structuring element with a 5x5 cross, iterating two times
ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2

# Read the shape from the file diamond.shape, iterating two times.
# The file diamond.shape may contain a pattern of characters like this
# *
# ***
# *****
# ***
# *
# The specified columns and rows are ignored
# but the anchor point coordinates are not
ocv=dilate:0x0+2x2/custom=diamond.shape|2

erode

Erode an image by using a specific structuring element. It corresponds
to the libopencv function "cvErode".

It accepts the parameters: struct_el:nb_iterations, with the same
syntax and semantics as the dilate filter.

smooth

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 must be one of the
following values: "blur", "blur_no_scale", "median", "gaussian", or
"bilateral". The default value is "gaussian".

The meaning of param1, param2, param3, and param4 depend on the smooth
type. param1 and param2 accept integer positive values or 0. param3 and
param4 accept floating point 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
Overlay one video on top of another.

It takes two inputs and has one output. The first input is the "main"
video on which the second input is overlaid.

It accepts the following parameters:

A description of the accepted options follows.

x
y Set the expression for the x and y coordinates of the overlaid
video on the main video. Default value is "0" for both expressions.
In case the expression is invalid, it is set to a huge value
(meaning that the overlay will not be displayed within the output
visible area).

eof_action
The action to take when EOF is encountered on the secondary input;
it accepts one of the following values:

repeat
Repeat the last frame (the default).

endall
End both streams.

pass
Pass the main input through.

eval
Set when the expressions for x, and y are evaluated.

It accepts the following values:

init
only evaluate expressions once during the filter initialization
or when a command is processed

frame
evaluate expressions for each incoming frame

Default value is frame.

shortest
If set to 1, force the output to terminate when the shortest input
terminates. Default value is 0.

format
Set the format for the output video.

It accepts the following values:

yuv420
force YUV420 output

yuv422
force YUV422 output

yuv444
force YUV444 output

rgb force RGB output

Default value is yuv420.

rgb (deprecated)
If set to 1, force the filter to accept inputs in the RGB color
space. Default value is 0. This option is deprecated, use format
instead.

repeatlast
If set to 1, force the filter to draw the last overlay frame over
the main input until the end of the stream. A value of 0 disables
this behavior. Default value is 1.

The x, and y expressions can contain the following parameters.

main_w, W
main_h, H
The main input width and height.

overlay_w, w
overlay_h, h
The overlay input width and height.

x
y The computed values for x and y. They are evaluated for each new
frame.

hsub
vsub
horizontal and vertical chroma subsample values of the output
format. For example for the pixel format "yuv422p" hsub is 2 and
vsub is 1.

n the number of input frame, starting from 0

pos the position in the file of the input frame, NAN if unknown

t The timestamp, expressed in seconds. It's NAN if the input
timestamp is unknown.

Note that the n, pos, t variables are available only when evaluation is
done per frame, and will evaluate to NAN when eval is set to init.

Be aware that frames are taken from each input video in timestamp
order, hence, if their initial timestamps differ, it is a good idea to
pass the two inputs through a setpts=PTS-STARTPTS filter to have them
begin in the same zero timestamp, as the example for the movie filter
does.

You can chain together more overlays but you should test the efficiency
of such approach.

Commands

This filter supports the following commands:

x
y Modify the x and y of the overlay input. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

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

Using named options the example above becomes:

overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10

· Insert a transparent PNG logo in the bottom left corner of the
input, using the ffmpeg tool with the "-filter_complex" option:

ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output

· Insert 2 different transparent PNG logos (second logo on bottom
right corner) using the ffmpeg tool:

ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output

· Add a transparent color layer on top of the main video; "WxH" must
specify the size of the main input to the overlay filter:

color=color=red@.3:size=WxH [over]; [in][over] overlay [out]

· Play an original video and a filtered version (here with the
deshake filter) side by side using the ffplay tool:

ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w'

The above command is the same as:

ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w'

· Make a sliding overlay appearing from the left to the right top
part of the screen starting since time 2:

overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0

· Compose output by putting two input videos side to side:

ffmpeg -i left.avi -i right.avi -filter_complex "
nullsrc=size=200x100 [background];
[0:v] setpts=PTS-STARTPTS, scale=100x100 [left];
[1:v] setpts=PTS-STARTPTS, scale=100x100 [right];
[background][left] overlay=shortest=1 [background+left];
[background+left][right] overlay=shortest=1:x=100 [left+right]
"

· Mask 10-20 seconds of a video by applying the delogo filter to a
section

ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k
-vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]'
masked.avi

· Chain several overlays in cascade:

nullsrc=s=200x200 [bg];
testsrc=s=100x100, split=4 [in0][in1][in2][in3];
[in0] lutrgb=r=0, [bg] overlay=0:0 [mid0];
[in1] lutrgb=g=0, [mid0] overlay=100:0 [mid1];
[in2] lutrgb=b=0, [mid1] overlay=0:100 [mid2];
[in3] null, [mid2] overlay=100:100 [out0]

owdenoise
Apply Overcomplete Wavelet denoiser.

The filter accepts the following options:

depth
Set depth.

Larger depth values will denoise lower frequency components more,
but slow down filtering.

Must be an int in the range 8-16, default is 8.

luma_strength, ls
Set luma strength.

Must be a double value in the range 0-1000, default is 1.0.

chroma_strength, cs
Set chroma strength.

Must be a double value in the range 0-1000, default is 1.0.

pad
Add paddings to the input image, and place the original input at the
provided x, y coordinates.

It accepts the following parameters:

width, w
height, h
Specify an expression for 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 vice versa.

The default value of width and height is 0.

x
y Specify the offsets to place the input image at within 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 vice versa.

The default value of x and y is 0.

color
Specify the color of the padded area. For the syntax of this
option, check the "Color" section in the ffmpeg-utils manual.

The default value of color is "black".

The value for the width, height, x, and y options are expressions
containing the following constants:

in_w
in_h
The input video width and height.

iw
ih These are the same as in_w and in_h.

out_w
out_h
The output width and height (the size of the padded area), as
specified by the width and height expressions.

ow
oh These are the same as out_w and out_h.

x
y The x and y offsets as specified by the x and y expressions, or NAN
if not yet specified.

a same as iw / ih

sar input sample aspect ratio

dar input display aspect ratio, it is the same as (iw / ih) * sar

hsub
vsub
The horizontal and vertical chroma subsample values. For example
for the pixel format "yuv422p" hsub is 2 and vsub is 1.

Examples

· Add paddings with the color "violet" to the input video. The output
video size is 640x480, and the top-left corner of the input video
is placed at column 0, row 40

pad=640:480:0:40:violet

The example above is equivalent to the following command:

pad=width=640:height=480:x=0:y=40:color=violet

· Pad the input to get an output with dimensions increased by 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"

· In case of anamorphic video, in order to set the output display
aspect correctly, it is necessary to use sar in the expression,
according to the relation:

(ih * X / ih) * sar = output_dar
X = output_dar / sar

Thus the previous example needs to be modified to:

pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2"

· Double the 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"

palettegen
Generate one palette for a whole video stream.

It accepts the following options:

max_colors
Set the maximum number of colors to quantize in the palette. Note:
the palette will still contain 256 colors; the unused palette
entries will be black.

reserve_transparent
Create a palette of 255 colors maximum and reserve the last one for
transparency. Reserving the transparency color is useful for GIF
optimization. If not set, the maximum of colors in the palette
will be 256. You probably want to disable this option for a
standalone image. Set by default.

stats_mode
Set statistics mode.

It accepts the following values:

full
Compute full frame histograms.

diff
Compute histograms only for the part that differs from previous
frame. This might be relevant to give more importance to the
moving part of your input if the background is static.

Default value is full.

The filter also exports the frame metadata "lavfi.color_quant_ratio"
("nb_color_in / nb_color_out") which you can use to evaluate the degree
of color quantization of the palette. This information is also visible
at info logging level.

Examples

· Generate a representative palette of a given video using ffmpeg:

ffmpeg -i input.mkv -vf palettegen palette.png

paletteuse
Use a palette to downsample an input video stream.

The filter takes two inputs: one video stream and a palette. The
palette must be a 256 pixels image.

It accepts the following options:

dither
Select dithering mode. Available algorithms are:

bayer
Ordered 8x8 bayer dithering (deterministic)

heckbert
Dithering as defined by Paul Heckbert in 1982 (simple error
diffusion). Note: this dithering is sometimes considered
"wrong" and is included as a reference.

floyd_steinberg
Floyd and Steingberg dithering (error diffusion)

sierra2
Frankie Sierra dithering v2 (error diffusion)

sierra2_4a
Frankie Sierra dithering v2 "Lite" (error diffusion)

Default is sierra2_4a.

bayer_scale
When bayer dithering is selected, this option defines the scale of
the pattern (how much the crosshatch pattern is visible). A low
value means more visible pattern for less banding, and higher value
means less visible pattern at the cost of more banding.

The option must be an integer value in the range [0,5]. Default is
2.

diff_mode
If set, define the zone to process

rectangle
Only the changing rectangle will be reprocessed. This is
similar to GIF cropping/offsetting compression mechanism. This
option can be useful for speed if only a part of the image is
changing, and has use cases such as limiting the scope of the
error diffusal dither to the rectangle that bounds the moving
scene (it leads to more deterministic output if the scene
doesn't change much, and as a result less moving noise and
better GIF compression).

Default is none.

Examples

· Use a palette (generated for example with palettegen) to encode a
GIF using ffmpeg:

ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif

perspective
Correct perspective of video not recorded perpendicular to the screen.

A description of the accepted parameters follows.

x0
y0
x1
y1
x2
y2
x3
y3 Set coordinates expression for top left, top right, bottom left and
bottom right corners. Default values are "0:0:W:0:0:H:W:H" with
which perspective will remain unchanged. If the "sense" option is
set to "source", then the specified points will be sent to the
corners of the destination. If the "sense" option is set to
"destination", then the corners of the source will be sent to the
specified coordinates.

The expressions can use the following variables:

W
H the width and height of video frame.

interpolation
Set interpolation for perspective correction.

It accepts the following values:

linear
cubic

Default value is linear.

sense
Set interpretation of coordinate options.

It accepts the following values:

0, source
Send point in the source specified by the given coordinates to
the corners of the destination.

1, destination
Send the corners of the source to the point in the destination
specified by the given coordinates.

Default value is source.

phase
Delay interlaced video by one field time so that the field order
changes.

The intended use is to fix PAL movies that have been captured with the
opposite field order to the film-to-video transfer.

A description of the accepted parameters follows.

mode
Set phase mode.

It accepts the following values:

t Capture field order top-first, transfer bottom-first. Filter
will delay the bottom field.

b Capture field order bottom-first, transfer top-first. Filter
will delay the top field.

p Capture and transfer with the same field order. This mode only
exists for the documentation of the other options to refer to,
but if you actually select it, the filter will faithfully do
nothing.

a Capture field order determined automatically by field flags,
transfer opposite. Filter selects among t and b modes on a
frame by frame basis using field flags. If no field information
is available, then this works just like u.

u Capture unknown or varying, transfer opposite. Filter selects
among t and b on a frame by frame basis by analyzing the images
and selecting the alternative that produces best match between
the fields.

T Capture top-first, transfer unknown or varying. Filter selects
among t and p using image analysis.

B Capture bottom-first, transfer unknown or varying. Filter
selects among b and p using image analysis.

A Capture determined by field flags, transfer unknown or varying.
Filter selects among t, b and p using field flags and image
analysis. If no field information is available, then this works
just like U. This is the default mode.

U Both capture and transfer unknown or varying. Filter selects
among t, b and p using image analysis only.

pixdesctest
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.

pp
Enable the specified chain of postprocessing subfilters using
libpostproc. This library should be automatically selected with a GPL
build ("--enable-gpl"). Subfilters must be separated by '/' and can be
disabled by prepending a '-'. Each subfilter and some options have a
short and a long name that can be used interchangeably, i.e. dr/dering
are the same.

The filters accept the following options:

subfilters
Set postprocessing subfilters string.

All subfilters share common options to determine their scope:

a/autoq
Honor the quality commands for this subfilter.

c/chrom
Do chrominance filtering, too (default).

y/nochrom
Do luminance filtering only (no chrominance).

n/noluma
Do chrominance filtering only (no luminance).

These options can be appended after the subfilter name, separated by a
'|'.

Available subfilters are:

hb/hdeblock[|difference[|flatness]]
Horizontal deblocking filter

difference
Difference factor where higher values mean more deblocking
(default: 32).

flatness
Flatness threshold where lower values mean more deblocking
(default: 39).

vb/vdeblock[|difference[|flatness]]
Vertical deblocking filter

difference
Difference factor where higher values mean more deblocking
(default: 32).

flatness
Flatness threshold where lower values mean more deblocking
(default: 39).

ha/hadeblock[|difference[|flatness]]
Accurate horizontal deblocking filter

difference
Difference factor where higher values mean more deblocking
(default: 32).

flatness
Flatness threshold where lower values mean more deblocking
(default: 39).

va/vadeblock[|difference[|flatness]]
Accurate vertical deblocking filter

difference
Difference factor where higher values mean more deblocking
(default: 32).

flatness
Flatness threshold where lower values mean more deblocking
(default: 39).

The horizontal and vertical deblocking filters share the difference and
flatness values so you cannot set different horizontal and vertical
thresholds.

h1/x1hdeblock
Experimental horizontal deblocking filter

v1/x1vdeblock
Experimental vertical deblocking filter

dr/dering
Deringing filter

tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise
reducer
threshold1
larger -> stronger filtering

threshold2
larger -> stronger filtering

threshold3
larger -> stronger filtering

al/autolevels[:f/fullyrange], automatic brightness / contrast
correction
f/fullyrange
Stretch luminance to "0-255".

lb/linblenddeint
Linear blend deinterlacing filter that deinterlaces the given block
by filtering all lines with a "(1 2 1)" filter.

li/linipoldeint
Linear interpolating deinterlacing filter that deinterlaces the
given block by linearly interpolating every second line.

ci/cubicipoldeint
Cubic interpolating deinterlacing filter deinterlaces the given
block by cubically interpolating every second line.

md/mediandeint
Median deinterlacing filter that deinterlaces the given block by
applying a median filter to every second line.

fd/ffmpegdeint
FFmpeg deinterlacing filter that deinterlaces the given block by
filtering every second line with a "(-1 4 2 4 -1)" filter.

l5/lowpass5
Vertically applied FIR lowpass deinterlacing filter that
deinterlaces the given block by filtering all lines with a "(-1 2 6
2 -1)" filter.

fq/forceQuant[|quantizer]
Overrides the quantizer table from the input with the constant
quantizer you specify.

quantizer
Quantizer to use

de/default
Default pp filter combination ("hb|a,vb|a,dr|a")

fa/fast
Fast pp filter combination ("h1|a,v1|a,dr|a")

ac High quality pp filter combination ("ha|a|128|7,va|a,dr|a")

Examples

· Apply horizontal and vertical deblocking, deringing and automatic
brightness/contrast:

pp=hb/vb/dr/al

· Apply default filters without brightness/contrast correction:

pp=de/-al

· Apply default filters and temporal denoiser:

pp=default/tmpnoise|1|2|3

· Apply deblocking on luminance only, and switch vertical deblocking
on or off automatically depending on available CPU time:

pp=hb|y/vb|a

pp7
Apply Postprocessing filter 7. It is variant of the spp filter, similar
to spp = 6 with 7 point DCT, where only the center sample is used after
IDCT.

The filter accepts the following options:

qp Force a constant quantization parameter. It accepts an integer in
range 0 to 63. If not set, the filter will use the QP from the
video stream (if available).

mode
Set thresholding mode. Available modes are:

hard
Set hard thresholding.

soft
Set soft thresholding (better de-ringing effect, but likely
blurrier).

medium
Set medium thresholding (good results, default).

psnr
Obtain the average, maximum and minimum PSNR (Peak Signal to Noise
Ratio) between two input videos.

This filter takes in input two input videos, the first input is
considered the "main" source and is passed unchanged to the output. The
second input is used as a "reference" video for computing the PSNR.

Both video inputs must have the same resolution and pixel format for
this filter to work correctly. Also it assumes that both inputs have
the same number of frames, which are compared one by one.

The obtained average PSNR is printed through the logging system.

The filter stores the accumulated MSE (mean squared error) of each
frame, and at the end of the processing it is averaged across all
frames equally, and the following formula is applied to obtain the
PSNR:

PSNR = 10*log10(MAX^2/MSE)

Where MAX is the average of the maximum values of each component of the
image.

The description of the accepted parameters follows.

stats_file, f
If specified the filter will use the named file to save the PSNR of
each individual frame.

The file printed if stats_file is selected, contains a sequence of
key/value pairs of the form key:value for each compared couple of
frames.

A description of each shown parameter follows:

n sequential number of the input frame, starting from 1

mse_avg
Mean Square Error pixel-by-pixel average difference of the compared
frames, averaged over all the image components.

mse_y, mse_u, mse_v, mse_r, mse_g, mse_g, mse_a
Mean Square Error pixel-by-pixel average difference of the compared
frames for the component specified by the suffix.

psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a
Peak Signal to Noise ratio of the compared frames for the component
specified by the suffix.

For example:

movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
[main][ref] psnr="stats_file=stats.log" [out]

On this example the input file being processed is compared with the
reference file ref_movie.mpg. The PSNR of each individual frame is
stored in stats.log.

pullup
Pulldown reversal (inverse telecine) filter, capable of handling mixed
hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps
progressive content.

The pullup filter is designed to take advantage of future context in
making its decisions. This filter is stateless in the sense that it
does not lock onto a pattern to follow, but it instead looks forward to
the following fields in order to identify matches and rebuild
progressive frames.

To produce content with an even framerate, insert the fps filter after
pullup, use "fps=24000/1001" if the input frame rate is 29.97fps,
"fps=24" for 30fps and the (rare) telecined 25fps input.

The filter accepts the following options:

jl
jr
jt
jb These options set the amount of "junk" to ignore at the left,
right, top, and bottom of the image, respectively. Left and right
are in units of 8 pixels, while top and bottom are in units of 2
lines. The default is 8 pixels on each side.

sb Set the strict breaks. Setting this option to 1 will reduce the
chances of filter generating an occasional mismatched frame, but it
may also cause an excessive number of frames to be dropped during
high motion sequences. Conversely, setting it to -1 will make
filter match fields more easily. This may help processing of video
where there is slight blurring between the fields, but may also
cause there to be interlaced frames in the output. Default value
is 0.

mp Set the metric plane to use. It accepts the following values:

l Use luma plane.

u Use chroma blue plane.

v Use chroma red plane.

This option may be set to use chroma plane instead of the default
luma plane for doing filter's computations. This may improve
accuracy on very clean source material, but more likely will
decrease accuracy, especially if there is chroma noise (rainbow
effect) or any grayscale video. The main purpose of setting mp to
a chroma plane is to reduce CPU load and make pullup usable in
realtime on slow machines.

For best results (without duplicated frames in the output file) it is
necessary to change the output frame rate. For example, to inverse
telecine NTSC input:

ffmpeg -i input -vf pullup -r 24000/1001 ...

qp
Change video quantization parameters (QP).

The filter accepts the following option:

qp Set expression for quantization parameter.

The expression is evaluated through the eval API and can contain, among
others, the following constants:

known
1 if index is not 129, 0 otherwise.

qp Sequentional index starting from -129 to 128.

Examples

· Some equation like:

qp=2+2*sin(PI*qp)

random
Flush video frames from internal cache of frames into a random order.
No frame is discarded. Inspired by frei0r nervous filter.

frames
Set size in number of frames of internal cache, in range from 2 to
512. Default is 30.

seed
Set seed for random number generator, must be an integer included
between 0 and "UINT32_MAX". If not specified, or if explicitly set
to less than 0, the filter will try to use a good random seed on a
best effort basis.

removegrain
The removegrain filter is a spatial denoiser for progressive video.

m0 Set mode for the first plane.

m1 Set mode for the second plane.

m2 Set mode for the third plane.

m3 Set mode for the fourth plane.

Range of mode is from 0 to 24. Description of each mode follows:

0 Leave input plane unchanged. Default.

1 Clips the pixel with the minimum and maximum of the 8 neighbour
pixels.

2 Clips the pixel with the second minimum and maximum of the 8
neighbour pixels.

3 Clips the pixel with the third minimum and maximum of the 8
neighbour pixels.

4 Clips the pixel with the fourth minimum and maximum of the 8
neighbour pixels. This is equivalent to a median filter.

5 Line-sensitive clipping giving the minimal change.

6 Line-sensitive clipping, intermediate.

7 Line-sensitive clipping, intermediate.

8 Line-sensitive clipping, intermediate.

9 Line-sensitive clipping on a line where the neighbours pixels are
the closest.

10 Replaces the target pixel with the closest neighbour.

11 [1 2 1] horizontal and vertical kernel blur.

12 Same as mode 11.

13 Bob mode, interpolates top field from the line where the neighbours
pixels are the closest.

14 Bob mode, interpolates bottom field from the line where the
neighbours pixels are the closest.

15 Bob mode, interpolates top field. Same as 13 but with a more
complicated interpolation formula.

16 Bob mode, interpolates bottom field. Same as 14 but with a more
complicated interpolation formula.

17 Clips the pixel with the minimum and maximum of respectively the
maximum and minimum of each pair of opposite neighbour pixels.

18 Line-sensitive clipping using opposite neighbours whose greatest
distance from the current pixel is minimal.

19 Replaces the pixel with the average of its 8 neighbours.

20 Averages the 9 pixels ([1 1 1] horizontal and vertical blur).

21 Clips pixels using the averages of opposite neighbour.

22 Same as mode 21 but simpler and faster.

23 Small edge and halo removal, but reputed useless.

24 Similar as 23.

removelogo
Suppress a TV station logo, using an image file to determine which
pixels comprise the logo. It works by filling in the pixels that
comprise the logo with neighboring pixels.

The filter accepts the following options:

filename, f
Set the filter bitmap file, which can be any image format supported
by libavformat. The width and height of the image file must match
those of the video stream being processed.

Pixels in the provided bitmap image with a value of zero are not
considered part of the logo, non-zero pixels are considered part of the
logo. If you use white (255) for the logo and black (0) for the rest,
you will be safe. For making the filter bitmap, it is recommended to
take a screen capture of a black frame with the logo visible, and then
using a threshold filter followed by the erode filter once or twice.

If needed, little splotches can be fixed manually. Remember that if
logo pixels are not covered, the filter quality will be much reduced.
Marking too many pixels as part of the logo does not hurt as much, but
it will increase the amount of blurring needed to cover over the image
and will destroy more information than necessary, and extra pixels will
slow things down on a large logo.

repeatfields
This filter uses the repeat_field flag from the Video ES headers and
hard repeats fields based on its value.

reverse, areverse
Reverse a clip.

Warning: This filter requires memory to buffer the entire clip, so
trimming is suggested.

Examples

· Take the first 5 seconds of a clip, and reverse it.

trim=end=5,reverse

rotate
Rotate video by an arbitrary angle expressed in radians.

The filter accepts the following options:

A description of the optional parameters follows.

angle, a
Set an expression for the angle by which to rotate the input video
clockwise, expressed as a number of radians. A negative value will
result in a counter-clockwise rotation. By default it is set to
"0".

This expression is evaluated for each frame.

out_w, ow
Set the output width expression, default value is "iw". This
expression is evaluated just once during configuration.

out_h, oh
Set the output height expression, default value is "ih". This
expression is evaluated just once during configuration.

bilinear
Enable bilinear interpolation if set to 1, a value of 0 disables
it. Default value is 1.

fillcolor, c
Set the color used to fill the output area not covered by the
rotated image. For the general syntax of this option, check the
"Color" section in the ffmpeg-utils manual. If the special value
"none" is selected then no background is printed (useful for
example if the background is never shown).

Default value is "black".

The expressions for the angle and the output size can contain the
following constants and functions:

n sequential number of the input frame, starting from 0. It is always
NAN before the first frame is filtered.

t time in seconds of the input frame, it is set to 0 when the filter
is configured. It is always NAN before the first frame is filtered.

hsub
vsub
horizontal and vertical chroma subsample values. For example for
the pixel format "yuv422p" hsub is 2 and vsub is 1.

in_w, iw
in_h, ih
the input video width and height

out_w, ow
out_h, oh
the output width and height, that is the size of the padded area as
specified by the width and height expressions

rotw(a)
roth(a)
the minimal width/height required for completely containing the
input video rotated by a radians.

These are only available when computing the out_w and out_h
expressions.

Examples

· Rotate the input by PI/6 radians clockwise:

rotate=PI/6

· Rotate the input by PI/6 radians counter-clockwise:

rotate=-PI/6

· Rotate the input by 45 degrees clockwise:

rotate=45*PI/180

· Apply a constant rotation with period T, starting from an angle of
PI/3:

rotate=PI/3+2*PI*t/T

· Make the input video rotation oscillating with a period of T
seconds and an amplitude of A radians:

rotate=A*sin(2*PI/T*t)

· Rotate the video, output size is chosen so that the whole rotating
input video is always completely contained in the output:

rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'

· Rotate the video, reduce the output size so that no background is
ever shown:

rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none

Commands

The filter supports the following commands:

a, angle
Set the angle expression. The command accepts the same syntax of
the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

sab
Apply Shape Adaptive Blur.

The filter accepts the following options:

luma_radius, lr
Set luma blur filter strength, must be a value in range 0.1-4.0,
default value is 1.0. A greater value will result in a more blurred
image, and in slower processing.

luma_pre_filter_radius, lpfr
Set luma pre-filter radius, must be a value in the 0.1-2.0 range,
default value is 1.0.

luma_strength, ls
Set luma maximum difference between pixels to still be considered,
must be a value in the 0.1-100.0 range, default value is 1.0.

chroma_radius, cr
Set chroma blur filter strength, must be a value in range 0.1-4.0.
A greater value will result in a more blurred image, and in slower
processing.

chroma_pre_filter_radius, cpfr
Set chroma pre-filter radius, must be a value in the 0.1-2.0 range.

chroma_strength, cs
Set chroma maximum difference between pixels to still be
considered, must be a value in the 0.1-100.0 range.

Each chroma option value, if not explicitly specified, is set to the
corresponding luma option value.

scale
Scale (resize) the input video, using the libswscale library.

The scale filter forces the output display aspect ratio to be the same
of the input, by changing the output sample aspect ratio.

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.

Options

The filter accepts the following options, or any of the options
supported by the libswscale scaler.

See the ffmpeg-scaler manual for the complete list of scaler options.

width, w
height, h
Set the output video dimension expression. Default value is the
input dimension.

If the value is 0, the input width is used for the output.

If one of the values is -1, the scale filter will use a value that
maintains the aspect ratio of the input image, calculated from the
other specified dimension. If both of them are -1, the input size
is used

If one of the values is -n with n > 1, the scale filter will also
use a value that maintains the aspect ratio of the input image,
calculated from the other specified dimension. After that it will,
however, make sure that the calculated dimension is divisible by n
and adjust the value if necessary.

See below for the list of accepted constants for use in the
dimension expression.

interl
Set the interlacing mode. It accepts the following values:

1 Force interlaced aware scaling.

0 Do not apply interlaced scaling.

-1 Select interlaced aware scaling depending on whether the source
frames are flagged as interlaced or not.

Default value is 0.

flags
Set libswscale scaling flags. See the ffmpeg-scaler manual for the
complete list of values. If not explicitly specified the filter
applies the default flags.

size, s
Set the video size. For the syntax of this option, check the "Video
size" section in the ffmpeg-utils manual.

in_color_matrix
out_color_matrix
Set in/output YCbCr color space type.

This allows the autodetected value to be overridden as well as
allows forcing a specific value used for the output and encoder.

If not specified, the color space type depends on the pixel format.

Possible values:

auto
Choose automatically.

bt709
Format conforming to International Telecommunication Union
(ITU) Recommendation BT.709.

fcc Set color space conforming to the United States Federal
Communications Commission (FCC) Code of Federal Regulations
(CFR) Title 47 (2003) 73.682 (a).

bt601
Set color space conforming to:

· ITU Radiocommunication Sector (ITU-R) Recommendation BT.601

· ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G

· Society of Motion Picture and Television Engineers (SMPTE)
ST 170:2004

smpte240m
Set color space conforming to SMPTE ST 240:1999.

in_range
out_range
Set in/output YCbCr sample range.

This allows the autodetected value to be overridden as well as
allows forcing a specific value used for the output and encoder. If
not specified, the range depends on the pixel format. Possible
values:

auto
Choose automatically.

jpeg/full/pc
Set full range (0-255 in case of 8-bit luma).

mpeg/tv
Set "MPEG" range (16-235 in case of 8-bit luma).

force_original_aspect_ratio
Enable decreasing or increasing output video width or height if
necessary to keep the original aspect ratio. Possible values:

disable
Scale the video as specified and disable this feature.

decrease
The output video dimensions will automatically be decreased if
needed.

increase
The output video dimensions will automatically be increased if
needed.

One useful instance of this option is that when you know a specific
device's maximum allowed resolution, you can use this to limit the
output video to that, while retaining the aspect ratio. For
example, device A allows 1280x720 playback, and your video is
1920x800. Using this option (set it to decrease) and specifying
1280x720 to the command line makes the output 1280x533.

Please note that this is a different thing than specifying -1 for w
or h, you still need to specify the output resolution for this
option to work.

The values of the w and h options are expressions containing the
following constants:

in_w
in_h
The input width and height

iw
ih These are the same as in_w and in_h.

out_w
out_h
The output (scaled) width and height

ow
oh These are the same as out_w and out_h

a The same as iw / ih

sar input sample aspect ratio

dar The input display aspect ratio. Calculated from "(iw / ih) * sar".

hsub
vsub
horizontal and vertical input chroma subsample values. For example
for the pixel format "yuv422p" hsub is 2 and vsub is 1.

ohsub
ovsub
horizontal and vertical output chroma subsample values. For example
for the pixel format "yuv422p" hsub is 2 and vsub is 1.

Examples

· Scale the input video to a size of 200x100

scale=w=200:h=100

This is equivalent to:

scale=200:100

or:

scale=200x100

· Specify a size abbreviation for the output size:

scale=qcif

which can also be written as:

scale=size=qcif

· Scale the input to 2x:

scale=w=2*iw:h=2*ih

· The above is the same as:

scale=2*in_w:2*in_h

· Scale the input to 2x with forced interlaced scaling:

scale=2*iw:2*ih:interl=1

· Scale the input to half size:

scale=w=iw/2:h=ih/2

· Increase the width, and set the height to the same size:

scale=3/2*iw:ow

· Seek 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=w=3/2*oh:h=3/5*ih

· Increase the size, making the size a multiple of the chroma
subsample values:

scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub"

· Increase the width to a maximum of 500 pixels, keeping the same
aspect ratio as the input:

scale=w='min(500\, iw*3/2):h=-1'

Commands

This filter supports the following commands:

width, w
height, h
Set the output video dimension expression. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

scale2ref
Scale (resize) the input video, based on a reference video.

See the scale filter for available options, scale2ref supports the same
but uses the reference video instead of the main input as basis.

Examples

· Scale a subtitle stream to match the main video in size before
overlaying

'scale2ref[b][a];[a][b]overlay'

separatefields
The "separatefields" takes a frame-based video input and splits each
frame into its components fields, producing a new half height clip with
twice the frame rate and twice the frame count.

This filter use field-dominance information in frame to decide which of
each pair of fields to place first in the output. If it gets it wrong
use setfield filter before "separatefields" filter.

setdar, setsar
The "setdar" filter sets 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:

Keep in mind that the "setdar" 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 "setsar" filter sets 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 equation
above.

Keep in mind that the sample aspect ratio set by the "setsar" filter
may be changed by later filters in the filterchain, e.g. if another
"setsar" or a "setdar" filter is applied.

It accepts the following parameters:

r, ratio, dar ("setdar" only), sar ("setsar" only)
Set the aspect ratio used by the filter.

The parameter can be a floating point number string, an expression,
or a string 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". In case the form
"num:den" is used, the ":" character should be escaped.

max Set the maximum integer value to use for expressing numerator and
denominator when reducing the expressed aspect ratio to a rational.
Default value is 100.

The parameter sar is an expression containing the following constants:

E, PI, PHI
These are approximated values for the mathematical constants e
(Euler's number), pi (Greek pi), and phi (the golden ratio).

w, h
The input width and height.

a These are the same as w / h.

sar The input sample aspect ratio.

dar The input display aspect ratio. It is the same as (w / h) * sar.

hsub, vsub
Horizontal and vertical chroma subsample values. For example, for
the pixel format "yuv422p" hsub is 2 and vsub is 1.

Examples

· To change the display aspect ratio to 16:9, specify one of the
following:

setdar=dar=1.77777
setdar=dar=16/9
setdar=dar=1.77777

· To change the sample aspect ratio to 10:11, specify:

setsar=sar=10/11

· To set a display aspect ratio of 16:9, and specify a maximum
integer value of 1000 in the aspect ratio reduction, use the
command:

setdar=ratio=16/9:max=1000

setfield
Force field for the output video frame.

The "setfield" filter marks the interlace type field for the output
frames. It does not change the input frame, but only sets the
corresponding property, which affects how the frame is treated by
following filters (e.g. "fieldorder" or "yadif").

The filter accepts the following options:

mode
Available values are:

auto
Keep the same field property.

bff Mark the frame as bottom-field-first.

tff Mark the frame as top-field-first.

prog
Mark the frame as progressive.

showinfo
Show a line containing various information for each input video frame.
The input video is not modified.

The shown line contains a sequence of key/value pairs of the form
key:value.

The following values are shown in the output:

n The (sequential) number of the input frame, starting from 0.

pts The Presentation TimeStamp of the input frame, expressed as a
number of time base units. The time base unit depends on the filter
input pad.

pts_time
The Presentation TimeStamp of the input frame, expressed as a
number of seconds.

pos The position of the frame in the input stream, or -1 if this
information is unavailable and/or meaningless (for example in case
of synthetic video).

fmt The pixel format name.

sar The sample aspect ratio of the input frame, expressed in the form
num/den.

s The size of the input frame. For the syntax of this option, check
the "Video size" section in the ffmpeg-utils manual.

i The type of interlaced mode ("P" for "progressive", "T" for top
field first, "B" for bottom field first).

iskey
This is 1 if the frame is a key frame, 0 otherwise.

type
The picture type of the input frame ("I" for an I-frame, "P" for a
P-frame, "B" for a B-frame, or "?" for an unknown type). Also
refer to the documentation of the "AVPictureType" enum and of the
"av_get_picture_type_char" function defined in libavutil/avutil.h.

checksum
The Adler-32 checksum (printed in hexadecimal) of all the planes of
the input frame.

plane_checksum
The Adler-32 checksum (printed in hexadecimal) of each plane of the
input frame, expressed in the form "[c0 c1 c2 c3]".

showpalette
Displays the 256 colors palette of each frame. This filter is only
relevant for pal8 pixel format frames.

It accepts the following option:

s Set the size of the box used to represent one palette color entry.
Default is 30 (for a "30x30" pixel box).

shuffleplanes
Reorder and/or duplicate video planes.

It accepts the following parameters:

map0
The index of the input plane to be used as the first output plane.

map1
The index of the input plane to be used as the second output plane.

map2
The index of the input plane to be used as the third output plane.

map3
The index of the input plane to be used as the fourth output plane.

The first plane has the index 0. The default is to keep the input
unchanged.

Swap the second and third planes of the input:

ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT

signalstats
Evaluate various visual metrics that assist in determining issues
associated with the digitization of analog video media.

By default the filter will log these metadata values:

YMIN
Display the minimal Y value contained within the input frame.
Expressed in range of [0-255].

YLOW
Display the Y value at the 10% percentile within the input frame.
Expressed in range of [0-255].

YAVG
Display the average Y value within the input frame. Expressed in
range of [0-255].

YHIGH
Display the Y value at the 90% percentile within the input frame.
Expressed in range of [0-255].

YMAX
Display the maximum Y value contained within the input frame.
Expressed in range of [0-255].

UMIN
Display the minimal U value contained within the input frame.
Expressed in range of [0-255].

ULOW
Display the U value at the 10% percentile within the input frame.
Expressed in range of [0-255].

UAVG
Display the average U value within the input frame. Expressed in
range of [0-255].

UHIGH
Display the U value at the 90% percentile within the input frame.
Expressed in range of [0-255].

UMAX
Display the maximum U value contained within the input frame.
Expressed in range of [0-255].

VMIN
Display the minimal V value contained within the input frame.
Expressed in range of [0-255].

VLOW
Display the V value at the 10% percentile within the input frame.
Expressed in range of [0-255].

VAVG
Display the average V value within the input frame. Expressed in
range of [0-255].

VHIGH
Display the V value at the 90% percentile within the input frame.
Expressed in range of [0-255].

VMAX
Display the maximum V value contained within the input frame.
Expressed in range of [0-255].

SATMIN
Display the minimal saturation value contained within the input
frame. Expressed in range of [0-~181.02].

SATLOW
Display the saturation value at the 10% percentile within the input
frame. Expressed in range of [0-~181.02].

SATAVG
Display the average saturation value within the input frame.
Expressed in range of [0-~181.02].

SATHIGH
Display the saturation value at the 90% percentile within the input
frame. Expressed in range of [0-~181.02].

SATMAX
Display the maximum saturation value contained within the input
frame. Expressed in range of [0-~181.02].

HUEMED
Display the median value for hue within the input frame. Expressed
in range of [0-360].

HUEAVG
Display the average value for hue within the input frame. Expressed
in range of [0-360].

YDIF
Display the average of sample value difference between all values
of the Y plane in the current frame and corresponding values of the
previous input frame. Expressed in range of [0-255].

UDIF
Display the average of sample value difference between all values
of the U plane in the current frame and corresponding values of the
previous input frame. Expressed in range of [0-255].

VDIF
Display the average of sample value difference between all values
of the V plane in the current frame and corresponding values of the
previous input frame. Expressed in range of [0-255].

The filter accepts the following options:

stat
out stat specify an additional form of image analysis. out output
video with the specified type of pixel highlighted.

Both options accept the following values:

tout
Identify temporal outliers pixels. A temporal outlier is a
pixel unlike the neighboring pixels of the same field. Examples
of temporal outliers include the results of video dropouts,
head clogs, or tape tracking issues.

vrep
Identify vertical line repetition. Vertical line repetition
includes similar rows of pixels within a frame. In born-digital
video vertical line repetition is common, but this pattern is
uncommon in video digitized from an analog source. When it
occurs in video that results from the digitization of an analog
source it can indicate concealment from a dropout compensator.

brng
Identify pixels that fall outside of legal broadcast range.

color, c
Set the highlight color for the out option. The default color is
yellow.

Examples

· Output data of various video metrics:

ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames

· Output specific data about the minimum and maximum values of the Y
plane per frame:

ffprobe -f lavfi movie=example.mov,signalstats -show_entries frame_tags=lavfi.signalstats.YMAX,lavfi.signalstats.YMIN

· Playback video while highlighting pixels that are outside of
broadcast range in red.

ffplay example.mov -vf signalstats="out=brng:color=red"

· Playback video with signalstats metadata drawn over the frame.

ffplay example.mov -vf signalstats=stat=brng+vrep+tout,drawtext=fontfile=FreeSerif.ttf:textfile=signalstat_drawtext.txt

The contents of signalstat_drawtext.txt used in the command are:

time %{pts:hms}
Y (%{metadata:lavfi.signalstats.YMIN}-%{metadata:lavfi.signalstats.YMAX})
U (%{metadata:lavfi.signalstats.UMIN}-%{metadata:lavfi.signalstats.UMAX})
V (%{metadata:lavfi.signalstats.VMIN}-%{metadata:lavfi.signalstats.VMAX})
saturation maximum: %{metadata:lavfi.signalstats.SATMAX}

smartblur
Blur the input video without impacting the outlines.

It accepts the following options:

luma_radius, lr
Set the luma radius. The option value must be a float number in the
range [0.1,5.0] that specifies the variance of the gaussian filter
used to blur the image (slower if larger). Default value is 1.0.

luma_strength, ls
Set the luma strength. The option value must be a float number in
the range [-1.0,1.0] that configures the blurring. A value included
in [0.0,1.0] will blur the image whereas a value included in
[-1.0,0.0] will sharpen the image. Default value is 1.0.

luma_threshold, lt
Set the luma threshold used as a coefficient to determine whether a
pixel should be blurred or not. The option value must be an integer
in the range [-30,30]. A value of 0 will filter all the image, a
value included in [0,30] will filter flat areas and a value
included in [-30,0] will filter edges. Default value is 0.

chroma_radius, cr
Set the chroma radius. The option value must be a float number in
the range [0.1,5.0] that specifies the variance of the gaussian
filter used to blur the image (slower if larger). Default value is
1.0.

chroma_strength, cs
Set the chroma strength. The option value must be a float number in
the range [-1.0,1.0] that configures the blurring. A value included
in [0.0,1.0] will blur the image whereas a value included in
[-1.0,0.0] will sharpen the image. Default value is 1.0.

chroma_threshold, ct
Set the chroma threshold used as a coefficient to determine whether
a pixel should be blurred or not. The option value must be an
integer in the range [-30,30]. A value of 0 will filter all the
image, a value included in [0,30] will filter flat areas and a
value included in [-30,0] will filter edges. Default value is 0.

If a chroma option is not explicitly set, the corresponding luma value
is set.

ssim
Obtain the SSIM (Structural SImilarity Metric) between two input
videos.

Both video inputs must have the same resolution and pixel format for
this filter to work correctly. Also it assumes that both inputs have
the same number of frames, which are compared one by one.

The filter stores the calculated SSIM of each frame.

The description of the accepted parameters follows.

stats_file, f
If specified the filter will use the named file to save the SSIM of
each individual frame.

The file printed if stats_file is selected, contains a sequence of
key/value pairs of the form key:value for each compared couple of
frames.

A description of each shown parameter follows:

n sequential number of the input frame, starting from 1

Y, U, V, R, G, B
SSIM of the compared frames for the component specified by the
suffix.

All SSIM of the compared frames for the whole frame.

dB Same as above but in dB representation.

For example:

movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
[main][ref] ssim="stats_file=stats.log" [out]

On this example the input file being processed is compared with the
reference file ref_movie.mpg. The SSIM of each individual frame is
stored in stats.log.

Another example with both psnr and ssim at same time:

ffmpeg -i main.mpg -i ref.mpg -lavfi "ssim;[0:v][1:v]psnr" -f null -

stereo3d
Convert between different stereoscopic image formats.

The filters accept the following options:

in Set stereoscopic image format of input.

Available values for input image formats are:

sbsl
side by side parallel (left eye left, right eye right)

sbsr
side by side crosseye (right eye left, left eye right)

sbs2l
side by side parallel with half width resolution (left eye
left, right eye right)

sbs2r
side by side crosseye with half width resolution (right eye
left, left eye right)

abl above-below (left eye above, right eye below)

abr above-below (right eye above, left eye below)

ab2l
above-below with half height resolution (left eye above, right
eye below)

ab2r
above-below with half height resolution (right eye above, left
eye below)

al alternating frames (left eye first, right eye second)

ar alternating frames (right eye first, left eye second)

Default value is sbsl.

out Set stereoscopic image format of output.

Available values for output image formats are all the input formats
as well as:

arbg
anaglyph red/blue gray (red filter on left eye, blue filter on
right eye)

argg
anaglyph red/green gray (red filter on left eye, green filter
on right eye)

arcg
anaglyph red/cyan gray (red filter on left eye, cyan filter on
right eye)

arch
anaglyph red/cyan half colored (red filter on left eye, cyan
filter on right eye)

arcc
anaglyph red/cyan color (red filter on left eye, cyan filter on
right eye)

arcd
anaglyph red/cyan color optimized with the least squares
projection of dubois (red filter on left eye, cyan filter on
right eye)

agmg
anaglyph green/magenta gray (green filter on left eye, magenta
filter on right eye)

agmh
anaglyph green/magenta half colored (green filter on left eye,
magenta filter on right eye)

agmc
anaglyph green/magenta colored (green filter on left eye,
magenta filter on right eye)

agmd
anaglyph green/magenta color optimized with the least squares
projection of dubois (green filter on left eye, magenta filter
on right eye)

aybg
anaglyph yellow/blue gray (yellow filter on left eye, blue
filter on right eye)

aybh
anaglyph yellow/blue half colored (yellow filter on left eye,
blue filter on right eye)

aybc
anaglyph yellow/blue colored (yellow filter on left eye, blue
filter on right eye)

aybd
anaglyph yellow/blue color optimized with the least squares
projection of dubois (yellow filter on left eye, blue filter on
right eye)

irl interleaved rows (left eye has top row, right eye starts on
next row)

irr interleaved rows (right eye has top row, left eye starts on
next row)

ml mono output (left eye only)

mr mono output (right eye only)

Default value is arcd.

Examples

· Convert input video from side by side parallel to anaglyph
yellow/blue dubois:

stereo3d=sbsl:aybd

· Convert input video from above below (left eye above, right eye
below) to side by side crosseye.

stereo3d=abl:sbsr

spp
Apply a simple postprocessing filter that compresses and decompresses
the image at several (or - in the case of quality level 6 - all) shifts
and average the results.

The filter accepts the following options:

quality
Set quality. This option defines the number of levels for
averaging. It accepts an integer in the range 0-6. If set to 0, the
filter will have no effect. A value of 6 means the higher quality.
For each increment of that value the speed drops by a factor of
approximately 2. Default value is 3.

qp Force a constant quantization parameter. If not set, the filter
will use the QP from the video stream (if available).

mode
Set thresholding mode. Available modes are:

hard
Set hard thresholding (default).

soft
Set soft thresholding (better de-ringing effect, but likely
blurrier).

use_bframe_qp
Enable the use of the QP from the B-Frames if set to 1. Using this
option may cause flicker since the B-Frames have often larger QP.
Default is 0 (not enabled).

subtitles
Draw subtitles on top of input video using the libass library.

To enable compilation of this filter you need to configure FFmpeg with
"--enable-libass". This filter also requires a build with libavcodec
and libavformat to convert the passed subtitles file to ASS (Advanced
Substation Alpha) subtitles format.

The filter accepts the following options:

filename, f
Set the filename of the subtitle file to read. It must be
specified.

original_size
Specify the size of the original video, the video for which the ASS
file was composed. For the syntax of this option, check the "Video
size" section in the ffmpeg-utils manual. Due to a misdesign in
ASS aspect ratio arithmetic, this is necessary to correctly scale
the fonts if the aspect ratio has been changed.

fontsdir
Set a directory path containing fonts that can be used by the
filter. These fonts will be used in addition to whatever the font
provider uses.

charenc
Set subtitles input character encoding. "subtitles" filter only.
Only useful if not UTF-8.

stream_index, si
Set subtitles stream index. "subtitles" filter only.

force_style
Override default style or script info parameters of the subtitles.
It accepts a string containing ASS style format "KEY=VALUE" couples
separated by ",".

If the first key is not specified, it is assumed that the first value
specifies the filename.

For example, to render the file sub.srt on top of the input video, use
the command:

subtitles=sub.srt

which is equivalent to:

subtitles=filename=sub.srt

To render the default subtitles stream from file video.mkv, use:

subtitles=video.mkv

To render the second subtitles stream from that file, use:

subtitles=video.mkv:si=1

To make the subtitles stream from sub.srt appear in transparent green
"DejaVu Serif", use:

subtitles=sub.srt:force_style='FontName=DejaVu Serif,PrimaryColour=&HAA00FF00'

super2xsai
Scale the input by 2x and smooth using the Super2xSaI (Scale and
Interpolate) pixel art scaling algorithm.

Useful for enlarging pixel art images without reducing sharpness.

swapuv
Swap U & V plane.

telecine
Apply telecine process to the video.

This filter accepts the following options:

first_field
top, t
top field first

bottom, b
bottom field first The default value is "top".

pattern
A string of numbers representing the pulldown pattern you wish to
apply. The default value is 23.

Some typical patterns:

NTSC output (30i):
27.5p: 32222
24p: 23 (classic)
24p: 2332 (preferred)
20p: 33
18p: 334
16p: 3444

PAL output (25i):
27.5p: 12222
24p: 222222222223 ("Euro pulldown")
16.67p: 33
16p: 33333334

thumbnail
Select the most representative frame in a given sequence of consecutive
frames.

The filter accepts the following options:

n Set the frames batch size to analyze; in a set of n frames, the
filter will pick one of them, and then handle the next batch of n
frames until the end. Default is 100.

Since the filter keeps track of the whole frames sequence, a bigger n
value will result in a higher memory usage, so a high value is not
recommended.

Examples

· Extract one picture each 50 frames:

thumbnail=50

· Complete example of a thumbnail creation with ffmpeg:

ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png

tile
Tile several successive frames together.

The filter accepts the following options:

layout
Set the grid size (i.e. the number of lines and columns). For the
syntax of this option, check the "Video size" section in the
ffmpeg-utils manual.

nb_frames
Set the maximum number of frames to render in the given area. It
must be less than or equal to wxh. The default value is 0, meaning
all the area will be used.

margin
Set the outer border margin in pixels.

padding
Set the inner border thickness (i.e. the number of pixels between
frames). For more advanced padding options (such as having
different values for the edges), refer to the pad video filter.

color
Specify the color of the unused area. For the syntax of this
option, check the "Color" section in the ffmpeg-utils manual. The
default value of color is "black".

Examples

· Produce 8x8 PNG tiles of all keyframes (-skip_frame nokey) in a
movie:

ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png

The -vsync 0 is necessary to prevent ffmpeg from duplicating each
output frame to accommodate the originally detected frame rate.

· Display 5 pictures in an area of "3x2" frames, with 7 pixels
between them, and 2 pixels of initial margin, using mixed flat and
named options:

tile=3x2:nb_frames=5:padding=7:margin=2

tinterlace
Perform various types of temporal field interlacing.

Frames are counted starting from 1, so the first input frame is
considered odd.

The filter accepts the following options:

mode
Specify the mode of the interlacing. This option can also be
specified as a value alone. See below for a list of values for this
option.

Available values are:

merge, 0
Move odd frames into the upper field, even into the lower
field, generating a double height frame at half frame rate.

------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4

11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444

Output:
11111 33333
22222 44444
11111 33333
22222 44444
11111 33333
22222 44444
11111 33333
22222 44444

drop_odd, 1
Only output even frames, odd frames are dropped, generating a
frame with unchanged height at half frame rate.

------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4

11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444

Output:
22222 44444
22222 44444
22222 44444
22222 44444

drop_even, 2
Only output odd frames, even frames are dropped, generating a
frame with unchanged height at half frame rate.

------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4

11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444

Output:
11111 33333
11111 33333
11111 33333
11111 33333

pad, 3
Expand each frame to full height, but pad alternate lines with
black, generating a frame with double height at the same input
frame rate.

------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4

11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444

Output:
11111 ..... 33333 .....
..... 22222 ..... 44444
11111 ..... 33333 .....
..... 22222 ..... 44444
11111 ..... 33333 .....
..... 22222 ..... 44444
11111 ..... 33333 .....
..... 22222 ..... 44444

interleave_top, 4
Interleave the upper field from odd frames with the lower field
from even frames, generating a frame with unchanged height at
half frame rate.

------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4

11111<- 22222 33333<- 44444
11111 22222<- 33333 44444<-
11111<- 22222 33333<- 44444
11111 22222<- 33333 44444<-

Output:
11111 33333
22222 44444
11111 33333
22222 44444

interleave_bottom, 5
Interleave the lower field from odd frames with the upper field
from even frames, generating a frame with unchanged height at
half frame rate.

------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4

11111 22222<- 33333 44444<-
11111<- 22222 33333<- 44444
11111 22222<- 33333 44444<-
11111<- 22222 33333<- 44444

Output:
22222 44444
11111 33333
22222 44444
11111 33333

interlacex2, 6
Double frame rate with unchanged height. Frames are inserted
each containing the second temporal field from the previous
input frame and the first temporal field from the next input
frame. This mode relies on the top_field_first flag. Useful for
interlaced video displays with no field synchronisation.

------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4

11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444

Output:
11111 22222 22222 33333 33333 44444 44444
11111 11111 22222 22222 33333 33333 44444
11111 22222 22222 33333 33333 44444 44444
11111 11111 22222 22222 33333 33333 44444

Numeric values are deprecated but are accepted for backward
compatibility reasons.

Default mode is "merge".

flags
Specify flags influencing the filter process.

Available value for flags is:

low_pass_filter, vlfp
Enable vertical low-pass filtering in the filter. Vertical
low-pass filtering is required when creating an interlaced
destination from a progressive source which contains high-
frequency vertical detail. Filtering will reduce interlace
'twitter' and Moire patterning.

Vertical low-pass filtering can only be enabled for mode
interleave_top and interleave_bottom.

transpose
Transpose rows with columns in the input video and optionally flip it.

It accepts the following parameters:

dir Specify the transposition direction.

Can assume the following values:

0, 4, cclock_flip
Rotate by 90 degrees counterclockwise and vertically flip
(default), that is:

L.R L.l
. . -> . .
l.r R.r

1, 5, clock
Rotate by 90 degrees clockwise, that is:

L.R l.L
. . -> . .
l.r r.R

2, 6, cclock
Rotate by 90 degrees counterclockwise, that is:

L.R R.r
. . -> . .
l.r L.l

3, 7, clock_flip
Rotate by 90 degrees clockwise and vertically flip, that is:

L.R r.R
. . -> . .
l.r l.L

For values between 4-7, the transposition is only done if the input
video geometry is portrait and not landscape. These values are
deprecated, the "passthrough" option should be used instead.

Numerical values are deprecated, and should be dropped in favor of
symbolic constants.

passthrough
Do not apply the transposition if the input geometry matches the
one specified by the specified value. It accepts the following
values:

none
Always apply transposition.

portrait
Preserve portrait geometry (when height >= width).

landscape
Preserve landscape geometry (when width >= height).

Default value is "none".

For example to rotate by 90 degrees clockwise and preserve portrait
layout:

transpose=dir=1:passthrough=portrait

The command above can also be specified as:

transpose=1:portrait

trim
Trim the input so that the output contains one continuous subpart of
the input.

It accepts the following parameters:

start
Specify the time of the start of the kept section, i.e. the frame
with the timestamp start will be the first frame in the output.

end Specify the time of the first frame that will be dropped, i.e. the
frame immediately preceding the one with the timestamp end will be
the last frame in the output.

start_pts
This is the same as start, except this option sets the start
timestamp in timebase units instead of seconds.

end_pts
This is the same as end, except this option sets the end timestamp
in timebase units instead of seconds.

duration
The maximum duration of the output in seconds.

start_frame
The number of the first frame that should be passed to the output.

end_frame
The number of the first frame that should be dropped.

start, end, and duration are expressed as time duration specifications;
see the Time duration section in the ffmpeg-utils(1) manual for the
accepted syntax.

Note that the first two sets of the start/end options and the duration
option look at the frame timestamp, while the _frame variants simply
count the frames that pass through the filter. Also note that this
filter does not modify the timestamps. If you wish for the output
timestamps to start at zero, insert a setpts filter after the trim
filter.

If multiple start or end options are set, this filter tries to be
greedy and keep all the frames that match at least one of the specified
constraints. To keep only the part that matches all the constraints at
once, chain multiple trim filters.

The defaults are such that all the input is kept. So it is possible to
set e.g. just the end values to keep everything before the specified
time.

Examples:

· Drop everything except the second minute of input:

ffmpeg -i INPUT -vf trim=60:120

· Keep only the first second:

ffmpeg -i INPUT -vf trim=duration=1

unsharp
Sharpen or blur the input video.

It accepts the following parameters:

luma_msize_x, lx
Set the luma matrix horizontal size. It must be an odd integer
between 3 and 63. The default value is 5.

luma_msize_y, ly
Set the luma matrix vertical size. It must be an odd integer
between 3 and 63. The default value is 5.

luma_amount, la
Set the luma effect strength. It must be a floating point number,
reasonable values lay between -1.5 and 1.5.

Negative values will blur the input video, while positive values
will sharpen it, a value of zero will disable the effect.

Default value is 1.0.

chroma_msize_x, cx
Set the chroma matrix horizontal size. It must be an odd integer
between 3 and 63. The default value is 5.

chroma_msize_y, cy
Set the chroma matrix vertical size. It must be an odd integer
between 3 and 63. The default value is 5.

chroma_amount, ca
Set the chroma effect strength. It must be a floating point number,
reasonable values lay between -1.5 and 1.5.

Negative values will blur the input video, while positive values
will sharpen it, a value of zero will disable the effect.

Default value is 0.0.

opencl
If set to 1, specify using OpenCL capabilities, only available if
FFmpeg was configured with "--enable-opencl". Default value is 0.

All parameters are optional and default to the equivalent of the string
'5:5:1.0:5:5:0.0'.

Examples

· Apply strong luma sharpen effect:

unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5

· Apply a strong blur of both luma and chroma parameters:

unsharp=7:7:-2:7:7:-2

uspp
Apply ultra slow/simple postprocessing filter that compresses and
decompresses the image at several (or - in the case of quality level 8
- all) shifts and average the results.

The way this differs from the behavior of spp is that uspp actually
encodes & decodes each case with libavcodec Snow, whereas spp uses a
simplified intra only 8x8 DCT similar to MJPEG.

The filter accepts the following options:

quality
Set quality. This option defines the number of levels for
averaging. It accepts an integer in the range 0-8. If set to 0, the
filter will have no effect. A value of 8 means the higher quality.
For each increment of that value the speed drops by a factor of
approximately 2. Default value is 3.

qp Force a constant quantization parameter. If not set, the filter
will use the QP from the video stream (if available).

vectorscope
Display 2 color component values in the two dimensional graph (which is
called a vectorscope).

This filter accepts the following options:

mode, m
Set vectorscope mode.

It accepts the following values:

gray
Gray values are displayed on graph, higher brightness means
more pixels have same component color value on location in
graph. This is the default mode.

color
Gray values are displayed on graph. Surrounding pixels values
which are not present in video frame are drawn in gradient of 2
color components which are set by option "x" and "y".

color2
Actual color components values present in video frame are
displayed on graph.

color3
Similar as color2 but higher frequency of same values "x" and
"y" on graph increases value of another color component, which
is luminance by default values of "x" and "y".

color4
Actual colors present in video frame are displayed on graph. If
two different colors map to same position on graph then color
with higher value of component not present in graph is picked.

x Set which color component will be represented on X-axis. Default is
1.

y Set which color component will be represented on Y-axis. Default is
2.

intensity, i
Set intensity, used by modes: gray, color and color3 for increasing
brightness of color component which represents frequency of (X, Y)
location in graph.

envelope, e
none
No envelope, this is default.

instant
Instant envelope, even darkest single pixel will be clearly
highlighted.

peak
Hold maximum and minimum values presented in graph over time.
This way you can still spot out of range values without
constantly looking at vectorscope.

peak+instant
Peak and instant envelope combined together.

vidstabdetect
Analyze video stabilization/deshaking. Perform pass 1 of 2, see
vidstabtransform for pass 2.

This filter generates a file with relative translation and rotation
transform information about subsequent frames, which is then used by
the vidstabtransform filter.

To enable compilation of this filter you need to configure FFmpeg with
"--enable-libvidstab".

This filter accepts the following options:

result
Set the path to the file used to write the transforms information.
Default value is transforms.trf.

shakiness
Set how shaky the video is and how quick the camera is. It accepts
an integer in the range 1-10, a value of 1 means little shakiness,
a value of 10 means strong shakiness. Default value is 5.

accuracy
Set the accuracy of the detection process. It must be a value in
the range 1-15. A value of 1 means low accuracy, a value of 15
means high accuracy. Default value is 15.

stepsize
Set stepsize of the search process. The region around minimum is
scanned with 1 pixel resolution. Default value is 6.

mincontrast
Set minimum contrast. Below this value a local measurement field is
discarded. Must be a floating point value in the range 0-1. Default
value is 0.3.

tripod
Set reference frame number for tripod mode.

If enabled, the motion of the frames is compared to a reference
frame in the filtered stream, identified by the specified number.
The idea is to compensate all movements in a more-or-less static
scene and keep the camera view absolutely still.

If set to 0, it is disabled. The frames are counted starting from
1.

show
Show fields and transforms in the resulting frames. It accepts an
integer in the range 0-2. Default value is 0, which disables any
visualization.

Examples

· Use default values:

vidstabdetect

· Analyze strongly shaky movie and put the results in file
mytransforms.trf:

vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"

· Visualize the result of internal transformations in the resulting
video:

vidstabdetect=show=1

· Analyze a video with medium shakiness using ffmpeg:

ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi

vidstabtransform
Video stabilization/deshaking: pass 2 of 2, see vidstabdetect for pass
1.

Read a file with transform information for each frame and
apply/compensate them. Together with the vidstabdetect filter this can
be used to deshake videos. See also
<http://public.hronopik.de/vid.stab>. It is important to also use the
unsharp filter, see below.

To enable compilation of this filter you need to configure FFmpeg with
"--enable-libvidstab".

Options

input
Set path to the file used to read the transforms. Default value is
transforms.trf.

smoothing
Set the number of frames (value*2 + 1) used for lowpass filtering
the camera movements. Default value is 10.

For example a number of 10 means that 21 frames are used (10 in the
past and 10 in the future) to smoothen the motion in the video. A
larger value leads to a smoother video, but limits the acceleration
of the camera (pan/tilt movements). 0 is a special case where a
static camera is simulated.

optalgo
Set the camera path optimization algorithm.

Accepted values are:

gauss
gaussian kernel low-pass filter on camera motion (default)

avg averaging on transformations

maxshift
Set maximal number of pixels to translate frames. Default value is
-1, meaning no limit.

maxangle
Set maximal angle in radians (degree*PI/180) to rotate frames.
Default value is -1, meaning no limit.

crop
Specify how to deal with borders that may be visible due to
movement compensation.

Available values are:

keep
keep image information from previous frame (default)

black
fill the border black

invert
Invert transforms if set to 1. Default value is 0.

relative
Consider transforms as relative to previous frame if set to 1,
absolute if set to 0. Default value is 0.

zoom
Set percentage to zoom. A positive value will result in a zoom-in
effect, a negative value in a zoom-out effect. Default value is 0
(no zoom).

optzoom
Set optimal zooming to avoid borders.

Accepted values are:

0 disabled

1 optimal static zoom value is determined (only very strong
movements will lead to visible borders) (default)

2 optimal adaptive zoom value is determined (no borders will be
visible), see zoomspeed

Note that the value given at zoom is added to the one calculated
here.

zoomspeed
Set percent to zoom maximally each frame (enabled when optzoom is
set to 2). Range is from 0 to 5, default value is 0.25.

interpol
Specify type of interpolation.

Available values are:

no no interpolation

linear
linear only horizontal

bilinear
linear in both directions (default)

bicubic
cubic in both directions (slow)

tripod
Enable virtual tripod mode if set to 1, which is equivalent to
"relative=0:smoothing=0". Default value is 0.

Use also "tripod" option of vidstabdetect.

debug
Increase log verbosity if set to 1. Also the detected global
motions are written to the temporary file global_motions.trf.
Default value is 0.

Examples

· Use ffmpeg for a typical stabilization with default values:

ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg

Note the use of the unsharp filter which is always recommended.

· Zoom in a bit more and load transform data from a given file:

vidstabtransform=zoom=5:input="mytransforms.trf"

· Smoothen the video even more:

vidstabtransform=smoothing=30

vflip
Flip the input video vertically.

For example, to vertically flip a video with ffmpeg:

ffmpeg -i in.avi -vf "vflip" out.avi

vignette
Make or reverse a natural vignetting effect.

The filter accepts the following options:

angle, a
Set lens angle expression as a number of radians.

The value is clipped in the "[0,PI/2]" range.

Default value: "PI/5"

x0
y0 Set center coordinates expressions. Respectively "w/2" and "h/2" by
default.

mode
Set forward/backward mode.

Available modes are:

forward
The larger the distance from the central point, the darker the
image becomes.

backward
The larger the distance from the central point, the brighter
the image becomes. This can be used to reverse a vignette
effect, though there is no automatic detection to extract the
lens angle and other settings (yet). It can also be used to
create a burning effect.

Default value is forward.

eval
Set evaluation mode for the expressions (angle, x0, y0).

It accepts the following values:

init
Evaluate expressions only once during the filter
initialization.

frame
Evaluate expressions for each incoming frame. This is way
slower than the init mode since it requires all the scalers to
be re-computed, but it allows advanced dynamic expressions.

Default value is init.

dither
Set dithering to reduce the circular banding effects. Default is 1
(enabled).

aspect
Set vignette aspect. This setting allows one to adjust the shape of
the vignette. Setting this value to the SAR of the input will make
a rectangular vignetting following the dimensions of the video.

Default is "1/1".

Expressions

The alpha, x0 and y0 expressions can contain the following parameters.

w
h input width and height

n the number of input frame, starting from 0

pts the PTS (Presentation TimeStamp) time of the filtered video frame,
expressed in TB units, NAN if undefined

r frame rate of the input video, NAN if the input frame rate is
unknown

t the PTS (Presentation TimeStamp) of the filtered video frame,
expressed in seconds, NAN if undefined

tb time base of the input video

Examples

· Apply simple strong vignetting effect:

vignette=PI/4

· Make a flickering vignetting:

vignette='PI/4+random(1)*PI/50':eval=frame

vstack
Stack input videos vertically.

All streams must be of same pixel format and of same width.

Note that this filter is faster than using overlay and pad filter to
create same output.

The filter accept the following option:

nb_inputs
Set number of input streams. Default is 2.

w3fdif
Deinterlace the input video ("w3fdif" stands for "Weston 3 Field
Deinterlacing Filter").

Based on the process described by Martin Weston for BBC R&D, and
implemented based on the de-interlace algorithm written by Jim
Easterbrook for BBC R&D, the Weston 3 field deinterlacing filter uses
filter coefficients calculated by BBC R&D.

There are two sets of filter coefficients, so called "simple": and
"complex". Which set of filter coefficients is used can be set by
passing an optional parameter:

filter
Set the interlacing filter coefficients. Accepts one of the
following values:

simple
Simple filter coefficient set.

complex
More-complex filter coefficient set.

Default value is complex.

deint
Specify which frames to deinterlace. Accept one of the following
values:

all Deinterlace all frames,

interlaced
Only deinterlace frames marked as interlaced.

Default value is all.

waveform
Video waveform monitor.

The waveform monitor plots color component intensity. By default
luminance only. Each column of the waveform corresponds to a column of
pixels in the source video.

It accepts the following options:

mode, m
Can be either "row", or "column". Default is "column". In row
mode, the graph on the left side represents color component value 0
and the right side represents value = 255. In column mode, the top
side represents color component value = 0 and bottom side
represents value = 255.

intensity, i
Set intensity. Smaller values are useful to find out how many
values of the same luminance are distributed across input
rows/columns. Default value is 0.04. Allowed range is [0, 1].

mirror, r
Set mirroring mode. 0 means unmirrored, 1 means mirrored. In
mirrored mode, higher values will be represented on the left side
for "row" mode and at the top for "column" mode. Default is 1
(mirrored).

display, d
Set display mode. It accepts the following values:

overlay
Presents information identical to that in the "parade", except
that the graphs representing color components are superimposed
directly over one another.

This display mode makes it easier to spot relative differences
or similarities in overlapping areas of the color components
that are supposed to be identical, such as neutral whites,
grays, or blacks.

parade
Display separate graph for the color components side by side in
"row" mode or one below the other in "column" mode.

Using this display mode makes it easy to spot color casts in
the highlights and shadows of an image, by comparing the
contours of the top and the bottom graphs of each waveform.
Since whites, grays, and blacks are characterized by exactly
equal amounts of red, green, and blue, neutral areas of the
picture should display three waveforms of roughly equal
width/height. If not, the correction is easy to perform by
making level adjustments the three waveforms.

Default is "parade".

components, c
Set which color components to display. Default is 1, which means
only luminance or red color component if input is in RGB
colorspace. If is set for example to 7 it will display all 3 (if)
available color components.

envelope, e
none
No envelope, this is default.

instant
Instant envelope, minimum and maximum values presented in graph
will be easily visible even with small "step" value.

peak
Hold minimum and maximum values presented in graph across time.
This way you can still spot out of range values without
constantly looking at waveforms.

peak+instant
Peak and instant envelope combined together.

filter, f
lowpass
No filtering, this is default.

flat
Luma and chroma combined together.

aflat
Similar as above, but shows difference between blue and red
chroma.

chroma
Displays only chroma.

achroma
Similar as above, but shows difference between blue and red
chroma.

color
Displays actual color value on waveform.

xbr
Apply the xBR high-quality magnification filter which is designed for
pixel art. It follows a set of edge-detection rules, see
<http://www.libretro.com/forums/viewtopic.php?f=6&t=134>.

It accepts the following option:

n Set the scaling dimension: 2 for "2xBR", 3 for "3xBR" and 4 for
"4xBR". Default is 3.

yadif
Deinterlace the input video ("yadif" means "yet another deinterlacing
filter").

It accepts the following parameters:

mode
The interlacing mode to adopt. It accepts one of the following
values:

0, send_frame
Output one frame for each frame.

1, send_field
Output one frame for each field.

2, send_frame_nospatial
Like "send_frame", but it skips the spatial interlacing check.

3, send_field_nospatial
Like "send_field", but it skips the spatial interlacing check.

The default value is "send_frame".

parity
The picture field parity assumed for the input interlaced video. It
accepts one of the following values:

0, tff
Assume the top field is first.

1, bff
Assume the bottom field is first.

-1, auto
Enable automatic detection of field parity.

The default value is "auto". If the interlacing is unknown or the
decoder does not export this information, top field first will be
assumed.

deint
Specify which frames to deinterlace. Accept one of the following
values:

0, all
Deinterlace all frames.

1, interlaced
Only deinterlace frames marked as interlaced.

The default value is "all".

zoompan
Apply Zoom & Pan effect.

This filter accepts the following options:

zoom, z
Set the zoom expression. Default is 1.

x
y Set the x and y expression. Default is 0.

d Set the duration expression in number of frames. This sets for how
many number of frames effect will last for single input image.

s Set the output image size, default is 'hd720'.

Each expression can contain the following constants:

in_w, iw
Input width.

in_h, ih
Input height.

out_w, ow
Output width.

out_h, oh
Output height.

in Input frame count.

on Output frame count.

x
y Last calculated 'x' and 'y' position from 'x' and 'y' expression
for current input frame.

px
py 'x' and 'y' of last output frame of previous input frame or 0 when
there was not yet such frame (first input frame).

zoom
Last calculated zoom from 'z' expression for current input frame.

pzoom
Last calculated zoom of last output frame of previous input frame.

duration
Number of output frames for current input frame. Calculated from
'd' expression for each input frame.

pduration
number of output frames created for previous input frame

a Rational number: input width / input height

sar sample aspect ratio

dar display aspect ratio

Examples

· Zoom-in up to 1.5 and pan at same time to some spot near center of
picture:

zoompan=z='min(zoom+0.0015,1.5)':d=700:x='if(gte(zoom,1.5),x,x+1/a)':y='if(gte(zoom,1.5),y,y+1)':s=640x360

· Zoom-in up to 1.5 and pan always at center of picture:

zoompan=z='min(zoom+0.0015,1.5)':d=700:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

VIDEO SOURCES
Below is a description of the currently available video sources.

buffer
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:

video_size
Specify the size (width and height) of the buffered video frames.
For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual.

width
The input video width.

height
The input video height.

pix_fmt
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.

time_base
Specify the timebase assumed by the timestamps of the buffered
frames.

frame_rate
Specify the frame rate expected for the video stream.

pixel_aspect, sar
The sample (pixel) aspect ratio of the input video.

sws_param
Specify the optional parameters to be used for the scale filter
which is automatically inserted when an input change is detected in
the input size or format.

For example:

buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=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
AVPixelFormat definition in libavutil/pixfmt.h), this example
corresponds to:

buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1

Alternatively, the options can be specified as a flat string, but this
syntax is deprecated:

width:height:pix_fmt:time_base.num:time_base.den:pixel_aspect.num:pixel_aspect.den[:sws_param]

cellauto
Create a pattern generated by an elementary cellular automaton.

The initial state of the cellular automaton can be defined through the
filename, and pattern options. If such options are not specified an
initial state is created randomly.

At each new frame a new row in the video is filled with the result of
the cellular automaton next generation. The behavior when the whole
frame is filled is defined by the scroll option.

This source accepts the following options:

filename, f
Read the initial cellular automaton state, i.e. the starting row,
from the specified file. In the file, each non-whitespace
character is considered an alive cell, a newline will terminate the
row, and further characters in the file will be ignored.

pattern, p
Read the initial cellular automaton state, i.e. the starting row,
from the specified string.

Each non-whitespace character in the string is considered an alive
cell, a newline will terminate the row, and further characters in
the string will be ignored.

rate, r
Set the video rate, that is the number of frames generated per
second. Default is 25.

random_fill_ratio, ratio
Set the random fill ratio for the initial cellular automaton row.
It is a floating point number value ranging from 0 to 1, defaults
to 1/PHI.

This option is ignored when a file or a pattern is specified.

random_seed, seed
Set the seed for filling randomly the initial row, must be an
integer included between 0 and UINT32_MAX. If not specified, or if
explicitly set to -1, the filter will try to use a good random seed
on a best effort basis.

rule
Set the cellular automaton rule, it is a number ranging from 0 to
255. Default value is 110.

size, s
Set the size of the output video. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual.

If filename or pattern is specified, the size is set by default to
the width of the specified initial state row, and the height is set
to width * PHI.

If size is set, it must contain the width of the specified pattern
string, and the specified pattern will be centered in the larger
row.

If a filename or a pattern string is not specified, the size value
defaults to "320x518" (used for a randomly generated initial
state).

scroll
If set to 1, scroll the output upward when all the rows in the
output have been already filled. If set to 0, the new generated row
will be written over the top row just after the bottom row is
filled. Defaults to 1.

start_full, full
If set to 1, completely fill the output with generated rows before
outputting the first frame. This is the default behavior, for
disabling set the value to 0.

stitch
If set to 1, stitch the left and right row edges together. This is
the default behavior, for disabling set the value to 0.

Examples

· Read the initial state from pattern, and specify an output of size
200x400.

cellauto=f=pattern:s=200x400

· Generate a random initial row with a width of 200 cells, with a
fill ratio of 2/3:

cellauto=ratio=2/3:s=200x200

· Create a pattern generated by rule 18 starting by a single alive
cell centered on an initial row with width 100:

cellauto=p=@s=100x400:full=0:rule=18

· Specify a more elaborated initial pattern:

cellauto=p='@@ @ @@':s=100x400:full=0:rule=18

mandelbrot
Generate a Mandelbrot set fractal, and progressively zoom towards the
point specified with start_x and start_y.

This source accepts the following options:

end_pts
Set the terminal pts value. Default value is 400.

end_scale
Set the terminal scale value. Must be a floating point value.
Default value is 0.3.

inner
Set the inner coloring mode, that is the algorithm used to draw the
Mandelbrot fractal internal region.

It shall assume one of the following values:

black
Set black mode.

convergence
Show time until convergence.

mincol
Set color based on point closest to the origin of the
iterations.

period
Set period mode.

Default value is mincol.

bailout
Set the bailout value. Default value is 10.0.

maxiter
Set the maximum of iterations performed by the rendering algorithm.
Default value is 7189.

outer
Set outer coloring mode. It shall assume one of following values:

iteration_count
Set iteration cound mode.

normalized_iteration_count
set normalized iteration count mode.

Default value is normalized_iteration_count.

rate, r
Set frame rate, expressed as number of frames per second. Default
value is "25".

size, s
Set frame size. For the syntax of this option, check the "Video
size" section in the ffmpeg-utils manual. Default value is
"640x480".

start_scale
Set the initial scale value. Default value is 3.0.

start_x
Set the initial x position. Must be a floating point value between
-100 and 100. Default value is
-0.743643887037158704752191506114774.

start_y
Set the initial y position. Must be a floating point value between
-100 and 100. Default value is
-0.131825904205311970493132056385139.

mptestsrc
Generate various test patterns, as generated by the MPlayer test
filter.

The size of the generated video is fixed, and is 256x256. This source
is useful in particular for testing encoding features.

This source accepts the following options:

rate, r
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 floating point
number or a valid video frame rate abbreviation. The default value
is "25".

duration, d
Set the duration of the sourced video. See the Time duration
section in the ffmpeg-utils(1) manual for the accepted syntax.

If not specified, or the expressed duration is negative, the video
is supposed to be generated forever.

test, t
Set the number or the name of the test to perform. Supported tests
are:

dc_luma
dc_chroma
freq_luma
freq_chroma
amp_luma
amp_chroma
cbp
mv
ring1
ring2
all

Default value is "all", which will cycle through the list of all
tests.

Some examples:

mptestsrc=t=dc_luma

will generate a "dc_luma" test pattern.

frei0r_src
Provide a frei0r source.

To enable compilation of this filter you need to install the frei0r
header and configure FFmpeg with "--enable-frei0r".

This source accepts the following parameters:

size
The size of the video to generate. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual.

framerate
The framerate of the generated video. It may be a string of the
form num/den or a frame rate abbreviation.

filter_name
The name to the frei0r source to load. For more information
regarding frei0r and how to set the parameters, read the frei0r
section in the video filters documentation.

filter_params
A '|'-separated list of parameters to pass to the frei0r source.

For example, to generate a frei0r partik0l source with size 200x200 and
frame rate 10 which is overlaid on the overlay filter main input:

frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay

life
Generate a life pattern.

This source is based on a generalization of John Conway's life game.

The sourced input represents a life grid, each pixel represents a cell
which can be in one of two possible states, alive or dead. Every cell
interacts with its eight neighbours, which are the cells that are
horizontally, vertically, or diagonally adjacent.

At each interaction the grid evolves according to the adopted rule,
which specifies the number of neighbor alive cells which will make a
cell stay alive or born. The rule option allows one to specify the rule
to adopt.

This source accepts the following options:

filename, f
Set the file from which to read the initial grid state. In the
file, each non-whitespace character is considered an alive cell,
and newline is used to delimit the end of each row.

If this option is not specified, the initial grid is generated
randomly.

rate, r
Set the video rate, that is the number of frames generated per
second. Default is 25.

random_fill_ratio, ratio
Set the random fill ratio for the initial random grid. It is a
floating point number value ranging from 0 to 1, defaults to 1/PHI.
It is ignored when a file is specified.

random_seed, seed
Set the seed for filling the initial random grid, must be an
integer included between 0 and UINT32_MAX. If not specified, or if
explicitly set to -1, the filter will try to use a good random seed
on a best effort basis.

rule
Set the life rule.

A rule can be specified with a code of the kind "SNS/BNB", where NS
and NB are sequences of numbers in the range 0-8, NS specifies the
number of alive neighbor cells which make a live cell stay alive,
and NB the number of alive neighbor cells which make a dead cell to
become alive (i.e. to "born"). "s" and "b" can be used in place of
"S" and "B", respectively.

Alternatively a rule can be specified by an 18-bits integer. The 9
high order bits are used to encode the next cell state if it is
alive for each number of neighbor alive cells, the low order bits
specify the rule for "borning" new cells. Higher order bits encode
for an higher number of neighbor cells. For example the number
6153 = "(12<<9)+9" specifies a stay alive rule of 12 and a born
rule of 9, which corresponds to "S23/B03".

Default value is "S23/B3", which is the original Conway's game of
life rule, and will keep a cell alive if it has 2 or 3 neighbor
alive cells, and will born a new cell if there are three alive
cells around a dead cell.

size, s
Set the size of the output video. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual.

If filename is specified, the size is set by default to the same
size of the input file. If size is set, it must contain the size
specified in the input file, and the initial grid defined in that
file is centered in the larger resulting area.

If a filename is not specified, the size value defaults to
"320x240" (used for a randomly generated initial grid).

stitch
If set to 1, stitch the left and right grid edges together, and the
top and bottom edges also. Defaults to 1.

mold
Set cell mold speed. If set, a dead cell will go from death_color
to mold_color with a step of mold. mold can have a value from 0 to
255.

life_color
Set the color of living (or new born) cells.

death_color
Set the color of dead cells. If mold is set, this is the first
color used to represent a dead cell.

mold_color
Set mold color, for definitely dead and moldy cells.

For the syntax of these 3 color options, check the "Color" section
in the ffmpeg-utils manual.

Examples

· Read a grid from pattern, and center it on a grid of size 300x300
pixels:

life=f=pattern:s=300x300

· Generate a random grid of size 200x200, with a fill ratio of 2/3:

life=ratio=2/3:s=200x200

· Specify a custom rule for evolving a randomly generated grid:

life=rule=S14/B34

· Full example with slow death effect (mold) using ffplay:

ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16

allrgb, allyuv, color, haldclutsrc, nullsrc, rgbtestsrc, smptebars,
smptehdbars, testsrc
The "allrgb" source returns frames of size 4096x4096 of all rgb colors.

The "allyuv" source returns frames of size 4096x4096 of all yuv colors.

The "color" source provides an uniformly colored input.

The "haldclutsrc" source provides an identity Hald CLUT. See also
haldclut filter.

The "nullsrc" source returns unprocessed video frames. It is mainly
useful to be employed in analysis / debugging tools, or as the source
for filters which ignore the input data.

The "rgbtestsrc" source generates an RGB test pattern useful for
detecting RGB vs BGR issues. You should see a red, green and blue
stripe from top to bottom.

The "smptebars" source generates a color bars pattern, based on the
SMPTE Engineering Guideline EG 1-1990.

The "smptehdbars" source generates a color bars pattern, based on the
SMPTE RP 219-2002.

The "testsrc" source generates a test video pattern, showing a color
pattern, a scrolling gradient and a timestamp. This is mainly intended
for testing purposes.

The sources accept the following parameters:

color, c
Specify the color of the source, only available in the "color"
source. For the syntax of this option, check the "Color" section in
the ffmpeg-utils manual.

level
Specify the level of the Hald CLUT, only available in the
"haldclutsrc" source. A level of "N" generates a picture of "N*N*N"
by "N*N*N" pixels to be used as identity matrix for 3D lookup
tables. Each component is coded on a "1/(N*N)" scale.

size, s
Specify the size of the sourced video. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual.
The default value is "320x240".

This option is not available with the "haldclutsrc" filter.

sar Set the sample aspect ratio of the sourced video.

duration, d
Set the duration of the sourced video. See the Time duration
section in the ffmpeg-utils(1) manual for the accepted syntax.

If not specified, or the expressed duration is negative, the video
is supposed to be generated forever.

decimals, n
Set the number of decimals to show in the timestamp, only available
in the "testsrc" source.

The displayed timestamp value will correspond to the original
timestamp value multiplied by the power of 10 of the specified
value. Default value is 0.

For example the following:

testsrc=duration=5.3:size=qcif:rate=10

will generate a video with a duration of 5.3 seconds, with size 176x144
and a frame rate of 10 frames per second.

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.

color=c=red@0.2:s=qcif:r=10

If the input content is to be ignored, "nullsrc" can be used. The
following command generates noise in the luminance plane by employing
the "geq" filter:

nullsrc=s=256x256, geq=random(1)*255:128:128

Commands

The "color" source supports the following commands:

c, color
Set the color of the created image. Accepts the same syntax of the
corresponding color option.

VIDEO SINKS
Below is a description of the currently available video sinks.

buffersink
Buffer video frames, and make them available to the end of the filter
graph.

This sink is mainly intended for programmatic use, in particular
through the interface defined in libavfilter/buffersink.h or the
options system.

It accepts a pointer to an AVBufferSinkContext structure, which defines
the incoming buffers' formats, to be passed as the opaque parameter to
"avfilter_init_filter" for initialization.

nullsink
Null video sink: do absolutely nothing with the input video. It is
mainly useful as a template and for use in analysis / debugging tools.

MULTIMEDIA FILTERS
Below is a description of the currently available multimedia filters.

aphasemeter
Convert input audio to a video output, displaying the audio phase.

The filter accepts the following options:

rate, r
Set the output frame rate. Default value is 25.

size, s
Set the video size for the output. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual. Default
value is "800x400".

rc
gc
bc Specify the red, green, blue contrast. Default values are 2, 7 and
1. Allowed range is "[0, 255]".

mpc Set color which will be used for drawing median phase. If color is
"none" which is default, no median phase value will be drawn.

The filter also exports the frame metadata "lavfi.aphasemeter.phase"
which represents mean phase of current audio frame. Value is in range
"[-1, 1]". The "-1" means left and right channels are completely out
of phase and 1 means channels are in phase.

avectorscope
Convert input audio to a video output, representing the audio vector
scope.

The filter is used to measure the difference between channels of stereo
audio stream. A monoaural signal, consisting of identical left and
right signal, results in straight vertical line. Any stereo separation
is visible as a deviation from this line, creating a Lissajous figure.
If the straight (or deviation from it) but horizontal line appears this
indicates that the left and right channels are out of phase.

The filter accepts the following options:

mode, m
Set the vectorscope mode.

Available values are:

lissajous
Lissajous rotated by 45 degrees.

lissajous_xy
Same as above but not rotated.

polar
Shape resembling half of circle.

Default value is lissajous.

size, s
Set the video size for the output. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual. Default
value is "400x400".

rate, r
Set the output frame rate. Default value is 25.

rc
gc
bc
ac Specify the red, green, blue and alpha contrast. Default values are
40, 160, 80 and 255. Allowed range is "[0, 255]".

rf
gf
bf
af Specify the red, green, blue and alpha fade. Default values are 15,
10, 5 and 5. Allowed range is "[0, 255]".

zoom
Set the zoom factor. Default value is 1. Allowed range is "[1,
10]".

Examples

· Complete example using ffplay:

ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
[a] avectorscope=zoom=1.3:rc=2:gc=200:bc=10:rf=1:gf=8:bf=7 [out0]'

concat
Concatenate audio and video streams, joining them together one after
the other.

The filter works on segments of synchronized video and audio streams.
All segments must have the same number of streams of each type, and
that will also be the number of streams at output.

The filter accepts the following options:

n Set the number of segments. Default is 2.

v Set the number of output video streams, that is also the number of
video streams in each segment. Default is 1.

a Set the number of output audio streams, that is also the number of
audio streams in each segment. Default is 0.

unsafe
Activate unsafe mode: do not fail if segments have a different
format.

The filter has v+a outputs: first v video outputs, then a audio
outputs.

There are nx(v+a) inputs: first the inputs for the first segment, in
the same order as the outputs, then the inputs for the second segment,
etc.

Related streams do not always have exactly the same duration, for
various reasons including codec frame size or sloppy authoring. For
that reason, related synchronized streams (e.g. a video and its audio
track) should be concatenated at once. The concat filter will use the
duration of the longest stream in each segment (except the last one),
and if necessary pad shorter audio streams with silence.

For this filter to work correctly, all segments must start at timestamp
0.

All corresponding streams must have the same parameters in all
segments; the filtering system will automatically select a common pixel
format for video streams, and a common sample format, sample rate and
channel layout for audio streams, but other settings, such as
resolution, must be converted explicitly by the user.

Different frame rates are acceptable but will result in variable frame
rate at output; be sure to configure the output file to handle it.

Examples

· Concatenate an opening, an episode and an ending, all in bilingual
version (video in stream 0, audio in streams 1 and 2):

ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \
'[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2]
concat=n=3:v=1:a=2 [v] [a1] [a2]' \
-map '[v]' -map '[a1]' -map '[a2]' output.mkv

· Concatenate two parts, handling audio and video separately, using
the (a)movie sources, and adjusting the resolution:

movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ;
movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ;
[v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa]

Note that a desync will happen at the stitch if the audio and video
streams do not have exactly the same duration in the first file.

ebur128
EBU R128 scanner filter. This filter takes an audio stream as input and
outputs it unchanged. By default, it logs a message at a frequency of
10Hz with the Momentary loudness (identified by "M"), Short-term
loudness ("S"), Integrated loudness ("I") and Loudness Range ("LRA").

The filter also has a video output (see the video option) with a real
time graph to observe the loudness evolution. The graphic contains the
logged message mentioned above, so it is not printed anymore when this
option is set, unless the verbose logging is set. The main graphing
area contains the short-term loudness (3 seconds of analysis), and the
gauge on the right is for the momentary loudness (400 milliseconds).

More information about the Loudness Recommendation EBU R128 on
<http://tech.ebu.ch/loudness>.

The filter accepts the following options:

video
Activate the video output. The audio stream is passed unchanged
whether this option is set or no. The video stream will be the
first output stream if activated. Default is 0.

size
Set the video size. This option is for video only. For the syntax
of this option, check the "Video size" section in the ffmpeg-utils
manual. Default and minimum resolution is "640x480".

meter
Set the EBU scale meter. Default is 9. Common values are 9 and 18,
respectively for EBU scale meter +9 and EBU scale meter +18. Any
other integer value between this range is allowed.

metadata
Set metadata injection. If set to 1, the audio input will be
segmented into 100ms output frames, each of them containing various
loudness information in metadata. All the metadata keys are
prefixed with "lavfi.r128.".

Default is 0.

framelog
Force the frame logging level.

Available values are:

info
information logging level

verbose
verbose logging level

By default, the logging level is set to info. If the video or the
metadata options are set, it switches to verbose.

peak
Set peak mode(s).

Available modes can be cumulated (the option is a "flag" type).
Possible values are:

none
Disable any peak mode (default).

sample
Enable sample-peak mode.

Simple peak mode looking for the higher sample value. It logs a
message for sample-peak (identified by "SPK").

true
Enable true-peak mode.

If enabled, the peak lookup is done on an over-sampled version
of the input stream for better peak accuracy. It logs a message
for true-peak. (identified by "TPK") and true-peak per frame
(identified by "FTPK"). This mode requires a build with
"libswresample".

Examples

· Real-time graph using ffplay, with a EBU scale meter +18:

ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]"

· Run an analysis with ffmpeg:

ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null -

interleave, ainterleave
Temporally interleave frames from several inputs.

"interleave" works with video inputs, "ainterleave" with audio.

These filters read frames from several inputs and send the oldest
queued frame to the output.

Input streams must have a well defined, monotonically increasing frame
timestamp values.

In order to submit one frame to output, these filters need to enqueue
at least one frame for each input, so they cannot work in case one
input is not yet terminated and will not receive incoming frames.

For example consider the case when one input is a "select" filter which
always drop input frames. The "interleave" filter will keep reading
from that input, but it will never be able to send new frames to output
until the input will send an end-of-stream signal.

Also, depending on inputs synchronization, the filters will drop frames
in case one input receives more frames than the other ones, and the
queue is already filled.

These filters accept the following options:

nb_inputs, n
Set the number of different inputs, it is 2 by default.

Examples

· Interleave frames belonging to different streams using ffmpeg:

ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi

· Add flickering blur effect:

select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave

perms, aperms
Set read/write permissions for the output frames.

These filters are mainly aimed at developers to test direct path in the
following filter in the filtergraph.

The filters accept the following options:

mode
Select the permissions mode.

It accepts the following values:

none
Do nothing. This is the default.

ro Set all the output frames read-only.

rw Set all the output frames directly writable.

toggle
Make the frame read-only if writable, and writable if read-
only.

random
Set each output frame read-only or writable randomly.

seed
Set the seed for the random mode, must be an integer included
between 0 and "UINT32_MAX". If not specified, or if explicitly set
to "-1", the filter will try to use a good random seed on a best
effort basis.

Note: in case of auto-inserted filter between the permission filter and
the following one, the permission might not be received as expected in
that following filter. Inserting a format or aformat filter before the
perms/aperms filter can avoid this problem.

select, aselect
Select frames to pass in output.

This filter accepts the following options:

expr, e
Set expression, which is evaluated for each input frame.

If the expression is evaluated to zero, the frame is discarded.

If the evaluation result is negative or NaN, the frame is sent to
the first output; otherwise it is sent to the output with index
"ceil(val)-1", assuming that the input index starts from 0.

For example a value of 1.2 corresponds to the output with index
"ceil(1.2)-1 = 2-1 = 1", that is the second output.

outputs, n
Set the number of outputs. The output to which to send the selected
frame is based on the result of the evaluation. Default value is 1.

The expression can contain the following constants:

n The (sequential) number of the filtered frame, starting from 0.

selected_n
The (sequential) number of the selected frame, starting from 0.

prev_selected_n
The sequential number of the last selected frame. It's NAN if
undefined.

TB The timebase of the input timestamps.

pts The PTS (Presentation TimeStamp) of the filtered video frame,
expressed in TB units. It's NAN if undefined.

t The PTS of the filtered video frame, expressed in seconds. It's NAN
if undefined.

prev_pts
The PTS of the previously filtered video frame. It's NAN if
undefined.

prev_selected_pts
The PTS of the last previously filtered video frame. It's NAN if
undefined.

prev_selected_t
The PTS of the last previously selected video frame. It's NAN if
undefined.

start_pts
The PTS of the first video frame in the video. It's NAN if
undefined.

start_t
The time of the first video frame in the video. It's NAN if
undefined.

pict_type (video only)
The type of the filtered frame. It can assume one of the following
values:

I
P
B
S
SI
SP
BI
interlace_type (video only)
The frame interlace type. It can assume one of the following
values:

PROGRESSIVE
The frame is progressive (not interlaced).

TOPFIRST
The frame is top-field-first.

BOTTOMFIRST
The frame is bottom-field-first.

consumed_sample_n (audio only)
the number of selected samples before the current frame

samples_n (audio only)
the number of samples in the current frame

sample_rate (audio only)
the input sample rate

key This is 1 if the filtered frame is a key-frame, 0 otherwise.

pos the position in the file of the filtered frame, -1 if the
information is not available (e.g. for synthetic video)

scene (video only)
value between 0 and 1 to indicate a new scene; a low value reflects
a low probability for the current frame to introduce a new scene,
while a higher value means the current frame is more likely to be
one (see the example below)

The default value of the select expression is "1".

Examples

· Select all frames in input:

select

The example above is the same as:

select=1

· Skip all frames:

select=0

· Select only I-frames:

select='eq(pict_type\,I)'

· Select one frame every 100:

select='not(mod(n\,100))'

· Select only frames contained in the 10-20 time interval:

select=between(t\,10\,20)

· Select only I frames contained in the 10-20 time interval:

select=between(t\,10\,20)*eq(pict_type\,I)

· Select frames with a minimum distance of 10 seconds:

select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)'

· Use aselect to select only audio frames with samples number > 100:

aselect='gt(samples_n\,100)'

· Create a mosaic of the first scenes:

ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png

Comparing scene against a value between 0.3 and 0.5 is generally a
sane choice.

· Send even and odd frames to separate outputs, and compose them:

select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] overlay=y=h

sendcmd, asendcmd
Send commands to filters in the filtergraph.

These filters read commands to be sent to other filters in the
filtergraph.

"sendcmd" must be inserted between two video filters, "asendcmd" must
be inserted between two audio filters, but apart from that they act the
same way.

The specification of commands can be provided in the filter arguments
with the commands option, or in a file specified by the filename
option.

These filters accept the following options:

commands, c
Set the commands to be read and sent to the other filters.

filename, f
Set the filename of the commands to be read and sent to the other
filters.

Commands syntax

A commands description consists of a sequence of interval
specifications, comprising a list of commands to be executed when a
particular event related to that interval occurs. The occurring event
is typically the current frame time entering or leaving a given time
interval.

An interval is specified by the following syntax:

<START>[-<END>] <COMMANDS>;

The time interval is specified by the START and END times. END is
optional and defaults to the maximum time.

The current frame time is considered within the specified interval if
it is included in the interval [START, END), that is when the time is
greater or equal to START and is lesser than END.

COMMANDS consists of a sequence of one or more command specifications,
separated by ",", relating to that interval. The syntax of a command
specification is given by:

[<FLAGS>] <TARGET> <COMMAND> <ARG>

FLAGS is optional and specifies the type of events relating to the time
interval which enable sending the specified command, and must be a non-
null sequence of identifier flags separated by "+" or "|" and enclosed
between "[" and "]".

The following flags are recognized:

enter
The command is sent when the current frame timestamp enters the
specified interval. In other words, the command is sent when the
previous frame timestamp was not in the given interval, and the
current is.

leave
The command is sent when the current frame timestamp leaves the
specified interval. In other words, the command is sent when the
previous frame timestamp was in the given interval, and the current
is not.

If FLAGS is not specified, a default value of "[enter]" is assumed.

TARGET specifies the target of the command, usually the name of the
filter class or a specific filter instance name.

COMMAND specifies the name of the command for the target filter.

ARG is optional and specifies the optional list of argument for the
given COMMAND.

Between one interval specification and another, whitespaces, or
sequences of characters starting with "#" until the end of line, are
ignored and can be used to annotate comments.

A simplified BNF description of the commands specification syntax
follows:

<COMMAND_FLAG> ::= "enter" | "leave"
<COMMAND_FLAGS> ::= <COMMAND_FLAG> [(+|"|")<COMMAND_FLAG>]
<COMMAND> ::= ["[" <COMMAND_FLAGS> "]"] <TARGET> <COMMAND> [<ARG>]
<COMMANDS> ::= <COMMAND> [,<COMMANDS>]
<INTERVAL> ::= <START>[-<END>] <COMMANDS>
<INTERVALS> ::= <INTERVAL>[;<INTERVALS>]

Examples

· Specify audio tempo change at second 4:

asendcmd=c='4.0 atempo tempo 1.5',atempo

· Specify a list of drawtext and hue commands in a file.

# show text in the interval 5-10
5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world',
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=';

# desaturate the image in the interval 15-20
15.0-20.0 [enter] hue s 0,
[enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor',
[leave] hue s 1,
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color';

# apply an exponential saturation fade-out effect, starting from time 25
25 [enter] hue s exp(25-t)

A filtergraph allowing to read and process the above command list
stored in a file test.cmd, can be specified with:

sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue

setpts, asetpts
Change the PTS (presentation timestamp) of the input frames.

"setpts" works on video frames, "asetpts" on audio frames.

This filter accepts the following options:

expr
The expression which is evaluated for each frame to construct its
timestamp.

The expression is evaluated through the eval API and can contain the
following constants:

FRAME_RATE
frame rate, only defined for constant frame-rate video

PTS The presentation timestamp in input

N The count of the input frame for video or the number of consumed
samples, not including the current frame for audio, starting from
0.

NB_CONSUMED_SAMPLES
The number of consumed samples, not including the current frame
(only audio)

NB_SAMPLES, S
The number of samples in the current frame (only audio)

SAMPLE_RATE, SR
The audio sample rate.

STARTPTS
The PTS of the first frame.

STARTT
the time in seconds of the first frame

INTERLACED
State whether the current frame is interlaced.

T the time in seconds of the current frame

POS original position in the file of the frame, or undefined if
undefined for the current frame

PREV_INPTS
The previous input PTS.

PREV_INT
previous input time in seconds

PREV_OUTPTS
The previous output PTS.

PREV_OUTT
previous output time in seconds

RTCTIME
The wallclock (RTC) time in microseconds. This is deprecated, use
time(0) instead.

RTCSTART
The wallclock (RTC) time at the start of the movie in microseconds.

TB The timebase of the input timestamps.

Examples

· Start counting PTS from zero

setpts=PTS-STARTPTS

· Apply fast motion effect:

setpts=0.5*PTS

· Apply slow motion effect:

setpts=2.0*PTS

· Set fixed rate of 25 frames per second:

setpts=N/(25*TB)

· Set fixed rate 25 fps with some jitter:

setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'

· Apply an offset of 10 seconds to the input PTS:

setpts=PTS+10/TB

· Generate timestamps from a "live source" and rebase onto the
current timebase:

setpts='(RTCTIME - RTCSTART) / (TB * 1000000)'

· Generate timestamps by counting samples:

asetpts=N/SR/TB

settb, asettb
Set the timebase to use for the output frames timestamps. It is mainly
useful for testing timebase configuration.

It accepts the following parameters:

expr, tb
The expression which is evaluated into the output timebase.

The value for tb is an arithmetic expression representing a rational.
The expression can contain the constants "AVTB" (the default timebase),
"intb" (the input timebase) and "sr" (the sample rate, audio only).
Default value is "intb".

Examples

· Set the timebase to 1/25:

settb=expr=1/25

· Set the timebase to 1/10:

settb=expr=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

showcqt
Convert input audio to a video output representing frequency spectrum
logarithmically (using constant Q transform with Brown-Puckette
algorithm), with musical tone scale, from E0 to D#10 (10 octaves).

The filter accepts the following options:

volume
Specify transform volume (multiplier) expression. The expression
can contain variables:

frequency, freq, f
the frequency where transform is evaluated

timeclamp, tc
value of timeclamp option

and functions:

a_weighting(f)
A-weighting of equal loudness

b_weighting(f)
B-weighting of equal loudness

c_weighting(f)
C-weighting of equal loudness

Default value is 16.

tlength
Specify transform length expression. The expression can contain
variables:

frequency, freq, f
the frequency where transform is evaluated

timeclamp, tc
value of timeclamp option

Default value is "384/f*tc/(384/f+tc)".

timeclamp
Specify the transform timeclamp. At low frequency, there is trade-
off between accuracy in time domain and frequency domain. If
timeclamp is lower, event in time domain is represented more
accurately (such as fast bass drum), otherwise event in frequency
domain is represented more accurately (such as bass guitar).
Acceptable value is [0.1, 1.0]. Default value is 0.17.

coeffclamp
Specify the transform coeffclamp. If coeffclamp is lower, transform
is more accurate, otherwise transform is faster. Acceptable value
is [0.1, 10.0]. Default value is 1.0.

gamma
Specify gamma. Lower gamma makes the spectrum more contrast, higher
gamma makes the spectrum having more range. Acceptable value is
[1.0, 7.0]. Default value is 3.0.

gamma2
Specify gamma of bargraph. Acceptable value is [1.0, 7.0]. Default
value is 1.0.

fontfile
Specify font file for use with freetype. If not specified, use
embedded font.

fontcolor
Specify font color expression. This is arithmetic expression that
should return integer value 0xRRGGBB. The expression can contain
variables:

frequency, freq, f
the frequency where transform is evaluated

timeclamp, tc
value of timeclamp option

and functions:

midi(f)
midi number of frequency f, some midi numbers: E0(16), C1(24),
C2(36), A4(69)

r(x), g(x), b(x)
red, green, and blue value of intensity x

Default value is "st(0, (midi(f)-59.5)/12); st(1,
if(between(ld(0),0,1), 0.5-0.5*cos(2*PI*ld(0)), 0)); r(1-ld(1)) +
b(ld(1))"

fullhd
If set to 1 (the default), the video size is 1920x1080 (full HD),
if set to 0, the video size is 960x540. Use this option to make CPU
usage lower.

fps Specify video fps. Default value is 25.

count
Specify number of transform per frame, so there are fps*count
transforms per second. Note that audio data rate must be divisible
by fps*count. Default value is 6.

Examples

· Playing audio while showing the spectrum:

ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]'

· Same as above, but with frame rate 30 fps:

ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=fps=30:count=5 [out0]'

· Playing at 960x540 and lower CPU usage:

ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=fullhd=0:count=3 [out0]'

· A1 and its harmonics: A1, A2, (near)E3, A3:

ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
asplit[a][out1]; [a] showcqt [out0]'

· Same as above, but with more accuracy in frequency domain (and
slower):

ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
asplit[a][out1]; [a] showcqt=timeclamp=0.5 [out0]'

· B-weighting of equal loudness

volume=16*b_weighting(f)

· Lower Q factor

tlength=100/f*tc/(100/f+tc)

· Custom fontcolor, C-note is colored green, others are colored blue

fontcolor='if(mod(floor(midi(f)+0.5),12), 0x0000FF, g(1))'

· Custom gamma, now spectrum is linear to the amplitude.

gamma=2:gamma2=2

showfreqs
Convert input audio to video output representing the audio power
spectrum. Audio amplitude is on Y-axis while frequency is on X-axis.

The filter accepts the following options:

size, s
Specify size of video. For the syntax of this option, check the
"Video size" section in the ffmpeg-utils manual. Default is
"1024x512".

mode
Set display mode. This set how each frequency bin will be
represented.

It accepts the following values:

line
bar
dot

Default is "bar".

ascale
Set amplitude scale.

It accepts the following values:

lin Linear scale.

sqrt
Square root scale.

cbrt
Cubic root scale.

log Logarithmic scale.

Default is "log".

fscale
Set frequency scale.

It accepts the following values:

lin Linear scale.

log Logarithmic scale.

rlog
Reverse logarithmic scale.

Default is "lin".

win_size
Set window size.

It accepts the following values:

w16
w32
w64
w128
w256
w512
w1024
w2048
w4096
w8192
w16384
w32768
w65536

Default is "w2048"

win_func
Set windowing function.

It accepts the following values:

rect
bartlett
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall

Default is "hanning".

overlap
Set window overlap. In range "[0, 1]". Default is 1, which means
optimal overlap for selected window function will be picked.

averaging
Set time averaging. Setting this to 0 will display current maximal
peaks. Default is 1, which means time averaging is disabled.

color
Specify list of colors separated by space or by '|' which will be
used to draw channel frequencies. Unrecognized or missing colors
will be replaced by white color.

showspectrum
Convert input audio to a video output, representing the audio frequency
spectrum.

The filter accepts the following options:

size, s
Specify the video size for the output. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual.
Default value is "640x512".

slide
Specify how the spectrum should slide along the window.

It accepts the following values:

replace
the samples start again on the left when they reach the right

scroll
the samples scroll from right to left

fullframe
frames are only produced when the samples reach the right

Default value is "replace".

mode
Specify display mode.

It accepts the following values:

combined
all channels are displayed in the same row

separate
all channels are displayed in separate rows

Default value is combined.

color
Specify display color mode.

It accepts the following values:

channel
each channel is displayed in a separate color

intensity
each channel is is displayed using the same color scheme

Default value is channel.

scale
Specify scale used for calculating intensity color values.

It accepts the following values:

lin linear

sqrt
square root, default

cbrt
cubic root

log logarithmic

Default value is sqrt.

saturation
Set saturation modifier for displayed colors. Negative values
provide alternative color scheme. 0 is no saturation at all.
Saturation must be in [-10.0, 10.0] range. Default value is 1.

win_func
Set window function.

It accepts the following values:

none
No samples pre-processing (do not expect this to be faster)

hann
Hann window

hamming
Hamming window

blackman
Blackman window

Default value is "hann".

The usage is very similar to the showwaves filter; see the examples in
that section.

Examples

· Large window with logarithmic color scaling:

showspectrum=s=1280x480:scale=log

· Complete example for a colored and sliding spectrum per channel
using ffplay:

ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
[a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]'

showvolume
Convert input audio volume to a video output.

The filter accepts the following options:

rate, r
Set video rate.

b Set border width, allowed range is [0, 5]. Default is 1.

w Set channel width, allowed range is [40, 1080]. Default is 400.

h Set channel height, allowed range is [1, 100]. Default is 20.

f Set fade, allowed range is [1, 255]. Default is 20.

c Set volume color expression.

The expression can use the following variables:

VOLUME
Current max volume of channel in dB.

CHANNEL
Current channel number, starting from 0.

t If set, displays channel names. Default is enabled.

showwaves
Convert input audio to a video output, representing the samples waves.

The filter accepts the following options:

size, s
Specify the video size for the output. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual.
Default value is "600x240".

mode
Set display mode.

Available values are:

point
Draw a point for each sample.

line
Draw a vertical line for each sample.

p2p Draw a point for each sample and a line between them.

cline
Draw a centered vertical line for each sample.

Default value is "point".

n Set the number of samples which are printed on the same column. A
larger value will decrease the frame rate. Must be a positive
integer. This option can be set only if the value for rate is not
explicitly specified.

rate, r
Set the (approximate) output frame rate. This is done by setting
the option n. Default value is "25".

split_channels
Set if channels should be drawn separately or overlap. Default
value is 0.

Examples

· Output the input file audio and the corresponding video
representation at the same time:

amovie=a.mp3,asplit[out0],showwaves[out1]

· Create a synthetic signal and show it with showwaves, forcing a
frame rate of 30 frames per second:

aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1]

showwavespic
Convert input audio to a single video frame, representing the samples
waves.

The filter accepts the following options:

size, s
Specify the video size for the output. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual.
Default value is "600x240".

split_channels
Set if channels should be drawn separately or overlap. Default
value is 0.

Examples

· Extract a channel split representation of the wave form of a whole
audio track in a 1024x800 picture using ffmpeg:

ffmpeg -i audio.flac -lavfi showwavespic=split_channels=1:s=1024x800 waveform.png

split, asplit
Split input into several identical outputs.

"asplit" works with audio input, "split" with video.

The filter accepts a single parameter which specifies the number of
outputs. If unspecified, it defaults to 2.

Examples

· Create two separate outputs from the same input:

[in] split [out0][out1]

· To create 3 or more outputs, you need to specify the number of
outputs, like in:

[in] asplit=3 [out0][out1][out2]

· Create two separate outputs from the same input, one cropped and
one padded:

[in] split [splitout1][splitout2];
[splitout1] crop=100:100:0:0 [cropout];
[splitout2] pad=200:200:100:100 [padout];

· Create 5 copies of the input audio with ffmpeg:

ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT

zmq, azmq
Receive commands sent through a libzmq client, and forward them to
filters in the filtergraph.

"zmq" and "azmq" work as a pass-through filters. "zmq" must be inserted
between two video filters, "azmq" between two audio filters.

To enable these filters you need to install the libzmq library and
headers and configure FFmpeg with "--enable-libzmq".

For more information about libzmq see: <http://www.zeromq.org/>

The "zmq" and "azmq" filters work as a libzmq server, which receives
messages sent through a network interface defined by the bind_address
option.

The received message must be in the form:

<TARGET> <COMMAND> [<ARG>]

TARGET specifies the target of the command, usually the name of the
filter class or a specific filter instance name.

COMMAND specifies the name of the command for the target filter.

ARG is optional and specifies the optional argument list for the given
COMMAND.

Upon reception, the message is processed and the corresponding command
is injected into the filtergraph. Depending on the result, the filter
will send a reply to the client, adopting the format:

<ERROR_CODE> <ERROR_REASON>
<MESSAGE>

MESSAGE is optional.

Examples

Look at tools/zmqsend for an example of a zmq client which can be used
to send commands processed by these filters.

Consider the following filtergraph generated by ffplay

ffplay -dumpgraph 1 -f lavfi "
color=s=100x100:c=red [l];
color=s=100x100:c=blue [r];
nullsrc=s=200x100, zmq [bg];
[bg][l] overlay [bg+l];
[bg+l][r] overlay=x=100 "

To change the color of the left side of the video, the following
command can be used:

echo Parsed_color_0 c yellow | tools/zmqsend

To change the right side:

echo Parsed_color_1 c pink | tools/zmqsend

MULTIMEDIA SOURCES
Below is a description of the currently available multimedia sources.

amovie
This is the same as movie source, except it selects an audio stream by
default.

movie
Read audio and/or video stream(s) from a movie container.

It accepts the following parameters:

filename
The name of the resource to read (not necessarily a file; it can
also be a device or a stream accessed through some protocol).

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. The default value is "0".

streams, s
Specifies the streams to read. Several streams can be specified,
separated by "+". The source will then have as many outputs, in the
same order. The syntax is explained in the ``Stream specifiers''
section in the ffmpeg manual. Two special names, "dv" and "da"
specify respectively the default (best suited) video and audio
stream. Default is "dv", or "da" if the filter is called as
"amovie".

stream_index, si
Specifies the index of the video stream to read. If the value is
-1, the most suitable video stream will be automatically selected.
The default value is "-1". Deprecated. If the filter is called
"amovie", it will select audio instead of video.

loop
Specifies how many times to read the stream in sequence. If the
value is less than 1, the stream will be read again and again.
Default value is "1".

Note that when the movie is looped the source timestamps are not
changed, so it will generate non monotonically increasing
timestamps.

It allows overlaying a second video on top of the main input of a
filtergraph, as shown in this graph:

input -----------> deltapts0 --> overlay --> output
^
|
movie --> scale--> deltapts1 -------+

Examples

· Skip 3.2 seconds from the start of the AVI file in.avi, and overlay
it on top of the input labelled "in":

movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [over];
[in] setpts=PTS-STARTPTS [main];
[main][over] overlay=16:16 [out]

· Read from a video4linux2 device, and overlay it on top of the input
labelled "in":

movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [over];
[in] setpts=PTS-STARTPTS [main];
[main][over] overlay=16:16 [out]

· Read the first video stream and the audio stream with id 0x81 from
dvd.vob; the video is connected to the pad named "video" and the
audio is connected to the pad named "audio":

movie=dvd.vob:s=v:0+#0x81 [video] [audio]

SEE ALSO
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), ffmpeg-utils(1),
ffmpeg-scaler(1), ffmpeg-resampler(1), ffmpeg-codecs(1),
ffmpeg-bitstream-filters(1), ffmpeg-formats(1), ffmpeg-devices(1),
ffmpeg-protocols(1), ffmpeg-filters(1)

AUTHORS
The FFmpeg developers.

For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
the FFmpeg source directory, or browsing the online repository at
<http://source.ffmpeg.org>.

Maintainers for the specific components are listed in the file
MAINTAINERS in the source code tree.

FFMPEG-ALL(1)

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