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FFMPEG-ALL(1) FFMPEG-ALL(1)
NAME
ffmpeg - ffmpeg video converter
SYNOPSIS
ffmpeg [global_options] {[input_file_options] -i input_url} ...
{[output_file_options] output_url} ...
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 url. Anything
found on the command line which cannot be interpreted as an option is
considered to be an output url.
Each input or output url 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.
路 To set the video bitrate of the output file to 64 kbit/s:
ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi
路 To force the frame rate of the output file to 24 fps:
ffmpeg -i input.avi -r 24 output.avi
路 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/-dn"
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).
-demuxers
Show available demuxers.
-muxers
Show available muxers.
-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 input 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 an assertion failure. This is not currently used for
anything.
fatal, 8
Only show fatal errors. These are errors after which the
process absolutely cannot continue.
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 future 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 url (input)
input file url
-y (global)
Overwrite output files without asking.
-n (global)
Do not overwrite output files, and exit immediately if a specified
output file already exists.
-stream_loop number (input)
Set number of times input stream shall be looped. Loop 0 means no
loop, loop -1 means infinite loop.
-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 url), 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. No further chunk of
bytes is written after the limit is exceeded. The size of the
output file is slightly more than the requested file size.
-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 url), 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, chapters or programs. 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
-disposition[:stream_specifier] value (output,per-stream)
Sets the disposition for a stream.
This option overrides the disposition copied from the input stream.
It is also possible to delete the disposition by setting it to 0.
The following dispositions are recognized:
default
dub
original
comment
lyrics
karaoke
forced
hearing_impaired
visual_impaired
clean_effects
captions
descriptions
metadata
For example, to make the second audio stream the default stream:
ffmpeg -i in.mkv -disposition:a:1 default out.mkv
To make the second subtitle stream the default stream and remove
the default disposition from the first subtitle stream:
ffmpeg -i INPUT -disposition:s:0 0 -disposition:s:1 default OUTPUT
-program
[title=title:][program_num=program_num:]st=stream[:st=stream...]
(output)
Creates a program with the specified title, program_num and adds
the specified stream(s) to it.
-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 obsolete alias
for "-frames:d", which you should use instead.
-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.
-filter_threads nb_threads (global)
Defines how many threads are used to process a filter pipeline.
Each pipeline will produce a thread pool with this many threads
available for parallel processing. The default is the number of
available CPUs.
-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 option "-fdebug ts".
-attach filename (output)
Add an attachment to the output file. This is supported by a few
formats like Matroska for e.g. fonts used in rendering subtitles.
Attachments are implemented as a specific type of stream, so this
option will add a new stream to the file. It is then possible to
use per-stream options on this stream in the usual way. Attachment
streams created with this option will be created after all the
other streams (i.e. those created with "-map" or automatic
mappings).
Note that for Matroska you also have to set the mimetype metadata
tag:
ffmpeg -i INPUT -attach DejaVuSans.ttf -metadata:s:2 mimetype=application/x-truetype-font out.mkv
(assuming that the attachment stream will be third in the output
file).
-dump_attachment[:stream_specifier] filename (input,per-stream)
Extract the matching attachment stream into a file named filename.
If filename is empty, then the value of the "filename" metadata tag
will be used.
E.g. to extract the first attachment to a file named 'out.ttf':
ffmpeg -dump_attachment:t:0 out.ttf -i INPUT
To extract all attachments to files determined by the "filename"
tag:
ffmpeg -dump_attachment:t "" -i INPUT
Technical note -- attachments are implemented as codec extradata,
so this option can actually be used to extract extradata from any
stream, not just attachments.
-noautorotate
Disable automatically rotating video based on file metadata.
Video Options
-vframes number (output)
Set the number of video frames to output. This is an obsolete alias
for "-frames:v", which you should use instead.
-r[:stream_specifier] fps (input/output,per-stream)
Set frame rate (Hz value, fraction or abbreviation).
As an input option, ignore any timestamps stored in the file and
instead generate timestamps assuming constant frame rate fps. This
is not the same as the -framerate option used for some input
formats like image2 or v4l2 (it used to be the same in older
versions of FFmpeg). If in doubt use -framerate instead of the
input option -r.
As an output option, duplicate or drop input frames to achieve
constant output frame rate fps.
-s[:stream_specifier] size (input/output,per-stream)
Set frame size.
As an input option, this is a shortcut for the video_size private
option, recognized by some demuxers for which the frame size is
either not stored in the file or is configurable -- e.g. raw video
or video grabbers.
As an output option, this inserts the "scale" video filter to the
end of the corresponding filtergraph. Please use the "scale" filter
directly to insert it at the beginning or some other place.
The format is wxh (default - same as source).
-aspect[:stream_specifier] aspect (output,per-stream)
Set the video display aspect ratio specified by aspect.
aspect can be a floating point number string, or a string of the
form num:den, where num and den are the numerator and denominator
of the aspect ratio. For example "4:3", "16:9", "1.3333", and
"1.7777" are valid argument values.
If used together with -vcodec copy, it will affect the aspect ratio
stored at container level, but not the aspect ratio stored in
encoded frames, if it exists.
-vn (output)
Disable video recording.
-vcodec codec (output)
Set the video codec. This is an alias for "-codec:v".
-pass[:stream_specifier] n (output,per-stream)
Select the pass number (1 or 2). It is used to do two-pass video
encoding. The statistics of the video are recorded in the first
pass into a log file (see also the option -passlogfile), and in the
second pass that log file is used to generate the video at the
exact requested bitrate. On pass 1, you may just deactivate audio
and set output to null, examples for Windows and Unix:
ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL
ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null
-passlogfile[:stream_specifier] prefix (output,per-stream)
Set two-pass log file name prefix to prefix, the default file name
prefix is ``ffmpeg2pass''. The complete file name will be
PREFIX-N.log, where N is a number specific to the output stream
-vf filtergraph (output)
Create the filtergraph specified by filtergraph and use it to
filter the stream.
This is an alias for "-filter:v", see the -filter option.
Advanced Video options
-pix_fmt[:stream_specifier] format (input/output,per-stream)
Set pixel format. Use "-pix_fmts" to show all the supported pixel
formats. If the selected pixel format can not be selected, ffmpeg
will print a warning and select the best pixel format supported by
the encoder. If pix_fmt is prefixed by a "+", ffmpeg will exit
with an error if the requested pixel format can not be selected,
and automatic conversions inside filtergraphs are disabled. If
pix_fmt is a single "+", ffmpeg selects the same pixel format as
the input (or graph output) and automatic conversions are disabled.
-sws_flags flags (input/output)
Set SwScaler flags.
-vdt n
Discard threshold.
-rc_override[:stream_specifier] override (output,per-stream)
Rate control override for specific intervals, formatted as
"int,int,int" list separated with slashes. Two first values are the
beginning and end frame numbers, last one is quantizer to use if
positive, or quality factor if negative.
-ilme
Force interlacing support in encoder (MPEG-2 and MPEG-4 only). Use
this option if your input file is interlaced and you want to keep
the interlaced format for minimum losses. The alternative is to
deinterlace the input stream with -deinterlace, but deinterlacing
introduces losses.
-psnr
Calculate PSNR of compressed frames.
-vstats
Dump video coding statistics to vstats_HHMMSS.log.
-vstats_file file
Dump video coding statistics to file.
-vstats_version file
Specifies which version of the vstats format to use. Default is 2.
version = 1 :
"frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time=
%0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s"
version > 1:
"out= %2d st= %2d frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d
s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s"
-top[:stream_specifier] n (output,per-stream)
top=1/bottom=0/auto=-1 field first
-dc precision
Intra_dc_precision.
-vtag fourcc/tag (output)
Force video tag/fourcc. This is an alias for "-tag:v".
-qphist (global)
Show QP histogram
-vbsf bitstream_filter
Deprecated see -bsf
-force_key_frames[:stream_specifier] time[,time...] (output,per-stream)
-force_key_frames[:stream_specifier] expr:expr (output,per-stream)
Force key frames at the specified timestamps, more precisely at the
first frames after each specified time.
If the argument is prefixed with "expr:", the string expr is
interpreted like an expression and is evaluated for each frame. A
key frame is forced in case the evaluation is non-zero.
If one of the times is ""chapters"[delta]", it is expanded into the
time of the beginning of all chapters in the file, shifted by
delta, expressed as a time in seconds. This option can be useful
to ensure that a seek point is present at a chapter mark or any
other designated place in the output file.
For example, to insert a key frame at 5 minutes, plus key frames
0.1 second before the beginning of every chapter:
-force_key_frames 0:05:00,chapters-0.1
The expression in expr can contain the following constants:
n the number of current processed frame, starting from 0
n_forced
the number of forced frames
prev_forced_n
the number of the previous forced frame, it is "NAN" when no
keyframe was forced yet
prev_forced_t
the time of the previous forced frame, it is "NAN" when no
keyframe was forced yet
t the time of the current processed frame
For example to force a key frame every 5 seconds, you can specify:
-force_key_frames expr:gte(t,n_forced*5)
To force a key frame 5 seconds after the time of the last forced
one, starting from second 13:
-force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))
Note that forcing too many keyframes is very harmful for the
lookahead algorithms of certain encoders: using fixed-GOP options
or similar would be more efficient.
-copyinkf[:stream_specifier] (output,per-stream)
When doing stream copy, copy also non-key frames found at the
beginning.
-init_hw_device type[=name][:device[,key=value...]]
Initialise a new hardware device of type type called name, using
the given device parameters. If no name is specified it will
receive a default name of the form "type%d".
The meaning of device and the following arguments depends on the
device type:
cuda
device is the number of the CUDA device.
dxva2
device is the number of the Direct3D 9 display adapter.
vaapi
device is either an X11 display name or a DRM render node. If
not specified, it will attempt to open the default X11 display
($DISPLAY) and then the first DRM render node
(/dev/dri/renderD128).
vdpau
device is an X11 display name. If not specified, it will
attempt to open the default X11 display ($DISPLAY).
qsv device selects a value in MFX_IMPL_*. Allowed values are:
auto
sw
hw
auto_any
hw_any
hw2
hw3
hw4
If not specified, auto_any is used. (Note that it may be
easier to achieve the desired result for QSV by creating the
platform-appropriate subdevice (dxva2 or vaapi) and then
deriving a QSV device from that.)
-init_hw_device type[=name]@source
Initialise a new hardware device of type type called name, deriving
it from the existing device with the name source.
-init_hw_device list
List all hardware device types supported in this build of ffmpeg.
-filter_hw_device name
Pass the hardware device called name to all filters in any filter
graph. This can be used to set the device to upload to with the
"hwupload" filter, or the device to map to with the "hwmap" filter.
Other filters may also make use of this parameter when they require
a hardware device. Note that this is typically only required when
the input is not already in hardware frames - when it is, filters
will derive the device they require from the context of the frames
they receive as input.
This is a global setting, so all filters will receive the same
device.
-hwaccel[:stream_specifier] hwaccel (input,per-stream)
Use hardware acceleration to decode the matching stream(s). The
allowed values of hwaccel are:
none
Do not use any hardware acceleration (the default).
auto
Automatically select the hardware acceleration method.
vda Use Apple VDA hardware acceleration.
vdpau
Use VDPAU (Video Decode and Presentation API for Unix) hardware
acceleration.
dxva2
Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
vaapi
Use VAAPI (Video Acceleration API) hardware acceleration.
qsv Use the Intel QuickSync Video acceleration for video
transcoding.
Unlike most other values, this option does not enable
accelerated decoding (that is used automatically whenever a qsv
decoder is selected), but accelerated transcoding, without
copying the frames into the system memory.
For it to work, both the decoder and the encoder must support
QSV acceleration and no filters must be used.
This option has no effect if the selected hwaccel is not available
or not supported by the chosen decoder.
Note that most acceleration methods are intended for playback and
will not be faster than software decoding on modern CPUs.
Additionally, ffmpeg will usually need to copy the decoded frames
from the GPU memory into the system memory, resulting in further
performance loss. This option is thus mainly useful for testing.
-hwaccel_device[:stream_specifier] hwaccel_device (input,per-stream)
Select a device to use for hardware acceleration.
This option only makes sense when the -hwaccel option is also
specified. It can either refer to an existing device created with
-init_hw_device by name, or it can create a new device as if
-init_hw_device type:hwaccel_device were called immediately before.
-hwaccels
List all hardware acceleration methods supported in this build of
ffmpeg.
Audio Options
-aframes number (output)
Set the number of audio frames to output. This is an obsolete alias
for "-frames:a", which you should use instead.
-ar[:stream_specifier] freq (input/output,per-stream)
Set the audio sampling frequency. For output streams it is set by
default to the frequency of the corresponding input stream. For
input streams this option only makes sense for audio grabbing
devices and raw demuxers and is mapped to the corresponding demuxer
options.
-aq q (output)
Set the audio quality (codec-specific, VBR). This is an alias for
-q:a.
-ac[:stream_specifier] channels (input/output,per-stream)
Set the number of audio channels. For output streams it is set by
default to the number of input audio channels. For input streams
this option only makes sense for audio grabbing devices and raw
demuxers and is mapped to the corresponding demuxer options.
-an (output)
Disable audio recording.
-acodec codec (input/output)
Set the audio codec. This is an alias for "-codec:a".
-sample_fmt[:stream_specifier] sample_fmt (output,per-stream)
Set the audio sample format. Use "-sample_fmts" to get a list of
supported sample formats.
-af filtergraph (output)
Create the filtergraph specified by filtergraph and use it to
filter the stream.
This is an alias for "-filter:a", see the -filter option.
Advanced Audio options
-atag fourcc/tag (output)
Force audio tag/fourcc. This is an alias for "-tag:a".
-absf bitstream_filter
Deprecated, see -bsf
-guess_layout_max channels (input,per-stream)
If some input channel layout is not known, try to guess only if it
corresponds to at most the specified number of channels. For
example, 2 tells to ffmpeg to recognize 1 channel as mono and 2
channels as stereo but not 6 channels as 5.1. The default is to
always try to guess. Use 0 to disable all guessing.
Subtitle options
-scodec codec (input/output)
Set the subtitle codec. This is an alias for "-codec:s".
-sn (output)
Disable subtitle recording.
-sbsf bitstream_filter
Deprecated, see -bsf
Advanced Subtitle options
-fix_sub_duration
Fix subtitles durations. For each subtitle, wait for the next
packet in the same stream and adjust the duration of the first to
avoid overlap. This is necessary with some subtitles codecs,
especially DVB subtitles, because the duration in the original
packet is only a rough estimate and the end is actually marked by
an empty subtitle frame. Failing to use this option when necessary
can result in exaggerated durations or muxing failures due to non-
monotonic timestamps.
Note that this option will delay the output of all data until the
next subtitle packet is decoded: it may increase memory consumption
and latency a lot.
-canvas_size size
Set the size of the canvas used to render subtitles.
Advanced options
-map
[-]input_file_id[:stream_specifier][?][,sync_file_id[:stream_specifier]]
| [linklabel] (output)
Designate one or more input streams as a source for the output
file. Each input stream is identified by the input file index
input_file_id and the input stream index input_stream_id within the
input file. Both indices start at 0. If specified,
sync_file_id:stream_specifier sets which input stream is used as a
presentation sync reference.
The first "-map" option on the command line specifies the source
for output stream 0, the second "-map" option specifies the source
for output stream 1, etc.
A "-" character before the stream identifier creates a "negative"
mapping. It disables matching streams from already created
mappings.
A trailing "?" after the stream index will allow the map to be
optional: if the map matches no streams the map will be ignored
instead of failing. Note the map will still fail if an invalid
input file index is used; such as if the map refers to a non-
existent input.
An alternative [linklabel] form will map outputs from complex
filter graphs (see the -filter_complex option) to the output file.
linklabel must correspond to a defined output link label in the
graph.
For example, to map ALL streams from the first input file to output
ffmpeg -i INPUT -map 0 output
For example, if you have two audio streams in the first input file,
these streams are identified by "0:0" and "0:1". You can use "-map"
to select which streams to place in an output file. For example:
ffmpeg -i INPUT -map 0:1 out.wav
will map the input stream in INPUT identified by "0:1" to the
(single) output stream in out.wav.
For example, to select the stream with index 2 from input file
a.mov (specified by the identifier "0:2"), and stream with index 6
from input b.mov (specified by the identifier "1:6"), and copy them
to the output file out.mov:
ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov
To select all video and the third audio stream from an input file:
ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT
To map all the streams except the second audio, use negative
mappings
ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
To map the video and audio streams from the first input, and using
the trailing "?", ignore the audio mapping if no audio streams
exist in the first input:
ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT
To pick the English audio stream:
ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
Note that using this option disables the default mappings for this
output file.
-ignore_unknown
Ignore input streams with unknown type instead of failing if
copying such streams is attempted.
-copy_unknown
Allow input streams with unknown type to be copied instead of
failing if copying such streams is attempted.
-map_channel
[input_file_id.stream_specifier.channel_id|-1][?][:output_file_id.stream_specifier]
Map an audio channel from a given input to an output. If
output_file_id.stream_specifier is not set, the audio channel will
be mapped on all the audio streams.
Using "-1" instead of input_file_id.stream_specifier.channel_id
will map a muted channel.
A trailing "?" will allow the map_channel to be optional: if the
map_channel matches no channel the map_channel will be ignored
instead of failing.
For example, assuming INPUT is a stereo audio file, you can switch
the two audio channels with the following command:
ffmpeg -i INPUT -map_channel 0.0.1 -map_channel 0.0.0 OUTPUT
If you want to mute the first channel and keep the second:
ffmpeg -i INPUT -map_channel -1 -map_channel 0.0.1 OUTPUT
The order of the "-map_channel" option specifies the order of the
channels in the output stream. The output channel layout is guessed
from the number of channels mapped (mono if one "-map_channel",
stereo if two, etc.). Using "-ac" in combination of "-map_channel"
makes the channel gain levels to be updated if input and output
channel layouts don't match (for instance two "-map_channel"
options and "-ac 6").
You can also extract each channel of an input to specific outputs;
the following command extracts two channels of the INPUT audio
stream (file 0, stream 0) to the respective OUTPUT_CH0 and
OUTPUT_CH1 outputs:
ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1
The following example splits the channels of a stereo input into
two separate streams, which are put into the same output file:
ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg
Note that currently each output stream can only contain channels
from a single input stream; you can't for example use
"-map_channel" to pick multiple input audio channels contained in
different streams (from the same or different files) and merge them
into a single output stream. It is therefore not currently
possible, for example, to turn two separate mono streams into a
single stereo stream. However splitting a stereo stream into two
single channel mono streams is possible.
If you need this feature, a possible workaround is to use the
amerge filter. For example, if you need to merge a media (here
input.mkv) with 2 mono audio streams into one single stereo channel
audio stream (and keep the video stream), you can use the following
command:
ffmpeg -i input.mkv -filter_complex "[0:1] [0:2] amerge" -c:a pcm_s16le -c:v copy output.mkv
To map the first two audio channels from the first input, and using
the trailing "?", ignore the audio channel mapping if the first
input is mono instead of stereo:
ffmpeg -i INPUT -map_channel 0.0.0 -map_channel 0.0.1? OUTPUT
-map_metadata[:metadata_spec_out] infile[:metadata_spec_in]
(output,per-metadata)
Set metadata information of the next output file from infile. Note
that those are file indices (zero-based), not filenames. Optional
metadata_spec_in/out parameters specify, which metadata to copy. A
metadata specifier can have the following forms:
g global metadata, i.e. metadata that applies to the whole file
s[:stream_spec]
per-stream metadata. stream_spec is a stream specifier as
described in the Stream specifiers chapter. In an input
metadata specifier, the first matching stream is copied from.
In an output metadata specifier, all matching streams are
copied to.
c:chapter_index
per-chapter metadata. chapter_index is the zero-based chapter
index.
p:program_index
per-program metadata. program_index is the zero-based program
index.
If metadata specifier is omitted, it defaults to global.
By default, global metadata is copied from the first input file,
per-stream and per-chapter metadata is copied along with
streams/chapters. These default mappings are disabled by creating
any mapping of the relevant type. A negative file index can be used
to create a dummy mapping that just disables automatic copying.
For example to copy metadata from the first stream of the input
file to global metadata of the output file:
ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3
To do the reverse, i.e. copy global metadata to all audio streams:
ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv
Note that simple 0 would work as well in this example, since global
metadata is assumed by default.
-map_chapters input_file_index (output)
Copy chapters from input file with index input_file_index to the
next output file. If no chapter mapping is specified, then chapters
are copied from the first input file with at least one chapter. Use
a negative file index to disable any chapter copying.
-benchmark (global)
Show benchmarking information at the end of an encode. Shows CPU
time used and maximum memory consumption. Maximum memory
consumption is not supported on all systems, it will usually
display as 0 if not supported.
-benchmark_all (global)
Show benchmarking information during the encode. Shows CPU time
used in various steps (audio/video encode/decode).
-timelimit duration (global)
Exit after ffmpeg has been running for duration seconds.
-dump (global)
Dump each input packet to stderr.
-hex (global)
When dumping packets, also dump the payload.
-re (input)
Read input at native frame rate. Mainly used to simulate a grab
device, or live input stream (e.g. when reading from a file).
Should not be used with actual grab devices or live input streams
(where it can cause packet loss). By default ffmpeg attempts to
read the input(s) as fast as possible. This option will slow down
the reading of the input(s) to the native frame rate of the
input(s). It is useful for real-time output (e.g. live streaming).
-loop_input
Loop over the input stream. Currently it works only for image
streams. This option is used for automatic FFserver testing. This
option is deprecated, use -loop 1.
-loop_output number_of_times
Repeatedly loop output for formats that support looping such as
animated GIF (0 will loop the output infinitely). This option is
deprecated, use -loop.
-vsync parameter
Video sync method. For compatibility reasons old values can be
specified as numbers. Newly added values will have to be specified
as strings always.
0, passthrough
Each frame is passed with its timestamp from the demuxer to the
muxer.
1, cfr
Frames will be duplicated and dropped to achieve exactly the
requested constant frame rate.
2, vfr
Frames are passed through with their timestamp or dropped so as
to prevent 2 frames from having the same timestamp.
drop
As passthrough but destroys all timestamps, making the muxer
generate fresh timestamps based on frame-rate.
-1, auto
Chooses between 1 and 2 depending on muxer capabilities. This
is the default method.
Note that the timestamps may be further modified by the muxer,
after this. For example, in the case that the format option
avoid_negative_ts is enabled.
With -map you can select from which stream the timestamps should be
taken. You can leave either video or audio unchanged and sync the
remaining stream(s) to the unchanged one.
-frame_drop_threshold parameter
Frame drop threshold, which specifies how much behind video frames
can be before they are dropped. In frame rate units, so 1.0 is one
frame. The default is -1.1. One possible usecase is to avoid
framedrops in case of noisy timestamps or to increase frame drop
precision in case of exact timestamps.
-async samples_per_second
Audio sync method. "Stretches/squeezes" the audio stream to match
the timestamps, the parameter is the maximum samples per second by
which the audio is changed. -async 1 is a special case where only
the start of the audio stream is corrected without any later
correction.
Note that the timestamps may be further modified by the muxer,
after this. For example, in the case that the format option
avoid_negative_ts is enabled.
This option has been deprecated. Use the "aresample" audio filter
instead.
-copyts
Do not process input timestamps, but keep their values without
trying to sanitize them. In particular, do not remove the initial
start time offset value.
Note that, depending on the vsync option or on specific muxer
processing (e.g. in case the format option avoid_negative_ts is
enabled) the output timestamps may mismatch with the input
timestamps even when this option is selected.
-start_at_zero
When used with copyts, shift input timestamps so they start at
zero.
This means that using e.g. "-ss 50" will make output timestamps
start at 50 seconds, regardless of what timestamp the input file
started at.
-copytb mode
Specify how to set the encoder timebase when stream copying. mode
is an integer numeric value, and can assume one of the following
values:
1 Use the demuxer timebase.
The time base is copied to the output encoder from the
corresponding input demuxer. This is sometimes required to
avoid non monotonically increasing timestamps when copying
video streams with variable frame rate.
0 Use the decoder timebase.
The time base is copied to the output encoder from the
corresponding input decoder.
-1 Try to make the choice automatically, in order to generate a
sane output.
Default value is -1.
-enc_time_base[:stream_specifier] timebase (output,per-stream)
Set the encoder timebase. timebase is a floating point number, and
can assume one of the following values:
0 Assign a default value according to the media type.
For video - use 1/framerate, for audio - use 1/samplerate.
-1 Use the input stream timebase when possible.
If an input stream is not available, the default timebase will
be used.
>0 Use the provided number as the timebase.
This field can be provided as a ratio of two integers (e.g.
1:24, 1:48000) or as a floating point number (e.g. 0.04166,
2.0833e-5)
Default value is 0.
-shortest (output)
Finish encoding when the shortest input stream ends.
-dts_delta_threshold
Timestamp discontinuity delta threshold.
-muxdelay seconds (input)
Set the maximum demux-decode delay.
-muxpreload seconds (input)
Set the initial demux-decode delay.
-streamid output-stream-index:new-value (output)
Assign a new stream-id value to an output stream. This option
should be specified prior to the output filename to which it
applies. For the situation where multiple output files exist, a
streamid may be reassigned to a different value.
For example, to set the stream 0 PID to 33 and the stream 1 PID to
36 for an output mpegts file:
ffmpeg -i inurl -streamid 0:33 -streamid 1:36 out.ts
-bsf[:stream_specifier] bitstream_filters (output,per-stream)
Set bitstream filters for matching streams. bitstream_filters is a
comma-separated list of bitstream filters. Use the "-bsfs" option
to get the list of bitstream filters.
ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
-tag[:stream_specifier] codec_tag (input/output,per-stream)
Force a tag/fourcc for matching streams.
-timecode hh:mm:ssSEPff
Specify Timecode for writing. SEP is ':' for non drop timecode and
';' (or '.') for drop.
ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg
-filter_complex filtergraph (global)
Define a complex filtergraph, i.e. one with arbitrary number of
inputs and/or outputs. For simple graphs -- those with one input
and one output of the same type -- see the -filter options.
filtergraph is a description of the filtergraph, as described in
the ``Filtergraph syntax'' section of the ffmpeg-filters manual.
Input link labels must refer to input streams using the
"[file_index:stream_specifier]" syntax (i.e. the same as -map
uses). If stream_specifier matches multiple streams, the first one
will be used. An unlabeled input will be connected to the first
unused input stream of the matching type.
Output link labels are referred to with -map. Unlabeled outputs are
added to the first output file.
Note that with this option it is possible to use only lavfi sources
without normal input files.
For example, to overlay an image over video
ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map
'[out]' out.mkv
Here "[0:v]" refers to the first video stream in the first input
file, which is linked to the first (main) input of the overlay
filter. Similarly the first video stream in the second input is
linked to the second (overlay) input of overlay.
Assuming there is only one video stream in each input file, we can
omit input labels, so the above is equivalent to
ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map
'[out]' out.mkv
Furthermore we can omit the output label and the single output from
the filter graph will be added to the output file automatically, so
we can simply write
ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv
To generate 5 seconds of pure red video using lavfi "color" source:
ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
-filter_complex_threads nb_threads (global)
Defines how many threads are used to process a filter_complex
graph. Similar to filter_threads but used for "-filter_complex"
graphs only. The default is the number of available CPUs.
-lavfi filtergraph (global)
Define a complex filtergraph, i.e. one with arbitrary number of
inputs and/or outputs. Equivalent to -filter_complex.
-filter_complex_script filename (global)
This option is similar to -filter_complex, the only difference is
that its argument is the name of the file from which a complex
filtergraph description is to be read.
-accurate_seek (input)
This option enables or disables accurate seeking in input files
with the -ss option. It is enabled by default, so seeking is
accurate when transcoding. Use -noaccurate_seek to disable it,
which may be useful e.g. when copying some streams and transcoding
the others.
-seek_timestamp (input)
This option enables or disables seeking by timestamp in input files
with the -ss option. It is disabled by default. If enabled, the
argument to the -ss option is considered an actual timestamp, and
is not offset by the start time of the file. This matters only for
files which do not start from timestamp 0, such as transport
streams.
-thread_queue_size size (input)
This option sets the maximum number of queued packets when reading
from the file or device. With low latency / high rate live streams,
packets may be discarded if they are not read in a timely manner;
raising this value can avoid it.
-override_ffserver (global)
Overrides the input specifications from ffserver. Using this option
you can map any input stream to ffserver and control many aspects
of the encoding from ffmpeg. Without this option ffmpeg will
transmit to ffserver what is requested by ffserver.
The option is intended for cases where features are needed that
cannot be specified to ffserver but can be to ffmpeg.
-sdp_file file (global)
Print sdp information for an output stream to file. This allows
dumping sdp information when at least one output isn't an rtp
stream. (Requires at least one of the output formats to be rtp).
-discard (input)
Allows discarding specific streams or frames of streams at the
demuxer. Not all demuxers support this.
none
Discard no frame.
default
Default, which discards no frames.
noref
Discard all non-reference frames.
bidir
Discard all bidirectional frames.
nokey
Discard all frames excepts keyframes.
all Discard all frames.
-abort_on flags (global)
Stop and abort on various conditions. The following flags are
available:
empty_output
No packets were passed to the muxer, the output is empty.
-xerror (global)
Stop and exit on error
-max_muxing_queue_size packets (output,per-stream)
When transcoding audio and/or video streams, ffmpeg will not begin
writing into the output until it has one packet for each such
stream. While waiting for that to happen, packets for other streams
are buffered. This option sets the size of this buffer, in packets,
for the matching output stream.
The default value of this option should be high enough for most
uses, so only touch this option if you are sure that you need it.
As a special exception, you can use a bitmap subtitle stream as input:
it will be converted into a video with the same size as the largest
video in the file, or 720x576 if no video is present. Note that this is
an experimental and temporary solution. It will be removed once
libavfilter has proper support for subtitles.
For example, to hardcode subtitles on top of a DVB-T recording stored
in MPEG-TS format, delaying the subtitles by 1 second:
ffmpeg -i input.ts -filter_complex \
'[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \
-sn -map '#0x2dc' output.mkv
(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video,
audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too)
Preset files
A preset file contains a sequence of option=value pairs, one for each
line, specifying a sequence of options which would be awkward to
specify on the command line. Lines starting with the hash ('#')
character are ignored and are used to provide comments. Check the
presets directory in the FFmpeg source tree for examples.
There are two types of preset files: ffpreset and avpreset files.
ffpreset files
ffpreset files are specified with the "vpre", "apre", "spre", and
"fpre" options. The "fpre" option takes the filename of the preset
instead of a preset name as input and can be used for any kind of
codec. For the "vpre", "apre", and "spre" options, the options
specified in a preset file are applied to the currently selected codec
of the same type as the preset option.
The argument passed to the "vpre", "apre", and "spre" preset options
identifies the preset file to use according to the following rules:
First ffmpeg searches for a file named arg.ffpreset in the directories
$FFMPEG_DATADIR (if set), and $HOME/.ffmpeg, and in the datadir defined
at configuration time (usually PREFIX/share/ffmpeg) or in a ffpresets
folder along the executable on win32, in that order. For example, if
the argument is "libvpx-1080p", it will search for the file
libvpx-1080p.ffpreset.
If no such file is found, then ffmpeg will search for a file named
codec_name-arg.ffpreset in the above-mentioned directories, where
codec_name is the name of the codec to which the preset file options
will be applied. For example, if you select the video codec with
"-vcodec libvpx" and use "-vpre 1080p", then it will search for the
file libvpx-1080p.ffpreset.
avpreset files
avpreset files are specified with the "pre" option. They work similar
to ffpreset files, but they only allow encoder- specific options.
Therefore, an option=value pair specifying an encoder cannot be used.
When the "pre" option is specified, ffmpeg will look for files with the
suffix .avpreset in the directories $AVCONV_DATADIR (if set), and
$HOME/.avconv, and in the datadir defined at configuration time
(usually PREFIX/share/ffmpeg), in that order.
First ffmpeg searches for a file named codec_name-arg.avpreset in the
above-mentioned directories, where codec_name is the name of the codec
to which the preset file options will be applied. For example, if you
select the video codec with "-vcodec libvpx" and use "-pre 1080p", then
it will search for the file libvpx-1080p.avpreset.
If no such file is found, then ffmpeg will search for a file named
arg.avpreset in the same directories.
EXAMPLES
Video and Audio grabbing
If you specify the input format and device then ffmpeg can grab video
and audio directly.
ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg
Or with an ALSA audio source (mono input, card id 1) instead of OSS:
ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg
Note that you must activate the right video source and channel before
launching ffmpeg with any TV viewer such as
<http://linux.bytesex.org/xawtv/> by Gerd Knorr. You also have to set
the audio recording levels correctly with a standard mixer.
X11 grabbing
Grab the X11 display with ffmpeg via
ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg
0.0 is display.screen number of your X11 server, same as the DISPLAY
environment variable.
ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg
0.0 is display.screen number of your X11 server, same as the DISPLAY
environment variable. 10 is the x-offset and 20 the y-offset for the
grabbing.
Video and Audio file format conversion
Any supported file format and protocol can serve as input to ffmpeg:
Examples:
路 You can use YUV files as input:
ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
It will use the files:
/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
The Y files use twice the resolution of the U and V files. They are
raw files, without header. They can be generated by all decent
video decoders. You must specify the size of the image with the -s
option if ffmpeg cannot guess it.
路 You can input from a raw YUV420P file:
ffmpeg -i /tmp/test.yuv /tmp/out.avi
test.yuv is a file containing raw YUV planar data. Each frame is
composed of the Y plane followed by the U and V planes at half
vertical and horizontal resolution.
路 You can output to a raw YUV420P file:
ffmpeg -i mydivx.avi hugefile.yuv
路 You can set several input files and output files:
ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
Converts the audio file a.wav and the raw YUV video file a.yuv to
MPEG file a.mpg.
路 You can also do audio and video conversions at the same time:
ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
Converts a.wav to MPEG audio at 22050 Hz sample rate.
路 You can encode to several formats at the same time and define a
mapping from input stream to output streams:
ffmpeg -i /tmp/a.wav -map 0:a -b:a 64k /tmp/a.mp2 -map 0:a -b:a 128k /tmp/b.mp2
Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits.
'-map file:index' specifies which input stream is used for each
output stream, in the order of the definition of output streams.
路 You can transcode decrypted VOBs:
ffmpeg -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi
This is a typical DVD ripping example; the input is a VOB file, the
output an AVI file with MPEG-4 video and MP3 audio. Note that in
this command we use B-frames so the MPEG-4 stream is DivX5
compatible, and GOP size is 300 which means one intra frame every
10 seconds for 29.97fps input video. Furthermore, the audio stream
is MP3-encoded so you need to enable LAME support by passing
"--enable-libmp3lame" to configure. The mapping is particularly
useful for DVD transcoding to get the desired audio language.
NOTE: To see the supported input formats, use "ffmpeg -demuxers".
路 You can extract images from a video, or create a video from many
images:
For extracting images from a video:
ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
This will extract one video frame per second from the video and
will output them in files named foo-001.jpeg, foo-002.jpeg, etc.
Images will be rescaled to fit the new WxH values.
If you want to extract just a limited number of frames, you can use
the above command in combination with the "-frames:v" or "-t"
option, or in combination with -ss to start extracting from a
certain point in time.
For creating a video from many images:
ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
The syntax "foo-%03d.jpeg" specifies to use a decimal number
composed of three digits padded with zeroes to express the sequence
number. It is the same syntax supported by the C printf function,
but only formats accepting a normal integer are suitable.
When importing an image sequence, -i also supports expanding shell-
like wildcard patterns (globbing) internally, by selecting the
image2-specific "-pattern_type glob" option.
For example, for creating a video from filenames matching the glob
pattern "foo-*.jpeg":
ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi
路 You can put many streams of the same type in the output:
ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
The resulting output file test12.nut will contain the first four
streams from the input files in reverse order.
路 To force CBR video output:
ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
路 The four options lmin, lmax, mblmin and mblmax use 'lambda' units,
but you may use the QP2LAMBDA constant to easily convert from 'q'
units:
ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
SYNTAX
This section documents the syntax and formats employed by the FFmpeg
libraries and tools.
Quoting and escaping
FFmpeg adopts the following quoting and escaping mechanism, unless
explicitly specified. The following rules are applied:
路 ' and \ are special characters (respectively used for quoting and
escaping). In addition to them, there might be other special
characters depending on the specific syntax where the escaping and
quoting are employed.
路 A special character is escaped by prefixing it with a \.
路 All characters enclosed between '' are included literally in the
parsed string. The quote character ' itself cannot be quoted, so
you may need to close the quote and escape it.
路 Leading and trailing whitespaces, unless escaped or quoted, are
removed from the parsed string.
Note that you may need to add a second level of escaping when using the
command line or a script, which depends on the syntax of the adopted
shell language.
The function "av_get_token" defined in libavutil/avstring.h can be used
to parse a token quoted or escaped according to the rules defined
above.
The tool tools/ffescape in the FFmpeg source tree can be used to
automatically quote or escape a string in a script.
Examples
路 Escape the string "Crime d'Amour" containing the "'" special
character:
Crime d\'Amour
路 The string above contains a quote, so the "'" needs to be escaped
when quoting it:
'Crime d'\''Amour'
路 Include leading or trailing whitespaces using quoting:
' this string starts and ends with whitespaces '
路 Escaping and quoting can be mixed together:
' The string '\'string\'' is a string '
路 To include a literal \ you can use either escaping or quoting:
'c:\foo' can be written as c:\\foo
Date
The accepted syntax is:
[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
now
If the value is "now" it takes the current time.
Time is local time unless Z is appended, in which case it is
interpreted as UTC. If the year-month-day part is not specified it
takes the current year-month-day.
Time duration
There are two accepted syntaxes for expressing time duration.
[-][<HH>:]<MM>:<SS>[.<m>...]
HH expresses the number of hours, MM the number of minutes for a
maximum of 2 digits, and SS the number of seconds for a maximum of 2
digits. The m at the end expresses decimal value for SS.
or
[-]<S>+[.<m>...]
S expresses the number of seconds, with the optional decimal part m.
In both expressions, the optional - indicates negative duration.
Examples
The following examples are all valid time duration:
55 55 seconds
12:03:45
12 hours, 03 minutes and 45 seconds
23.189
23.189 seconds
Video size
Specify the size of the sourced video, it may be a string of the form
widthxheight, or the name of a size abbreviation.
The following abbreviations are recognized:
ntsc
720x480
pal 720x576
qntsc
352x240
qpal
352x288
sntsc
640x480
spal
768x576
film
352x240
ntsc-film
352x240
sqcif
128x96
qcif
176x144
cif 352x288
4cif
704x576
16cif
1408x1152
qqvga
160x120
qvga
320x240
vga 640x480
svga
800x600
xga 1024x768
uxga
1600x1200
qxga
2048x1536
sxga
1280x1024
qsxga
2560x2048
hsxga
5120x4096
wvga
852x480
wxga
1366x768
wsxga
1600x1024
wuxga
1920x1200
woxga
2560x1600
wqsxga
3200x2048
wquxga
3840x2400
whsxga
6400x4096
whuxga
7680x4800
cga 320x200
ega 640x350
hd480
852x480
hd720
1280x720
hd1080
1920x1080
2k 2048x1080
2kflat
1998x1080
2kscope
2048x858
4k 4096x2160
4kflat
3996x2160
4kscope
4096x1716
nhd 640x360
hqvga
240x160
wqvga
400x240
fwqvga
432x240
hvga
480x320
qhd 960x540
2kdci
2048x1080
4kdci
4096x2160
uhd2160
3840x2160
uhd4320
7680x4320
Video rate
Specify the frame rate of a video, expressed as the number of frames
generated per second. It has to be a string in the format
frame_rate_num/frame_rate_den, an integer number, a float number or a
valid video frame rate abbreviation.
The following abbreviations are recognized:
ntsc
30000/1001
pal 25/1
qntsc
30000/1001
qpal
25/1
sntsc
30000/1001
spal
25/1
film
24/1
ntsc-film
24000/1001
Ratio
A ratio can be expressed as an expression, or in the form
numerator:denominator.
Note that a ratio with infinite (1/0) or negative value is considered
valid, so you should check on the returned value if you want to exclude
those values.
The undefined value can be expressed using the "0:0" string.
Color
It can be the name of a color as defined below (case insensitive match)
or a "[0x|#]RRGGBB[AA]" sequence, possibly followed by @ and a string
representing the alpha component.
The alpha component may be a string composed by "0x" followed by an
hexadecimal number or a decimal number between 0.0 and 1.0, which
represents the opacity value (0x00 or 0.0 means completely transparent,
0xff or 1.0 completely opaque). If the alpha component is not specified
then 0xff is assumed.
The string random will result in a random color.
The following names of colors are recognized:
AliceBlue
0xF0F8FF
AntiqueWhite
0xFAEBD7
Aqua
0x00FFFF
Aquamarine
0x7FFFD4
Azure
0xF0FFFF
Beige
0xF5F5DC
Bisque
0xFFE4C4
Black
0x000000
BlanchedAlmond
0xFFEBCD
Blue
0x0000FF
BlueViolet
0x8A2BE2
Brown
0xA52A2A
BurlyWood
0xDEB887
CadetBlue
0x5F9EA0
Chartreuse
0x7FFF00
Chocolate
0xD2691E
Coral
0xFF7F50
CornflowerBlue
0x6495ED
Cornsilk
0xFFF8DC
Crimson
0xDC143C
Cyan
0x00FFFF
DarkBlue
0x00008B
DarkCyan
0x008B8B
DarkGoldenRod
0xB8860B
DarkGray
0xA9A9A9
DarkGreen
0x006400
DarkKhaki
0xBDB76B
DarkMagenta
0x8B008B
DarkOliveGreen
0x556B2F
Darkorange
0xFF8C00
DarkOrchid
0x9932CC
DarkRed
0x8B0000
DarkSalmon
0xE9967A
DarkSeaGreen
0x8FBC8F
DarkSlateBlue
0x483D8B
DarkSlateGray
0x2F4F4F
DarkTurquoise
0x00CED1
DarkViolet
0x9400D3
DeepPink
0xFF1493
DeepSkyBlue
0x00BFFF
DimGray
0x696969
DodgerBlue
0x1E90FF
FireBrick
0xB22222
FloralWhite
0xFFFAF0
ForestGreen
0x228B22
Fuchsia
0xFF00FF
Gainsboro
0xDCDCDC
GhostWhite
0xF8F8FF
Gold
0xFFD700
GoldenRod
0xDAA520
Gray
0x808080
Green
0x008000
GreenYellow
0xADFF2F
HoneyDew
0xF0FFF0
HotPink
0xFF69B4
IndianRed
0xCD5C5C
Indigo
0x4B0082
Ivory
0xFFFFF0
Khaki
0xF0E68C
Lavender
0xE6E6FA
LavenderBlush
0xFFF0F5
LawnGreen
0x7CFC00
LemonChiffon
0xFFFACD
LightBlue
0xADD8E6
LightCoral
0xF08080
LightCyan
0xE0FFFF
LightGoldenRodYellow
0xFAFAD2
LightGreen
0x90EE90
LightGrey
0xD3D3D3
LightPink
0xFFB6C1
LightSalmon
0xFFA07A
LightSeaGreen
0x20B2AA
LightSkyBlue
0x87CEFA
LightSlateGray
0x778899
LightSteelBlue
0xB0C4DE
LightYellow
0xFFFFE0
Lime
0x00FF00
LimeGreen
0x32CD32
Linen
0xFAF0E6
Magenta
0xFF00FF
Maroon
0x800000
MediumAquaMarine
0x66CDAA
MediumBlue
0x0000CD
MediumOrchid
0xBA55D3
MediumPurple
0x9370D8
MediumSeaGreen
0x3CB371
MediumSlateBlue
0x7B68EE
MediumSpringGreen
0x00FA9A
MediumTurquoise
0x48D1CC
MediumVioletRed
0xC71585
MidnightBlue
0x191970
MintCream
0xF5FFFA
MistyRose
0xFFE4E1
Moccasin
0xFFE4B5
NavajoWhite
0xFFDEAD
Navy
0x000080
OldLace
0xFDF5E6
Olive
0x808000
OliveDrab
0x6B8E23
Orange
0xFFA500
OrangeRed
0xFF4500
Orchid
0xDA70D6
PaleGoldenRod
0xEEE8AA
PaleGreen
0x98FB98
PaleTurquoise
0xAFEEEE
PaleVioletRed
0xD87093
PapayaWhip
0xFFEFD5
PeachPuff
0xFFDAB9
Peru
0xCD853F
Pink
0xFFC0CB
Plum
0xDDA0DD
PowderBlue
0xB0E0E6
Purple
0x800080
Red 0xFF0000
RosyBrown
0xBC8F8F
RoyalBlue
0x4169E1
SaddleBrown
0x8B4513
Salmon
0xFA8072
SandyBrown
0xF4A460
SeaGreen
0x2E8B57
SeaShell
0xFFF5EE
Sienna
0xA0522D
Silver
0xC0C0C0
SkyBlue
0x87CEEB
SlateBlue
0x6A5ACD
SlateGray
0x708090
Snow
0xFFFAFA
SpringGreen
0x00FF7F
SteelBlue
0x4682B4
Tan 0xD2B48C
Teal
0x008080
Thistle
0xD8BFD8
Tomato
0xFF6347
Turquoise
0x40E0D0
Violet
0xEE82EE
Wheat
0xF5DEB3
White
0xFFFFFF
WhiteSmoke
0xF5F5F5
Yellow
0xFFFF00
YellowGreen
0x9ACD32
Channel Layout
A channel layout specifies the spatial disposition of the channels in a
multi-channel audio stream. To specify a channel layout, FFmpeg makes
use of a special syntax.
Individual channels are identified by an id, as given by the table
below:
FL front left
FR front right
FC front center
LFE low frequency
BL back left
BR back right
FLC front left-of-center
FRC front right-of-center
BC back center
SL side left
SR side right
TC top center
TFL top front left
TFC top front center
TFR top front right
TBL top back left
TBC top back center
TBR top back right
DL downmix left
DR downmix right
WL wide left
WR wide right
SDL surround direct left
SDR surround direct right
LFE2
low frequency 2
Standard channel layout compositions can be specified by using the
following identifiers:
mono
FC
stereo
FL+FR
2.1 FL+FR+LFE
3.0 FL+FR+FC
3.0(back)
FL+FR+BC
4.0 FL+FR+FC+BC
quad
FL+FR+BL+BR
quad(side)
FL+FR+SL+SR
3.1 FL+FR+FC+LFE
5.0 FL+FR+FC+BL+BR
5.0(side)
FL+FR+FC+SL+SR
4.1 FL+FR+FC+LFE+BC
5.1 FL+FR+FC+LFE+BL+BR
5.1(side)
FL+FR+FC+LFE+SL+SR
6.0 FL+FR+FC+BC+SL+SR
6.0(front)
FL+FR+FLC+FRC+SL+SR
hexagonal
FL+FR+FC+BL+BR+BC
6.1 FL+FR+FC+LFE+BC+SL+SR
6.1 FL+FR+FC+LFE+BL+BR+BC
6.1(front)
FL+FR+LFE+FLC+FRC+SL+SR
7.0 FL+FR+FC+BL+BR+SL+SR
7.0(front)
FL+FR+FC+FLC+FRC+SL+SR
7.1 FL+FR+FC+LFE+BL+BR+SL+SR
7.1(wide)
FL+FR+FC+LFE+BL+BR+FLC+FRC
7.1(wide-side)
FL+FR+FC+LFE+FLC+FRC+SL+SR
octagonal
FL+FR+FC+BL+BR+BC+SL+SR
downmix
DL+DR
A custom channel layout can be specified as a sequence of terms,
separated by '+' or '|'. Each term can be:
路 the name of a standard channel layout (e.g. mono, stereo, 4.0,
quad, 5.0, etc.)
路 the name of a single channel (e.g. FL, FR, FC, LFE, etc.)
路 a number of channels, in decimal, followed by 'c', yielding the
default channel layout for that number of channels (see the
function "av_get_default_channel_layout"). Note that not all
channel counts have a default layout.
路 a number of channels, in decimal, followed by 'C', yielding an
unknown channel layout with the specified number of channels. Note
that not all channel layout specification strings support unknown
channel layouts.
路 a channel layout mask, in hexadecimal starting with "0x" (see the
"AV_CH_*" macros in libavutil/channel_layout.h.
Before libavutil version 53 the trailing character "c" to specify a
number of channels was optional, but now it is required, while a
channel layout mask can also be specified as a decimal number (if and
only if not followed by "c" or "C").
See also the function "av_get_channel_layout" defined in
libavutil/channel_layout.h.
EXPRESSION EVALUATION
When evaluating an arithmetic expression, FFmpeg uses an internal
formula evaluator, implemented through the libavutil/eval.h interface.
An expression may contain unary, binary operators, constants, and
functions.
Two expressions expr1 and expr2 can be combined to form another
expression "expr1;expr2". expr1 and expr2 are evaluated in turn, and
the new expression evaluates to the value of expr2.
The following binary operators are available: "+", "-", "*", "/", "^".
The following unary operators are available: "+", "-".
The following functions are available:
abs(x)
Compute absolute value of x.
acos(x)
Compute arccosine of x.
asin(x)
Compute arcsine of x.
atan(x)
Compute arctangent of x.
atan2(x, y)
Compute principal value of the arc tangent of y/x.
between(x, min, max)
Return 1 if x is greater than or equal to min and lesser than or
equal to max, 0 otherwise.
bitand(x, y)
bitor(x, y)
Compute bitwise and/or operation on x and y.
The results of the evaluation of x and y are converted to integers
before executing the bitwise operation.
Note that both the conversion to integer and the conversion back to
floating point can lose precision. Beware of unexpected results for
large numbers (usually 2^53 and larger).
ceil(expr)
Round the value of expression expr upwards to the nearest integer.
For example, "ceil(1.5)" is "2.0".
clip(x, min, max)
Return the value of x clipped between min and max.
cos(x)
Compute cosine of x.
cosh(x)
Compute hyperbolic cosine of x.
eq(x, y)
Return 1 if x and y are equivalent, 0 otherwise.
exp(x)
Compute exponential of x (with base "e", the Euler's number).
floor(expr)
Round the value of expression expr downwards to the nearest
integer. For example, "floor(-1.5)" is "-2.0".
gauss(x)
Compute Gauss function of x, corresponding to "exp(-x*x/2) /
sqrt(2*PI)".
gcd(x, y)
Return the greatest common divisor of x and y. If both x and y are
0 or either or both are less than zero then behavior is undefined.
gt(x, y)
Return 1 if x is greater than y, 0 otherwise.
gte(x, y)
Return 1 if x is greater than or equal to y, 0 otherwise.
hypot(x, y)
This function is similar to the C function with the same name; it
returns "sqrt(x*x + y*y)", the length of the hypotenuse of a right
triangle with sides of length x and y, or the distance of the point
(x, y) from the origin.
if(x, y)
Evaluate x, and if the result is non-zero return the result of the
evaluation of y, return 0 otherwise.
if(x, y, z)
Evaluate x, and if the result is non-zero return the evaluation
result of y, otherwise the evaluation result of z.
ifnot(x, y)
Evaluate x, and if the result is zero return the result of the
evaluation of y, return 0 otherwise.
ifnot(x, y, z)
Evaluate x, and if the result is zero return the evaluation result
of y, otherwise the evaluation result of z.
isinf(x)
Return 1.0 if x is +/-INFINITY, 0.0 otherwise.
isnan(x)
Return 1.0 if x is NAN, 0.0 otherwise.
ld(var)
Load the value of the internal variable with number var, which was
previously stored with st(var, expr). The function returns the
loaded value.
lerp(x, y, z)
Return linear interpolation between x and y by amount of z.
log(x)
Compute natural logarithm of x.
lt(x, y)
Return 1 if x is lesser than y, 0 otherwise.
lte(x, y)
Return 1 if x is lesser than or equal to y, 0 otherwise.
max(x, y)
Return the maximum between x and y.
min(x, y)
Return the minimum between x and y.
mod(x, y)
Compute the remainder of division of x by y.
not(expr)
Return 1.0 if expr is zero, 0.0 otherwise.
pow(x, y)
Compute the power of x elevated y, it is equivalent to "(x)^(y)".
print(t)
print(t, l)
Print the value of expression t with loglevel l. If l is not
specified then a default log level is used. Returns the value of
the expression printed.
Prints t with loglevel l
random(x)
Return a pseudo random value between 0.0 and 1.0. x is the index of
the internal variable which will be used to save the seed/state.
root(expr, max)
Find an input value for which the function represented by expr with
argument ld(0) is 0 in the interval 0..max.
The expression in expr must denote a continuous function or the
result is undefined.
ld(0) is used to represent the function input value, which means
that the given expression will be evaluated multiple times with
various input values that the expression can access through ld(0).
When the expression evaluates to 0 then the corresponding input
value will be returned.
round(expr)
Round the value of expression expr to the nearest integer. For
example, "round(1.5)" is "2.0".
sin(x)
Compute sine of x.
sinh(x)
Compute hyperbolic sine of x.
sqrt(expr)
Compute the square root of expr. This is equivalent to "(expr)^.5".
squish(x)
Compute expression "1/(1 + exp(4*x))".
st(var, expr)
Store the value of the expression expr in an internal variable. var
specifies the number of the variable where to store the value, and
it is a value ranging from 0 to 9. The function returns the value
stored in the internal variable. Note, Variables are currently not
shared between expressions.
tan(x)
Compute tangent of x.
tanh(x)
Compute hyperbolic tangent of x.
taylor(expr, x)
taylor(expr, x, id)
Evaluate a Taylor series at x, given an expression representing the
"ld(id)"-th derivative of a function at 0.
When the series does not converge the result is undefined.
ld(id) is used to represent the derivative order in expr, which
means that the given expression will be evaluated multiple times
with various input values that the expression can access through
"ld(id)". If id is not specified then 0 is assumed.
Note, when you have the derivatives at y instead of 0,
"taylor(expr, x-y)" can be used.
time(0)
Return the current (wallclock) time in seconds.
trunc(expr)
Round the value of expression expr towards zero to the nearest
integer. For example, "trunc(-1.5)" is "-1.0".
while(cond, expr)
Evaluate expression expr while the expression cond is non-zero, and
returns the value of the last expr evaluation, or NAN if cond was
always false.
The following constants are available:
PI area of the unit disc, approximately 3.14
E exp(1) (Euler's number), approximately 2.718
PHI golden ratio (1+sqrt(5))/2, approximately 1.618
Assuming that an expression is considered "true" if it has a non-zero
value, note that:
"*" works like AND
"+" works like OR
For example the construct:
if (A AND B) then C
is equivalent to:
if(A*B, C)
In your C code, you can extend the list of unary and binary functions,
and define recognized constants, so that they are available for your
expressions.
The evaluator also recognizes the International System unit prefixes.
If 'i' is appended after the prefix, binary prefixes are used, which
are based on powers of 1024 instead of powers of 1000. The 'B' postfix
multiplies the value by 8, and can be appended after a unit prefix or
used alone. This allows using for example 'KB', 'MiB', 'G' and 'B' as
number postfix.
The list of available International System prefixes follows, with
indication of the corresponding powers of 10 and of 2.
y 10^-24 / 2^-80
z 10^-21 / 2^-70
a 10^-18 / 2^-60
f 10^-15 / 2^-50
p 10^-12 / 2^-40
n 10^-9 / 2^-30
u 10^-6 / 2^-20
m 10^-3 / 2^-10
c 10^-2
d 10^-1
h 10^2
k 10^3 / 2^10
K 10^3 / 2^10
M 10^6 / 2^20
G 10^9 / 2^30
T 10^12 / 2^40
P 10^15 / 2^40
E 10^18 / 2^50
Z 10^21 / 2^60
Y 10^24 / 2^70
OPENCL OPTIONS
When FFmpeg is configured with "--enable-opencl", it is possible to set
the options for the global OpenCL context.
The list of supported options follows:
build_options
Set build options used to compile the registered kernels.
See reference "OpenCL Specification Version: 1.2 chapter 5.6.4".
platform_idx
Select the index of the platform to run OpenCL code.
The specified index must be one of the indexes in the device list
which can be obtained with "ffmpeg -opencl_bench" or
"av_opencl_get_device_list()".
device_idx
Select the index of the device used to run OpenCL code.
The specified index must be one of the indexes in the device list
which can be obtained with "ffmpeg -opencl_bench" or
"av_opencl_get_device_list()".
CODEC OPTIONS
libavcodec provides some generic global options, which can be set on
all the encoders and decoders. In addition each codec may support so-
called private options, which are specific for a given codec.
Sometimes, a global option may only affect a specific kind of codec,
and may be nonsensical or ignored by another, so you need to be aware
of the meaning of the specified options. Also some options are meant
only for decoding or encoding.
Options may be set by specifying -option value in the FFmpeg tools, or
by setting the value explicitly in the "AVCodecContext" options or
using the libavutil/opt.h API for programmatic use.
The list of supported options follow:
b integer (encoding,audio,video)
Set bitrate in bits/s. Default value is 200K.
ab integer (encoding,audio)
Set audio bitrate (in bits/s). Default value is 128K.
bt integer (encoding,video)
Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate
tolerance specifies how far ratecontrol is willing to deviate from
the target average bitrate value. This is not related to min/max
bitrate. Lowering tolerance too much has an adverse effect on
quality.
flags flags (decoding/encoding,audio,video,subtitles)
Set generic flags.
Possible values:
mv4 Use four motion vector by macroblock (mpeg4).
qpel
Use 1/4 pel motion compensation.
loop
Use loop filter.
qscale
Use fixed qscale.
gmc Use gmc.
mv0 Always try a mb with mv=<0,0>.
input_preserved
pass1
Use internal 2pass ratecontrol in first pass mode.
pass2
Use internal 2pass ratecontrol in second pass mode.
gray
Only decode/encode grayscale.
emu_edge
Do not draw edges.
psnr
Set error[?] variables during encoding.
truncated
naq Normalize adaptive quantization.
ildct
Use interlaced DCT.
low_delay
Force low delay.
global_header
Place global headers in extradata instead of every keyframe.
bitexact
Only write platform-, build- and time-independent data. (except
(I)DCT). This ensures that file and data checksums are
reproducible and match between platforms. Its primary use is
for regression testing.
aic Apply H263 advanced intra coding / mpeg4 ac prediction.
cbp Deprecated, use mpegvideo private options instead.
qprd
Deprecated, use mpegvideo private options instead.
ilme
Apply interlaced motion estimation.
cgop
Use closed gop.
me_method integer (encoding,video)
Set motion estimation method.
Possible values:
zero
zero motion estimation (fastest)
full
full motion estimation (slowest)
epzs
EPZS motion estimation (default)
esa esa motion estimation (alias for full)
tesa
tesa motion estimation
dia dia motion estimation (alias for epzs)
log log motion estimation
phods
phods motion estimation
x1 X1 motion estimation
hex hex motion estimation
umh umh motion estimation
iter
iter motion estimation
extradata_size integer
Set extradata size.
time_base rational number
Set codec time base.
It is the fundamental unit of time (in seconds) in terms of which
frame timestamps are represented. For fixed-fps content, timebase
should be "1 / frame_rate" and timestamp increments should be
identically 1.
g integer (encoding,video)
Set the group of picture (GOP) size. Default value is 12.
ar integer (decoding/encoding,audio)
Set audio sampling rate (in Hz).
ac integer (decoding/encoding,audio)
Set number of audio channels.
cutoff integer (encoding,audio)
Set cutoff bandwidth. (Supported only by selected encoders, see
their respective documentation sections.)
frame_size integer (encoding,audio)
Set audio frame size.
Each submitted frame except the last must contain exactly
frame_size samples per channel. May be 0 when the codec has
CODEC_CAP_VARIABLE_FRAME_SIZE set, in that case the frame size is
not restricted. It is set by some decoders to indicate constant
frame size.
frame_number integer
Set the frame number.
delay integer
qcomp float (encoding,video)
Set video quantizer scale compression (VBR). It is used as a
constant in the ratecontrol equation. Recommended range for default
rc_eq: 0.0-1.0.
qblur float (encoding,video)
Set video quantizer scale blur (VBR).
qmin integer (encoding,video)
Set min video quantizer scale (VBR). Must be included between -1
and 69, default value is 2.
qmax integer (encoding,video)
Set max video quantizer scale (VBR). Must be included between -1
and 1024, default value is 31.
qdiff integer (encoding,video)
Set max difference between the quantizer scale (VBR).
bf integer (encoding,video)
Set max number of B frames between non-B-frames.
Must be an integer between -1 and 16. 0 means that B-frames are
disabled. If a value of -1 is used, it will choose an automatic
value depending on the encoder.
Default value is 0.
b_qfactor float (encoding,video)
Set qp factor between P and B frames.
rc_strategy integer (encoding,video)
Set ratecontrol method.
b_strategy integer (encoding,video)
Set strategy to choose between I/P/B-frames.
ps integer (encoding,video)
Set RTP payload size in bytes.
mv_bits integer
header_bits integer
i_tex_bits integer
p_tex_bits integer
i_count integer
p_count integer
skip_count integer
misc_bits integer
frame_bits integer
codec_tag integer
bug flags (decoding,video)
Workaround not auto detected encoder bugs.
Possible values:
autodetect
old_msmpeg4
some old lavc generated msmpeg4v3 files (no autodetection)
xvid_ilace
Xvid interlacing bug (autodetected if fourcc==XVIX)
ump4
(autodetected if fourcc==UMP4)
no_padding
padding bug (autodetected)
amv
ac_vlc
illegal vlc bug (autodetected per fourcc)
qpel_chroma
std_qpel
old standard qpel (autodetected per fourcc/version)
qpel_chroma2
direct_blocksize
direct-qpel-blocksize bug (autodetected per fourcc/version)
edge
edge padding bug (autodetected per fourcc/version)
hpel_chroma
dc_clip
ms Workaround various bugs in microsoft broken decoders.
trunc
trancated frames
lelim integer (encoding,video)
Set single coefficient elimination threshold for luminance
(negative values also consider DC coefficient).
celim integer (encoding,video)
Set single coefficient elimination threshold for chrominance
(negative values also consider dc coefficient)
strict integer (decoding/encoding,audio,video)
Specify how strictly to follow the standards.
Possible values:
very
strictly conform to an older more strict version of the spec or
reference software
strict
strictly conform to all the things in the spec no matter what
consequences
normal
unofficial
allow unofficial extensions
experimental
allow non standardized experimental things, experimental
(unfinished/work in progress/not well tested) decoders and
encoders. Note: experimental decoders can pose a security
risk, do not use this for decoding untrusted input.
b_qoffset float (encoding,video)
Set QP offset between P and B frames.
err_detect flags (decoding,audio,video)
Set error detection flags.
Possible values:
crccheck
verify embedded CRCs
bitstream
detect bitstream specification deviations
buffer
detect improper bitstream length
explode
abort decoding on minor error detection
ignore_err
ignore decoding errors, and continue decoding. This is useful
if you want to analyze the content of a video and thus want
everything to be decoded no matter what. This option will not
result in a video that is pleasing to watch in case of errors.
careful
consider things that violate the spec and have not been seen in
the wild as errors
compliant
consider all spec non compliancies as errors
aggressive
consider things that a sane encoder should not do as an error
has_b_frames integer
block_align integer
mpeg_quant integer (encoding,video)
Use MPEG quantizers instead of H.263.
qsquish float (encoding,video)
How to keep quantizer between qmin and qmax (0 = clip, 1 = use
differentiable function).
rc_qmod_amp float (encoding,video)
Set experimental quantizer modulation.
rc_qmod_freq integer (encoding,video)
Set experimental quantizer modulation.
rc_override_count integer
rc_eq string (encoding,video)
Set rate control equation. When computing the expression, besides
the standard functions defined in the section 'Expression
Evaluation', the following functions are available: bits2qp(bits),
qp2bits(qp). Also the following constants are available: iTex pTex
tex mv fCode iCount mcVar var isI isP isB avgQP qComp avgIITex
avgPITex avgPPTex avgBPTex avgTex.
maxrate integer (encoding,audio,video)
Set max bitrate tolerance (in bits/s). Requires bufsize to be set.
minrate integer (encoding,audio,video)
Set min bitrate tolerance (in bits/s). Most useful in setting up a
CBR encode. It is of little use elsewise.
bufsize integer (encoding,audio,video)
Set ratecontrol buffer size (in bits).
rc_buf_aggressivity float (encoding,video)
Currently useless.
i_qfactor float (encoding,video)
Set QP factor between P and I frames.
i_qoffset float (encoding,video)
Set QP offset between P and I frames.
rc_init_cplx float (encoding,video)
Set initial complexity for 1-pass encoding.
dct integer (encoding,video)
Set DCT algorithm.
Possible values:
auto
autoselect a good one (default)
fastint
fast integer
int accurate integer
mmx
altivec
faan
floating point AAN DCT
lumi_mask float (encoding,video)
Compress bright areas stronger than medium ones.
tcplx_mask float (encoding,video)
Set temporal complexity masking.
scplx_mask float (encoding,video)
Set spatial complexity masking.
p_mask float (encoding,video)
Set inter masking.
dark_mask float (encoding,video)
Compress dark areas stronger than medium ones.
idct integer (decoding/encoding,video)
Select IDCT implementation.
Possible values:
auto
int
simple
simplemmx
simpleauto
Automatically pick a IDCT compatible with the simple one
arm
altivec
sh4
simplearm
simplearmv5te
simplearmv6
simpleneon
simplealpha
ipp
xvidmmx
faani
floating point AAN IDCT
slice_count integer
ec flags (decoding,video)
Set error concealment strategy.
Possible values:
guess_mvs
iterative motion vector (MV) search (slow)
deblock
use strong deblock filter for damaged MBs
favor_inter
favor predicting from the previous frame instead of the current
bits_per_coded_sample integer
pred integer (encoding,video)
Set prediction method.
Possible values:
left
plane
median
aspect rational number (encoding,video)
Set sample aspect ratio.
sar rational number (encoding,video)
Set sample aspect ratio. Alias to aspect.
debug flags (decoding/encoding,audio,video,subtitles)
Print specific debug info.
Possible values:
pict
picture info
rc rate control
bitstream
mb_type
macroblock (MB) type
qp per-block quantization parameter (QP)
mv motion vector
dct_coeff
green_metadata
display complexity metadata for the upcoming frame, GoP or for
a given duration.
skip
startcode
pts
er error recognition
mmco
memory management control operations (H.264)
bugs
vis_qp
visualize quantization parameter (QP), lower QP are tinted
greener
vis_mb_type
visualize block types
buffers
picture buffer allocations
thread_ops
threading operations
nomc
skip motion compensation
vismv integer (decoding,video)
Visualize motion vectors (MVs).
This option is deprecated, see the codecview filter instead.
Possible values:
pf forward predicted MVs of P-frames
bf forward predicted MVs of B-frames
bb backward predicted MVs of B-frames
cmp integer (encoding,video)
Set full pel me compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
subcmp integer (encoding,video)
Set sub pel me compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
mbcmp integer (encoding,video)
Set macroblock compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
ildctcmp integer (encoding,video)
Set interlaced dct compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
dia_size integer (encoding,video)
Set diamond type & size for motion estimation.
last_pred integer (encoding,video)
Set amount of motion predictors from the previous frame.
preme integer (encoding,video)
Set pre motion estimation.
precmp integer (encoding,video)
Set pre motion estimation compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
pre_dia_size integer (encoding,video)
Set diamond type & size for motion estimation pre-pass.
subq integer (encoding,video)
Set sub pel motion estimation quality.
dtg_active_format integer
me_range integer (encoding,video)
Set limit motion vectors range (1023 for DivX player).
ibias integer (encoding,video)
Set intra quant bias.
pbias integer (encoding,video)
Set inter quant bias.
color_table_id integer
global_quality integer (encoding,audio,video)
coder integer (encoding,video)
Possible values:
vlc variable length coder / huffman coder
ac arithmetic coder
raw raw (no encoding)
rle run-length coder
deflate
deflate-based coder
context integer (encoding,video)
Set context model.
slice_flags integer
xvmc_acceleration integer
mbd integer (encoding,video)
Set macroblock decision algorithm (high quality mode).
Possible values:
simple
use mbcmp (default)
bits
use fewest bits
rd use best rate distortion
stream_codec_tag integer
sc_threshold integer (encoding,video)
Set scene change threshold.
lmin integer (encoding,video)
Set min lagrange factor (VBR).
lmax integer (encoding,video)
Set max lagrange factor (VBR).
nr integer (encoding,video)
Set noise reduction.
rc_init_occupancy integer (encoding,video)
Set number of bits which should be loaded into the rc buffer before
decoding starts.
flags2 flags (decoding/encoding,audio,video)
Possible values:
fast
Allow non spec compliant speedup tricks.
sgop
Deprecated, use mpegvideo private options instead.
noout
Skip bitstream encoding.
ignorecrop
Ignore cropping information from sps.
local_header
Place global headers at every keyframe instead of in extradata.
chunks
Frame data might be split into multiple chunks.
showall
Show all frames before the first keyframe.
skiprd
Deprecated, use mpegvideo private options instead.
export_mvs
Export motion vectors into frame side-data (see
"AV_FRAME_DATA_MOTION_VECTORS") for codecs that support it. See
also doc/examples/export_mvs.c.
error integer (encoding,video)
qns integer (encoding,video)
Deprecated, use mpegvideo private options instead.
threads integer (decoding/encoding,video)
Set the number of threads to be used, in case the selected codec
implementation supports multi-threading.
Possible values:
auto, 0
automatically select the number of threads to set
Default value is auto.
me_threshold integer (encoding,video)
Set motion estimation threshold.
mb_threshold integer (encoding,video)
Set macroblock threshold.
dc integer (encoding,video)
Set intra_dc_precision.
nssew integer (encoding,video)
Set nsse weight.
skip_top integer (decoding,video)
Set number of macroblock rows at the top which are skipped.
skip_bottom integer (decoding,video)
Set number of macroblock rows at the bottom which are skipped.
profile integer (encoding,audio,video)
Possible values:
unknown
aac_main
aac_low
aac_ssr
aac_ltp
aac_he
aac_he_v2
aac_ld
aac_eld
mpeg2_aac_low
mpeg2_aac_he
mpeg4_sp
mpeg4_core
mpeg4_main
mpeg4_asp
dts
dts_es
dts_96_24
dts_hd_hra
dts_hd_ma
level integer (encoding,audio,video)
Possible values:
unknown
lowres integer (decoding,audio,video)
Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.
skip_threshold integer (encoding,video)
Set frame skip threshold.
skip_factor integer (encoding,video)
Set frame skip factor.
skip_exp integer (encoding,video)
Set frame skip exponent. Negative values behave identical to the
corresponding positive ones, except that the score is normalized.
Positive values exist primarily for compatibility reasons and are
not so useful.
skipcmp integer (encoding,video)
Set frame skip compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
border_mask float (encoding,video)
Increase the quantizer for macroblocks close to borders.
mblmin integer (encoding,video)
Set min macroblock lagrange factor (VBR).
mblmax integer (encoding,video)
Set max macroblock lagrange factor (VBR).
mepc integer (encoding,video)
Set motion estimation bitrate penalty compensation (1.0 = 256).
skip_loop_filter integer (decoding,video)
skip_idct integer (decoding,video)
skip_frame integer (decoding,video)
Make decoder discard processing depending on the frame type
selected by the option value.
skip_loop_filter skips frame loop filtering, skip_idct skips frame
IDCT/dequantization, skip_frame skips decoding.
Possible values:
none
Discard no frame.
default
Discard useless frames like 0-sized frames.
noref
Discard all non-reference frames.
bidir
Discard all bidirectional frames.
nokey
Discard all frames excepts keyframes.
all Discard all frames.
Default value is default.
bidir_refine integer (encoding,video)
Refine the two motion vectors used in bidirectional macroblocks.
brd_scale integer (encoding,video)
Downscale frames for dynamic B-frame decision.
keyint_min integer (encoding,video)
Set minimum interval between IDR-frames.
refs integer (encoding,video)
Set reference frames to consider for motion compensation.
chromaoffset integer (encoding,video)
Set chroma qp offset from luma.
trellis integer (encoding,audio,video)
Set rate-distortion optimal quantization.
sc_factor integer (encoding,video)
Set value multiplied by qscale for each frame and added to
scene_change_score.
mv0_threshold integer (encoding,video)
b_sensitivity integer (encoding,video)
Adjust sensitivity of b_frame_strategy 1.
compression_level integer (encoding,audio,video)
min_prediction_order integer (encoding,audio)
max_prediction_order integer (encoding,audio)
timecode_frame_start integer (encoding,video)
Set GOP timecode frame start number, in non drop frame format.
request_channels integer (decoding,audio)
Set desired number of audio channels.
bits_per_raw_sample integer
channel_layout integer (decoding/encoding,audio)
Possible values:
request_channel_layout integer (decoding,audio)
Possible values:
rc_max_vbv_use float (encoding,video)
rc_min_vbv_use float (encoding,video)
ticks_per_frame integer (decoding/encoding,audio,video)
color_primaries integer (decoding/encoding,video)
Possible values:
bt709
BT.709
bt470m
BT.470 M
bt470bg
BT.470 BG
smpte170m
SMPTE 170 M
smpte240m
SMPTE 240 M
film
Film
bt2020
BT.2020
smpte428
smpte428_1
SMPTE ST 428-1
smpte431
SMPTE 431-2
smpte432
SMPTE 432-1
jedec-p22
JEDEC P22
color_trc integer (decoding/encoding,video)
Possible values:
bt709
BT.709
gamma22
BT.470 M
gamma28
BT.470 BG
smpte170m
SMPTE 170 M
smpte240m
SMPTE 240 M
linear
Linear
log
log100
Log
log_sqrt
log316
Log square root
iec61966_2_4
iec61966-2-4
IEC 61966-2-4
bt1361
bt1361e
BT.1361
iec61966_2_1
iec61966-2-1
IEC 61966-2-1
bt2020_10
bt2020_10bit
BT.2020 - 10 bit
bt2020_12
bt2020_12bit
BT.2020 - 12 bit
smpte2084
SMPTE ST 2084
smpte428
smpte428_1
SMPTE ST 428-1
arib-std-b67
ARIB STD-B67
colorspace integer (decoding/encoding,video)
Possible values:
rgb RGB
bt709
BT.709
fcc FCC
bt470bg
BT.470 BG
smpte170m
SMPTE 170 M
smpte240m
SMPTE 240 M
ycocg
YCOCG
bt2020nc
bt2020_ncl
BT.2020 NCL
bt2020c
bt2020_cl
BT.2020 CL
smpte2085
SMPTE 2085
color_range integer (decoding/encoding,video)
If used as input parameter, it serves as a hint to the decoder,
which color_range the input has. Possible values:
tv
mpeg
MPEG (219*2^(n-8))
pc
jpeg
JPEG (2^n-1)
chroma_sample_location integer (decoding/encoding,video)
Possible values:
left
center
topleft
top
bottomleft
bottom
log_level_offset integer
Set the log level offset.
slices integer (encoding,video)
Number of slices, used in parallelized encoding.
thread_type flags (decoding/encoding,video)
Select which multithreading methods to use.
Use of frame will increase decoding delay by one frame per thread,
so clients which cannot provide future frames should not use it.
Possible values:
slice
Decode more than one part of a single frame at once.
Multithreading using slices works only when the video was
encoded with slices.
frame
Decode more than one frame at once.
Default value is slice+frame.
audio_service_type integer (encoding,audio)
Set audio service type.
Possible values:
ma Main Audio Service
ef Effects
vi Visually Impaired
hi Hearing Impaired
di Dialogue
co Commentary
em Emergency
vo Voice Over
ka Karaoke
request_sample_fmt sample_fmt (decoding,audio)
Set sample format audio decoders should prefer. Default value is
"none".
pkt_timebase rational number
sub_charenc encoding (decoding,subtitles)
Set the input subtitles character encoding.
field_order field_order (video)
Set/override the field order of the video. Possible values:
progressive
Progressive video
tt Interlaced video, top field coded and displayed first
bb Interlaced video, bottom field coded and displayed first
tb Interlaced video, top coded first, bottom displayed first
bt Interlaced video, bottom coded first, top displayed first
skip_alpha bool (decoding,video)
Set to 1 to disable processing alpha (transparency). This works
like the gray flag in the flags option which skips chroma
information instead of alpha. Default is 0.
codec_whitelist list (input)
"," separated list of allowed decoders. By default all are allowed.
dump_separator string (input)
Separator used to separate the fields printed on the command line
about the Stream parameters. For example to separate the fields
with newlines and indention:
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
max_pixels integer (decoding/encoding,video)
Maximum number of pixels per image. This value can be used to avoid
out of memory failures due to large images.
apply_cropping bool (decoding,video)
Enable cropping if cropping parameters are multiples of the
required alignment for the left and top parameters. If the
alignment is not met the cropping will be partially applied to
maintain alignment. Default is 1 (enabled). Note: The required
alignment depends on if "AV_CODEC_FLAG_UNALIGNED" is set and the
CPU. "AV_CODEC_FLAG_UNALIGNED" cannot be changed from the command
line. Also hardware decoders will not apply left/top Cropping.
DECODERS
Decoders are configured elements in FFmpeg which allow the decoding of
multimedia streams.
When you configure your FFmpeg build, all the supported native decoders
are enabled by default. Decoders requiring an external library must be
enabled manually via the corresponding "--enable-lib" option. You can
list all available decoders using the configure option
"--list-decoders".
You can disable all the decoders with the configure option
"--disable-decoders" and selectively enable / disable single decoders
with the options "--enable-decoder=DECODER" /
"--disable-decoder=DECODER".
The option "-decoders" of the ff* tools will display the list of
enabled decoders.
VIDEO DECODERS
A description of some of the currently available video decoders
follows.
hevc
HEVC / H.265 decoder.
Note: the skip_loop_filter option has effect only at level "all".
rawvideo
Raw video decoder.
This decoder decodes rawvideo streams.
Options
top top_field_first
Specify the assumed field type of the input video.
-1 the video is assumed to be progressive (default)
0 bottom-field-first is assumed
1 top-field-first is assumed
AUDIO DECODERS
A description of some of the currently available audio decoders
follows.
ac3
AC-3 audio decoder.
This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as
well as the undocumented RealAudio 3 (a.k.a. dnet).
AC-3 Decoder Options
-drc_scale value
Dynamic Range Scale Factor. The factor to apply to dynamic range
values from the AC-3 stream. This factor is applied exponentially.
There are 3 notable scale factor ranges:
drc_scale == 0
DRC disabled. Produces full range audio.
0 < drc_scale <= 1
DRC enabled. Applies a fraction of the stream DRC value.
Audio reproduction is between full range and full compression.
drc_scale > 1
DRC enabled. Applies drc_scale asymmetrically. Loud sounds are
fully compressed. Soft sounds are enhanced.
flac
FLAC audio decoder.
This decoder aims to implement the complete FLAC specification from
Xiph.
FLAC Decoder options
-use_buggy_lpc
The lavc FLAC encoder used to produce buggy streams with high lpc
values (like the default value). This option makes it possible to
decode such streams correctly by using lavc's old buggy lpc logic
for decoding.
ffwavesynth
Internal wave synthesizer.
This decoder generates wave patterns according to predefined sequences.
Its use is purely internal and the format of the data it accepts is not
publicly documented.
libcelt
libcelt decoder wrapper.
libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio
codec. Requires the presence of the libcelt headers and library during
configuration. You need to explicitly configure the build with
"--enable-libcelt".
libgsm
libgsm decoder wrapper.
libgsm allows libavcodec to decode the GSM full rate audio codec.
Requires the presence of the libgsm headers and library during
configuration. You need to explicitly configure the build with
"--enable-libgsm".
This decoder supports both the ordinary GSM and the Microsoft variant.
libilbc
libilbc decoder wrapper.
libilbc allows libavcodec to decode the Internet Low Bitrate Codec
(iLBC) audio codec. Requires the presence of the libilbc headers and
library during configuration. You need to explicitly configure the
build with "--enable-libilbc".
Options
The following option is supported by the libilbc wrapper.
enhance
Enable the enhancement of the decoded audio when set to 1. The
default value is 0 (disabled).
libopencore-amrnb
libopencore-amrnb decoder wrapper.
libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
Narrowband audio codec. Using it requires the presence of the
libopencore-amrnb headers and library during configuration. You need to
explicitly configure the build with "--enable-libopencore-amrnb".
An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
without this library.
libopencore-amrwb
libopencore-amrwb decoder wrapper.
libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
Wideband audio codec. Using it requires the presence of the
libopencore-amrwb headers and library during configuration. You need to
explicitly configure the build with "--enable-libopencore-amrwb".
An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
without this library.
libopus
libopus decoder wrapper.
libopus allows libavcodec to decode the Opus Interactive Audio Codec.
Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
"--enable-libopus".
An FFmpeg native decoder for Opus exists, so users can decode Opus
without this library.
SUBTITLES DECODERS
dvbsub
Options
compute_clut
-1 Compute clut if no matching CLUT is in the stream.
0 Never compute CLUT
1 Always compute CLUT and override the one provided in the
stream.
dvb_substream
Selects the dvb substream, or all substreams if -1 which is
default.
dvdsub
This codec decodes the bitmap subtitles used in DVDs; the same
subtitles can also be found in VobSub file pairs and in some Matroska
files.
Options
palette
Specify the global palette used by the bitmaps. When stored in
VobSub, the palette is normally specified in the index file; in
Matroska, the palette is stored in the codec extra-data in the same
format as in VobSub. In DVDs, the palette is stored in the IFO
file, and therefore not available when reading from dumped VOB
files.
The format for this option is a string containing 16 24-bits
hexadecimal numbers (without 0x prefix) separated by comas, for
example "0d00ee, ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b,
0d617a, 7b7b7b, d1d1d1, 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c,
7c127b".
ifo_palette
Specify the IFO file from which the global palette is obtained.
(experimental)
forced_subs_only
Only decode subtitle entries marked as forced. Some titles have
forced and non-forced subtitles in the same track. Setting this
flag to 1 will only keep the forced subtitles. Default value is 0.
libzvbi-teletext
Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
subtitles. Requires the presence of the libzvbi headers and library
during configuration. You need to explicitly configure the build with
"--enable-libzvbi".
Options
txt_page
List of teletext page numbers to decode. You may use the special *
string to match all pages. Pages that do not match the specified
list are dropped. Default value is *.
txt_chop_top
Discards the top teletext line. Default value is 1.
txt_format
Specifies the format of the decoded subtitles. The teletext decoder
is capable of decoding the teletext pages to bitmaps or to simple
text, you should use "bitmap" for teletext pages, because certain
graphics and colors cannot be expressed in simple text. You might
use "text" for teletext based subtitles if your application can
handle simple text based subtitles. Default value is bitmap.
txt_left
X offset of generated bitmaps, default is 0.
txt_top
Y offset of generated bitmaps, default is 0.
txt_chop_spaces
Chops leading and trailing spaces and removes empty lines from the
generated text. This option is useful for teletext based subtitles
where empty spaces may be present at the start or at the end of the
lines or empty lines may be present between the subtitle lines
because of double-sized teletext characters. Default value is 1.
txt_duration
Sets the display duration of the decoded teletext pages or
subtitles in milliseconds. Default value is 30000 which is 30
seconds.
txt_transparent
Force transparent background of the generated teletext bitmaps.
Default value is 0 which means an opaque background.
txt_opacity
Sets the opacity (0-255) of the teletext background. If
txt_transparent is not set, it only affects characters between a
start box and an end box, typically subtitles. Default value is 0
if txt_transparent is set, 255 otherwise.
ENCODERS
Encoders are configured elements in FFmpeg which allow the encoding of
multimedia streams.
When you configure your FFmpeg build, all the supported native encoders
are enabled by default. Encoders requiring an external library must be
enabled manually via the corresponding "--enable-lib" option. You can
list all available encoders using the configure option
"--list-encoders".
You can disable all the encoders with the configure option
"--disable-encoders" and selectively enable / disable single encoders
with the options "--enable-encoder=ENCODER" /
"--disable-encoder=ENCODER".
The option "-encoders" of the ff* tools will display the list of
enabled encoders.
AUDIO ENCODERS
A description of some of the currently available audio encoders
follows.
aac
Advanced Audio Coding (AAC) encoder.
This encoder is the default AAC encoder, natively implemented into
FFmpeg. Its quality is on par or better than libfdk_aac at the default
bitrate of 128kbps. This encoder also implements more options,
profiles and samplerates than other encoders (with only the AAC-HE
profile pending to be implemented) so this encoder has become the
default and is the recommended choice.
Options
b Set bit rate in bits/s. Setting this automatically activates
constant bit rate (CBR) mode. If this option is unspecified it is
set to 128kbps.
q Set quality for variable bit rate (VBR) mode. This option is valid
only using the ffmpeg command-line tool. For library interface
users, use global_quality.
cutoff
Set cutoff frequency. If unspecified will allow the encoder to
dynamically adjust the cutoff to improve clarity on low bitrates.
aac_coder
Set AAC encoder coding method. Possible values:
twoloop
Two loop searching (TLS) method.
This method first sets quantizers depending on band thresholds
and then tries to find an optimal combination by adding or
subtracting a specific value from all quantizers and adjusting
some individual quantizer a little. Will tune itself based on
whether aac_is, aac_ms and aac_pns are enabled. This is the
default choice for a coder.
anmr
Average noise to mask ratio (ANMR) trellis-based solution.
This is an experimental coder which currently produces a lower
quality, is more unstable and is slower than the default
twoloop coder but has potential. Currently has no support for
the aac_is or aac_pns options. Not currently recommended.
fast
Constant quantizer method.
This method sets a constant quantizer for all bands. This is
the fastest of all the methods and has no rate control or
support for aac_is or aac_pns. Not recommended.
aac_ms
Sets mid/side coding mode. The default value of "auto" will
automatically use M/S with bands which will benefit from such
coding. Can be forced for all bands using the value "enable", which
is mainly useful for debugging or disabled using "disable".
aac_is
Sets intensity stereo coding tool usage. By default, it's enabled
and will automatically toggle IS for similar pairs of stereo bands
if it's beneficial. Can be disabled for debugging by setting the
value to "disable".
aac_pns
Uses perceptual noise substitution to replace low entropy high
frequency bands with imperceptible white noise during the decoding
process. By default, it's enabled, but can be disabled for
debugging purposes by using "disable".
aac_tns
Enables the use of a multitap FIR filter which spans through the
high frequency bands to hide quantization noise during the encoding
process and is reverted by the decoder. As well as decreasing
unpleasant artifacts in the high range this also reduces the
entropy in the high bands and allows for more bits to be used by
the mid-low bands. By default it's enabled but can be disabled for
debugging by setting the option to "disable".
aac_ltp
Enables the use of the long term prediction extension which
increases coding efficiency in very low bandwidth situations such
as encoding of voice or solo piano music by extending constant
harmonic peaks in bands throughout frames. This option is implied
by profile:a aac_low and is incompatible with aac_pred. Use in
conjunction with -ar to decrease the samplerate.
aac_pred
Enables the use of a more traditional style of prediction where the
spectral coefficients transmitted are replaced by the difference of
the current coefficients minus the previous "predicted"
coefficients. In theory and sometimes in practice this can improve
quality for low to mid bitrate audio. This option implies the
aac_main profile and is incompatible with aac_ltp.
profile
Sets the encoding profile, possible values:
aac_low
The default, AAC "Low-complexity" profile. Is the most
compatible and produces decent quality.
mpeg2_aac_low
Equivalent to "-profile:a aac_low -aac_pns 0". PNS was
introduced with the MPEG4 specifications.
aac_ltp
Long term prediction profile, is enabled by and will enable the
aac_ltp option. Introduced in MPEG4.
aac_main
Main-type prediction profile, is enabled by and will enable the
aac_pred option. Introduced in MPEG2.
If this option is unspecified it is set to aac_low.
ac3 and ac3_fixed
AC-3 audio encoders.
These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as
well as the undocumented RealAudio 3 (a.k.a. dnet).
The ac3 encoder uses floating-point math, while the ac3_fixed encoder
only uses fixed-point integer math. This does not mean that one is
always faster, just that one or the other may be better suited to a
particular system. The floating-point encoder will generally produce
better quality audio for a given bitrate. The ac3_fixed encoder is not
the default codec for any of the output formats, so it must be
specified explicitly using the option "-acodec ac3_fixed" in order to
use it.
AC-3 Metadata
The AC-3 metadata options are used to set parameters that describe the
audio, but in most cases do not affect the audio encoding itself. Some
of the options do directly affect or influence the decoding and
playback of the resulting bitstream, while others are just for
informational purposes. A few of the options will add bits to the
output stream that could otherwise be used for audio data, and will
thus affect the quality of the output. Those will be indicated
accordingly with a note in the option list below.
These parameters are described in detail in several publicly-available
documents.
*<<http://www.atsc.org/cms/standards/a_52-2010.pdf>>
*<<http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf>>
*<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf>>
*<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf>>
Metadata Control Options
-per_frame_metadata boolean
Allow Per-Frame Metadata. Specifies if the encoder should check for
changing metadata for each frame.
0 The metadata values set at initialization will be used for
every frame in the stream. (default)
1 Metadata values can be changed before encoding each frame.
Downmix Levels
-center_mixlev level
Center Mix Level. The amount of gain the decoder should apply to
the center channel when downmixing to stereo. This field will only
be written to the bitstream if a center channel is present. The
value is specified as a scale factor. There are 3 valid values:
0.707
Apply -3dB gain
0.595
Apply -4.5dB gain (default)
0.500
Apply -6dB gain
-surround_mixlev level
Surround Mix Level. The amount of gain the decoder should apply to
the surround channel(s) when downmixing to stereo. This field will
only be written to the bitstream if one or more surround channels
are present. The value is specified as a scale factor. There are 3
valid values:
0.707
Apply -3dB gain
0.500
Apply -6dB gain (default)
0.000
Silence Surround Channel(s)
Audio Production Information
Audio Production Information is optional information describing the
mixing environment. Either none or both of the fields are written to
the bitstream.
-mixing_level number
Mixing Level. Specifies peak sound pressure level (SPL) in the
production environment when the mix was mastered. Valid values are
80 to 111, or -1 for unknown or not indicated. The default value is
-1, but that value cannot be used if the Audio Production
Information is written to the bitstream. Therefore, if the
"room_type" option is not the default value, the "mixing_level"
option must not be -1.
-room_type type
Room Type. Describes the equalization used during the final mixing
session at the studio or on the dubbing stage. A large room is a
dubbing stage with the industry standard X-curve equalization; a
small room has flat equalization. This field will not be written
to the bitstream if both the "mixing_level" option and the
"room_type" option have the default values.
0
notindicated
Not Indicated (default)
1
large
Large Room
2
small
Small Room
Other Metadata Options
-copyright boolean
Copyright Indicator. Specifies whether a copyright exists for this
audio.
0
off No Copyright Exists (default)
1
on Copyright Exists
-dialnorm value
Dialogue Normalization. Indicates how far the average dialogue
level of the program is below digital 100% full scale (0 dBFS).
This parameter determines a level shift during audio reproduction
that sets the average volume of the dialogue to a preset level. The
goal is to match volume level between program sources. A value of
-31dB will result in no volume level change, relative to the source
volume, during audio reproduction. Valid values are whole numbers
in the range -31 to -1, with -31 being the default.
-dsur_mode mode
Dolby Surround Mode. Specifies whether the stereo signal uses Dolby
Surround (Pro Logic). This field will only be written to the
bitstream if the audio stream is stereo. Using this option does NOT
mean the encoder will actually apply Dolby Surround processing.
0
notindicated
Not Indicated (default)
1
off Not Dolby Surround Encoded
2
on Dolby Surround Encoded
-original boolean
Original Bit Stream Indicator. Specifies whether this audio is from
the original source and not a copy.
0
off Not Original Source
1
on Original Source (default)
Extended Bitstream Information
The extended bitstream options are part of the Alternate Bit Stream
Syntax as specified in Annex D of the A/52:2010 standard. It is grouped
into 2 parts. If any one parameter in a group is specified, all values
in that group will be written to the bitstream. Default values are
used for those that are written but have not been specified. If the
mixing levels are written, the decoder will use these values instead of
the ones specified in the "center_mixlev" and "surround_mixlev" options
if it supports the Alternate Bit Stream Syntax.
Extended Bitstream Information - Part 1
-dmix_mode mode
Preferred Stereo Downmix Mode. Allows the user to select either
Lt/Rt (Dolby Surround) or Lo/Ro (normal stereo) as the preferred
stereo downmix mode.
0
notindicated
Not Indicated (default)
1
ltrt
Lt/Rt Downmix Preferred
2
loro
Lo/Ro Downmix Preferred
-ltrt_cmixlev level
Lt/Rt Center Mix Level. The amount of gain the decoder should apply
to the center channel when downmixing to stereo in Lt/Rt mode.
1.414
Apply +3dB gain
1.189
Apply +1.5dB gain
1.000
Apply 0dB gain
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain (default)
0.500
Apply -6.0dB gain
0.000
Silence Center Channel
-ltrt_surmixlev level
Lt/Rt Surround Mix Level. The amount of gain the decoder should
apply to the surround channel(s) when downmixing to stereo in Lt/Rt
mode.
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain
0.500
Apply -6.0dB gain (default)
0.000
Silence Surround Channel(s)
-loro_cmixlev level
Lo/Ro Center Mix Level. The amount of gain the decoder should apply
to the center channel when downmixing to stereo in Lo/Ro mode.
1.414
Apply +3dB gain
1.189
Apply +1.5dB gain
1.000
Apply 0dB gain
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain (default)
0.500
Apply -6.0dB gain
0.000
Silence Center Channel
-loro_surmixlev level
Lo/Ro Surround Mix Level. The amount of gain the decoder should
apply to the surround channel(s) when downmixing to stereo in Lo/Ro
mode.
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain
0.500
Apply -6.0dB gain (default)
0.000
Silence Surround Channel(s)
Extended Bitstream Information - Part 2
-dsurex_mode mode
Dolby Surround EX Mode. Indicates whether the stream uses Dolby
Surround EX (7.1 matrixed to 5.1). Using this option does NOT mean
the encoder will actually apply Dolby Surround EX processing.
0
notindicated
Not Indicated (default)
1
on Dolby Surround EX Off
2
off Dolby Surround EX On
-dheadphone_mode mode
Dolby Headphone Mode. Indicates whether the stream uses Dolby
Headphone encoding (multi-channel matrixed to 2.0 for use with
headphones). Using this option does NOT mean the encoder will
actually apply Dolby Headphone processing.
0
notindicated
Not Indicated (default)
1
on Dolby Headphone Off
2
off Dolby Headphone On
-ad_conv_type type
A/D Converter Type. Indicates whether the audio has passed through
HDCD A/D conversion.
0
standard
Standard A/D Converter (default)
1
hdcd
HDCD A/D Converter
Other AC-3 Encoding Options
-stereo_rematrixing boolean
Stereo Rematrixing. Enables/Disables use of rematrixing for stereo
input. This is an optional AC-3 feature that increases quality by
selectively encoding the left/right channels as mid/side. This
option is enabled by default, and it is highly recommended that it
be left as enabled except for testing purposes.
cutoff frequency
Set lowpass cutoff frequency. If unspecified, the encoder selects a
default determined by various other encoding parameters.
Floating-Point-Only AC-3 Encoding Options
These options are only valid for the floating-point encoder and do not
exist for the fixed-point encoder due to the corresponding features not
being implemented in fixed-point.
-channel_coupling boolean
Enables/Disables use of channel coupling, which is an optional AC-3
feature that increases quality by combining high frequency
information from multiple channels into a single channel. The per-
channel high frequency information is sent with less accuracy in
both the frequency and time domains. This allows more bits to be
used for lower frequencies while preserving enough information to
reconstruct the high frequencies. This option is enabled by default
for the floating-point encoder and should generally be left as
enabled except for testing purposes or to increase encoding speed.
-1
auto
Selected by Encoder (default)
0
off Disable Channel Coupling
1
on Enable Channel Coupling
-cpl_start_band number
Coupling Start Band. Sets the channel coupling start band, from 1
to 15. If a value higher than the bandwidth is used, it will be
reduced to 1 less than the coupling end band. If auto is used, the
start band will be determined by the encoder based on the bit rate,
sample rate, and channel layout. This option has no effect if
channel coupling is disabled.
-1
auto
Selected by Encoder (default)
flac
FLAC (Free Lossless Audio Codec) Encoder
Options
The following options are supported by FFmpeg's flac encoder.
compression_level
Sets the compression level, which chooses defaults for many other
options if they are not set explicitly. Valid values are from 0 to
12, 5 is the default.
frame_size
Sets the size of the frames in samples per channel.
lpc_coeff_precision
Sets the LPC coefficient precision, valid values are from 1 to 15,
15 is the default.
lpc_type
Sets the first stage LPC algorithm
none
LPC is not used
fixed
fixed LPC coefficients
levinson
cholesky
lpc_passes
Number of passes to use for Cholesky factorization during LPC
analysis
min_partition_order
The minimum partition order
max_partition_order
The maximum partition order
prediction_order_method
estimation
2level
4level
8level
search
Bruteforce search
log
ch_mode
Channel mode
auto
The mode is chosen automatically for each frame
indep
Channels are independently coded
left_side
right_side
mid_side
exact_rice_parameters
Chooses if rice parameters are calculated exactly or approximately.
if set to 1 then they are chosen exactly, which slows the code down
slightly and improves compression slightly.
multi_dim_quant
Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC
algorithm is applied after the first stage to finetune the
coefficients. This is quite slow and slightly improves compression.
opus
Opus encoder.
This is a native FFmpeg encoder for the Opus format. Currently its in
development and only implements the CELT part of the codec. Its quality
is usually worse and at best is equal to the libopus encoder.
Options
b Set bit rate in bits/s. If unspecified it uses the number of
channels and the layout to make a good guess.
opus_delay
Sets the maximum delay in milliseconds. Lower delays than 20ms will
very quickly decrease quality.
libfdk_aac
libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.
The libfdk-aac library is based on the Fraunhofer FDK AAC code from the
Android project.
Requires the presence of the libfdk-aac headers and library during
configuration. You need to explicitly configure the build with
"--enable-libfdk-aac". The library is also incompatible with GPL, so if
you allow the use of GPL, you should configure with "--enable-gpl
--enable-nonfree --enable-libfdk-aac".
This encoder is considered to produce output on par or worse at 128kbps
to the the native FFmpeg AAC encoder but can often produce better
sounding audio at identical or lower bitrates and has support for the
AAC-HE profiles.
VBR encoding, enabled through the vbr or flags +qscale options, is
experimental and only works with some combinations of parameters.
Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3
or higher.
For more information see the fdk-aac project at
<http://sourceforge.net/p/opencore-amr/fdk-aac/>.
Options
The following options are mapped on the shared FFmpeg codec options.
b Set bit rate in bits/s. If the bitrate is not explicitly specified,
it is automatically set to a suitable value depending on the
selected profile.
In case VBR mode is enabled the option is ignored.
ar Set audio sampling rate (in Hz).
channels
Set the number of audio channels.
flags +qscale
Enable fixed quality, VBR (Variable Bit Rate) mode. Note that VBR
is implicitly enabled when the vbr value is positive.
cutoff
Set cutoff frequency. If not specified (or explicitly set to 0) it
will use a value automatically computed by the library. Default
value is 0.
profile
Set audio profile.
The following profiles are recognized:
aac_low
Low Complexity AAC (LC)
aac_he
High Efficiency AAC (HE-AAC)
aac_he_v2
High Efficiency AAC version 2 (HE-AACv2)
aac_ld
Low Delay AAC (LD)
aac_eld
Enhanced Low Delay AAC (ELD)
If not specified it is set to aac_low.
The following are private options of the libfdk_aac encoder.
afterburner
Enable afterburner feature if set to 1, disabled if set to 0. This
improves the quality but also the required processing power.
Default value is 1.
eld_sbr
Enable SBR (Spectral Band Replication) for ELD if set to 1,
disabled if set to 0.
Default value is 0.
signaling
Set SBR/PS signaling style.
It can assume one of the following values:
default
choose signaling implicitly (explicit hierarchical by default,
implicit if global header is disabled)
implicit
implicit backwards compatible signaling
explicit_sbr
explicit SBR, implicit PS signaling
explicit_hierarchical
explicit hierarchical signaling
Default value is default.
latm
Output LATM/LOAS encapsulated data if set to 1, disabled if set to
0.
Default value is 0.
header_period
Set StreamMuxConfig and PCE repetition period (in frames) for
sending in-band configuration buffers within LATM/LOAS transport
layer.
Must be a 16-bits non-negative integer.
Default value is 0.
vbr Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty
good) and 5 is highest quality. A value of 0 will disable VBR, and
CBR (Constant Bit Rate) is enabled.
Currently only the aac_low profile supports VBR encoding.
VBR modes 1-5 correspond to roughly the following average bit
rates:
1 32 kbps/channel
2 40 kbps/channel
3 48-56 kbps/channel
4 64 kbps/channel
5 about 80-96 kbps/channel
Default value is 0.
Examples
路 Use ffmpeg to convert an audio file to VBR AAC in an M4A (MP4)
container:
ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a
路 Use ffmpeg to convert an audio file to CBR 64k kbps AAC, using the
High-Efficiency AAC profile:
ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a
libmp3lame
LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper.
Requires the presence of the libmp3lame headers and library during
configuration. You need to explicitly configure the build with
"--enable-libmp3lame".
See libshine for a fixed-point MP3 encoder, although with a lower
quality.
Options
The following options are supported by the libmp3lame wrapper. The
lame-equivalent of the options are listed in parentheses.
b (-b)
Set bitrate expressed in bits/s for CBR or ABR. LAME "bitrate" is
expressed in kilobits/s.
q (-V)
Set constant quality setting for VBR. This option is valid only
using the ffmpeg command-line tool. For library interface users,
use global_quality.
compression_level (-q)
Set algorithm quality. Valid arguments are integers in the 0-9
range, with 0 meaning highest quality but slowest, and 9 meaning
fastest while producing the worst quality.
cutoff (--lowpass)
Set lowpass cutoff frequency. If unspecified, the encoder
dynamically adjusts the cutoff.
reservoir
Enable use of bit reservoir when set to 1. Default value is 1. LAME
has this enabled by default, but can be overridden by use --nores
option.
joint_stereo (-m j)
Enable the encoder to use (on a frame by frame basis) either L/R
stereo or mid/side stereo. Default value is 1.
abr (--abr)
Enable the encoder to use ABR when set to 1. The lame --abr sets
the target bitrate, while this options only tells FFmpeg to use ABR
still relies on b to set bitrate.
libopencore-amrnb
OpenCORE Adaptive Multi-Rate Narrowband encoder.
Requires the presence of the libopencore-amrnb headers and library
during configuration. You need to explicitly configure the build with
"--enable-libopencore-amrnb --enable-version3".
This is a mono-only encoder. Officially it only supports 8000Hz sample
rate, but you can override it by setting strict to unofficial or lower.
Options
b Set bitrate in bits per second. Only the following bitrates are
supported, otherwise libavcodec will round to the nearest valid
bitrate.
4750
5150
5900
6700
7400
7950
10200
12200
dtx Allow discontinuous transmission (generate comfort noise) when set
to 1. The default value is 0 (disabled).
libopus
libopus Opus Interactive Audio Codec encoder wrapper.
Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
"--enable-libopus".
Option Mapping
Most libopus options are modelled after the opusenc utility from opus-
tools. The following is an option mapping chart describing options
supported by the libopus wrapper, and their opusenc-equivalent in
parentheses.
b (bitrate)
Set the bit rate in bits/s. FFmpeg's b option is expressed in
bits/s, while opusenc's bitrate in kilobits/s.
vbr (vbr, hard-cbr, and cvbr)
Set VBR mode. The FFmpeg vbr option has the following valid
arguments, with the opusenc equivalent options in parentheses:
off (hard-cbr)
Use constant bit rate encoding.
on (vbr)
Use variable bit rate encoding (the default).
constrained (cvbr)
Use constrained variable bit rate encoding.
compression_level (comp)
Set encoding algorithm complexity. Valid options are integers in
the 0-10 range. 0 gives the fastest encodes but lower quality,
while 10 gives the highest quality but slowest encoding. The
default is 10.
frame_duration (framesize)
Set maximum frame size, or duration of a frame in milliseconds. The
argument must be exactly the following: 2.5, 5, 10, 20, 40, 60.
Smaller frame sizes achieve lower latency but less quality at a
given bitrate. Sizes greater than 20ms are only interesting at
fairly low bitrates. The default is 20ms.
packet_loss (expect-loss)
Set expected packet loss percentage. The default is 0.
application (N.A.)
Set intended application type. Valid options are listed below:
voip
Favor improved speech intelligibility.
audio
Favor faithfulness to the input (the default).
lowdelay
Restrict to only the lowest delay modes.
cutoff (N.A.)
Set cutoff bandwidth in Hz. The argument must be exactly one of the
following: 4000, 6000, 8000, 12000, or 20000, corresponding to
narrowband, mediumband, wideband, super wideband, and fullband
respectively. The default is 0 (cutoff disabled).
mapping_family (mapping_family)
Set channel mapping family to be used by the encoder. The default
value of -1 uses mapping family 0 for mono and stereo inputs, and
mapping family 1 otherwise. The default also disables the surround
masking and LFE bandwidth optimzations in libopus, and requires
that the input contains 8 channels or fewer.
Other values include 0 for mono and stereo, 1 for surround sound
with masking and LFE bandwidth optimizations, and 255 for
independent streams with an unspecified channel layout.
libshine
Shine Fixed-Point MP3 encoder wrapper.
Shine is a fixed-point MP3 encoder. It has a far better performance on
platforms without an FPU, e.g. armel CPUs, and some phones and tablets.
However, as it is more targeted on performance than quality, it is not
on par with LAME and other production-grade encoders quality-wise.
Also, according to the project's homepage, this encoder may not be free
of bugs as the code was written a long time ago and the project was
dead for at least 5 years.
This encoder only supports stereo and mono input. This is also CBR-
only.
The original project (last updated in early 2007) is at
<http://sourceforge.net/projects/libshine-fxp/>. We only support the
updated fork by the Savonet/Liquidsoap project at
<https://github.com/savonet/shine>.
Requires the presence of the libshine headers and library during
configuration. You need to explicitly configure the build with
"--enable-libshine".
See also libmp3lame.
Options
The following options are supported by the libshine wrapper. The
shineenc-equivalent of the options are listed in parentheses.
b (-b)
Set bitrate expressed in bits/s for CBR. shineenc -b option is
expressed in kilobits/s.
libtwolame
TwoLAME MP2 encoder wrapper.
Requires the presence of the libtwolame headers and library during
configuration. You need to explicitly configure the build with
"--enable-libtwolame".
Options
The following options are supported by the libtwolame wrapper. The
twolame-equivalent options follow the FFmpeg ones and are in
parentheses.
b (-b)
Set bitrate expressed in bits/s for CBR. twolame b option is
expressed in kilobits/s. Default value is 128k.
q (-V)
Set quality for experimental VBR support. Maximum value range is
from -50 to 50, useful range is from -10 to 10. The higher the
value, the better the quality. This option is valid only using the
ffmpeg command-line tool. For library interface users, use
global_quality.
mode (--mode)
Set the mode of the resulting audio. Possible values:
auto
Choose mode automatically based on the input. This is the
default.
stereo
Stereo
joint_stereo
Joint stereo
dual_channel
Dual channel
mono
Mono
psymodel (--psyc-mode)
Set psychoacoustic model to use in encoding. The argument must be
an integer between -1 and 4, inclusive. The higher the value, the
better the quality. The default value is 3.
energy_levels (--energy)
Enable energy levels extensions when set to 1. The default value is
0 (disabled).
error_protection (--protect)
Enable CRC error protection when set to 1. The default value is 0
(disabled).
copyright (--copyright)
Set MPEG audio copyright flag when set to 1. The default value is 0
(disabled).
original (--original)
Set MPEG audio original flag when set to 1. The default value is 0
(disabled).
libvo-amrwbenc
VisualOn Adaptive Multi-Rate Wideband encoder.
Requires the presence of the libvo-amrwbenc headers and library during
configuration. You need to explicitly configure the build with
"--enable-libvo-amrwbenc --enable-version3".
This is a mono-only encoder. Officially it only supports 16000Hz sample
rate, but you can override it by setting strict to unofficial or lower.
Options
b Set bitrate in bits/s. Only the following bitrates are supported,
otherwise libavcodec will round to the nearest valid bitrate.
6600
8850
12650
14250
15850
18250
19850
23050
23850
dtx Allow discontinuous transmission (generate comfort noise) when set
to 1. The default value is 0 (disabled).
libvorbis
libvorbis encoder wrapper.
Requires the presence of the libvorbisenc headers and library during
configuration. You need to explicitly configure the build with
"--enable-libvorbis".
Options
The following options are supported by the libvorbis wrapper. The
oggenc-equivalent of the options are listed in parentheses.
To get a more accurate and extensive documentation of the libvorbis
options, consult the libvorbisenc's and oggenc's documentations. See
<http://xiph.org/vorbis/>, <http://wiki.xiph.org/Vorbis-tools>, and
oggenc(1).
b (-b)
Set bitrate expressed in bits/s for ABR. oggenc -b is expressed in
kilobits/s.
q (-q)
Set constant quality setting for VBR. The value should be a float
number in the range of -1.0 to 10.0. The higher the value, the
better the quality. The default value is 3.0.
This option is valid only using the ffmpeg command-line tool. For
library interface users, use global_quality.
cutoff (--advanced-encode-option lowpass_frequency=N)
Set cutoff bandwidth in Hz, a value of 0 disables cutoff. oggenc's
related option is expressed in kHz. The default value is 0 (cutoff
disabled).
minrate (-m)
Set minimum bitrate expressed in bits/s. oggenc -m is expressed in
kilobits/s.
maxrate (-M)
Set maximum bitrate expressed in bits/s. oggenc -M is expressed in
kilobits/s. This only has effect on ABR mode.
iblock (--advanced-encode-option impulse_noisetune=N)
Set noise floor bias for impulse blocks. The value is a float
number from -15.0 to 0.0. A negative bias instructs the encoder to
pay special attention to the crispness of transients in the encoded
audio. The tradeoff for better transient response is a higher
bitrate.
libwavpack
A wrapper providing WavPack encoding through libwavpack.
Only lossless mode using 32-bit integer samples is supported currently.
Requires the presence of the libwavpack headers and library during
configuration. You need to explicitly configure the build with
"--enable-libwavpack".
Note that a libavcodec-native encoder for the WavPack codec exists so
users can encode audios with this codec without using this encoder. See
wavpackenc.
Options
wavpack command line utility's corresponding options are listed in
parentheses, if any.
frame_size (--blocksize)
Default is 32768.
compression_level
Set speed vs. compression tradeoff. Acceptable arguments are listed
below:
0 (-f)
Fast mode.
1 Normal (default) settings.
2 (-h)
High quality.
3 (-hh)
Very high quality.
4-8 (-hh -xEXTRAPROC)
Same as 3, but with extra processing enabled.
4 is the same as -x2 and 8 is the same as -x6.
mjpeg
Motion JPEG encoder.
Options
huffman
Set the huffman encoding strategy. Possible values:
default
Use the default huffman tables. This is the default strategy.
optimal
Compute and use optimal huffman tables.
wavpack
WavPack lossless audio encoder.
This is a libavcodec-native WavPack encoder. There is also an encoder
based on libwavpack, but there is virtually no reason to use that
encoder.
See also libwavpack.
Options
The equivalent options for wavpack command line utility are listed in
parentheses.
Shared options
The following shared options are effective for this encoder. Only
special notes about this particular encoder will be documented here.
For the general meaning of the options, see the Codec Options chapter.
frame_size (--blocksize)
For this encoder, the range for this option is between 128 and
131072. Default is automatically decided based on sample rate and
number of channel.
For the complete formula of calculating default, see
libavcodec/wavpackenc.c.
compression_level (-f, -h, -hh, and -x)
This option's syntax is consistent with libwavpack's.
Private options
joint_stereo (-j)
Set whether to enable joint stereo. Valid values are:
on (1)
Force mid/side audio encoding.
off (0)
Force left/right audio encoding.
auto
Let the encoder decide automatically.
optimize_mono
Set whether to enable optimization for mono. This option is only
effective for non-mono streams. Available values:
on enabled
off disabled
VIDEO ENCODERS
A description of some of the currently available video encoders
follows.
Hap
Vidvox Hap video encoder.
Options
format integer
Specifies the Hap format to encode.
hap
hap_alpha
hap_q
Default value is hap.
chunks integer
Specifies the number of chunks to split frames into, between 1 and
64. This permits multithreaded decoding of large frames,
potentially at the cost of data-rate. The encoder may modify this
value to divide frames evenly.
Default value is 1.
compressor integer
Specifies the second-stage compressor to use. If set to none,
chunks will be limited to 1, as chunked uncompressed frames offer
no benefit.
none
snappy
Default value is snappy.
jpeg2000
The native jpeg 2000 encoder is lossy by default, the "-q:v" option can
be used to set the encoding quality. Lossless encoding can be selected
with "-pred 1".
Options
format
Can be set to either "j2k" or "jp2" (the default) that makes it
possible to store non-rgb pix_fmts.
libkvazaar
Kvazaar H.265/HEVC encoder.
Requires the presence of the libkvazaar headers and library during
configuration. You need to explicitly configure the build with
--enable-libkvazaar.
Options
b Set target video bitrate in bit/s and enable rate control.
kvazaar-params
Set kvazaar parameters as a list of name=value pairs separated by
commas (,). See kvazaar documentation for a list of options.
libopenh264
Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.
This encoder requires the presence of the libopenh264 headers and
library during configuration. You need to explicitly configure the
build with "--enable-libopenh264". The library is detected using pkg-
config.
For more information about the library see <http://www.openh264.org>.
Options
The following FFmpeg global options affect the configurations of the
libopenh264 encoder.
b Set the bitrate (as a number of bits per second).
g Set the GOP size.
maxrate
Set the max bitrate (as a number of bits per second).
flags +global_header
Set global header in the bitstream.
slices
Set the number of slices, used in parallelized encoding. Default
value is 0. This is only used when slice_mode is set to fixed.
slice_mode
Set slice mode. Can assume one of the following possible values:
fixed
a fixed number of slices
rowmb
one slice per row of macroblocks
auto
automatic number of slices according to number of threads
dyn dynamic slicing
Default value is auto.
loopfilter
Enable loop filter, if set to 1 (automatically enabled). To disable
set a value of 0.
profile
Set profile restrictions. If set to the value of main enable CABAC
(set the "SEncParamExt.iEntropyCodingModeFlag" flag to 1).
max_nal_size
Set maximum NAL size in bytes.
allow_skip_frames
Allow skipping frames to hit the target bitrate if set to 1.
libtheora
libtheora Theora encoder wrapper.
Requires the presence of the libtheora headers and library during
configuration. You need to explicitly configure the build with
"--enable-libtheora".
For more information about the libtheora project see
<http://www.theora.org/>.
Options
The following global options are mapped to internal libtheora options
which affect the quality and the bitrate of the encoded stream.
b Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode.
In case VBR (Variable Bit Rate) mode is enabled this option is
ignored.
flags
Used to enable constant quality mode (VBR) encoding through the
qscale flag, and to enable the "pass1" and "pass2" modes.
g Set the GOP size.
global_quality
Set the global quality as an integer in lambda units.
Only relevant when VBR mode is enabled with "flags +qscale". The
value is converted to QP units by dividing it by "FF_QP2LAMBDA",
clipped in the [0 - 10] range, and then multiplied by 6.3 to get a
value in the native libtheora range [0-63]. A higher value
corresponds to a higher quality.
q Enable VBR mode when set to a non-negative value, and set constant
quality value as a double floating point value in QP units.
The value is clipped in the [0-10] range, and then multiplied by
6.3 to get a value in the native libtheora range [0-63].
This option is valid only using the ffmpeg command-line tool. For
library interface users, use global_quality.
Examples
路 Set maximum constant quality (VBR) encoding with ffmpeg:
ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg
路 Use ffmpeg to convert a CBR 1000 kbps Theora video stream:
ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
libvpx
VP8/VP9 format supported through libvpx.
Requires the presence of the libvpx headers and library during
configuration. You need to explicitly configure the build with
"--enable-libvpx".
Options
The following options are supported by the libvpx wrapper. The
vpxenc-equivalent options or values are listed in parentheses for easy
migration.
To reduce the duplication of documentation, only the private options
and some others requiring special attention are documented here. For
the documentation of the undocumented generic options, see the Codec
Options chapter.
To get more documentation of the libvpx options, invoke the command
ffmpeg -h encoder=libvpx, ffmpeg -h encoder=libvpx-vp9 or vpxenc
--help. Further information is available in the libvpx API
documentation.
b (target-bitrate)
Set bitrate in bits/s. Note that FFmpeg's b option is expressed in
bits/s, while vpxenc's target-bitrate is in kilobits/s.
g (kf-max-dist)
keyint_min (kf-min-dist)
qmin (min-q)
qmax (max-q)
bufsize (buf-sz, buf-optimal-sz)
Set ratecontrol buffer size (in bits). Note vpxenc's options are
specified in milliseconds, the libvpx wrapper converts this value
as follows: "buf-sz = bufsize * 1000 / bitrate", "buf-optimal-sz =
bufsize * 1000 / bitrate * 5 / 6".
rc_init_occupancy (buf-initial-sz)
Set number of bits which should be loaded into the rc buffer before
decoding starts. Note vpxenc's option is specified in milliseconds,
the libvpx wrapper converts this value as follows:
"rc_init_occupancy * 1000 / bitrate".
undershoot-pct
Set datarate undershoot (min) percentage of the target bitrate.
overshoot-pct
Set datarate overshoot (max) percentage of the target bitrate.
skip_threshold (drop-frame)
qcomp (bias-pct)
maxrate (maxsection-pct)
Set GOP max bitrate in bits/s. Note vpxenc's option is specified as
a percentage of the target bitrate, the libvpx wrapper converts
this value as follows: "(maxrate * 100 / bitrate)".
minrate (minsection-pct)
Set GOP min bitrate in bits/s. Note vpxenc's option is specified as
a percentage of the target bitrate, the libvpx wrapper converts
this value as follows: "(minrate * 100 / bitrate)".
minrate, maxrate, b end-usage=cbr
"(minrate == maxrate == bitrate)".
crf (end-usage=cq, cq-level)
tune (tune)
psnr (psnr)
ssim (ssim)
quality, deadline (deadline)
best
Use best quality deadline. Poorly named and quite slow, this
option should be avoided as it may give worse quality output
than good.
good
Use good quality deadline. This is a good trade-off between
speed and quality when used with the cpu-used option.
realtime
Use realtime quality deadline.
speed, cpu-used (cpu-used)
Set quality/speed ratio modifier. Higher values speed up the encode
at the cost of quality.
nr (noise-sensitivity)
static-thresh
Set a change threshold on blocks below which they will be skipped
by the encoder.
slices (token-parts)
Note that FFmpeg's slices option gives the total number of
partitions, while vpxenc's token-parts is given as
"log2(partitions)".
max-intra-rate
Set maximum I-frame bitrate as a percentage of the target bitrate.
A value of 0 means unlimited.
force_key_frames
"VPX_EFLAG_FORCE_KF"
Alternate reference frame related
auto-alt-ref
Enable use of alternate reference frames (2-pass only).
arnr-max-frames
Set altref noise reduction max frame count.
arnr-type
Set altref noise reduction filter type: backward, forward,
centered.
arnr-strength
Set altref noise reduction filter strength.
rc-lookahead, lag-in-frames (lag-in-frames)
Set number of frames to look ahead for frametype and
ratecontrol.
error-resilient
Enable error resiliency features.
VP9-specific options
lossless
Enable lossless mode.
tile-columns
Set number of tile columns to use. Note this is given as
"log2(tile_columns)". For example, 8 tile columns would be
requested by setting the tile-columns option to 3.
tile-rows
Set number of tile rows to use. Note this is given as
"log2(tile_rows)". For example, 4 tile rows would be requested
by setting the tile-rows option to 2.
frame-parallel
Enable frame parallel decodability features.
aq-mode
Set adaptive quantization mode (0: off (default), 1: variance
2: complexity, 3: cyclic refresh, 4: equator360).
colorspace color-space
Set input color space. The VP9 bitstream supports signaling the
following colorspaces:
rgb sRGB
bt709 bt709
unspecified unknown
bt470bg bt601
smpte170m smpte170
smpte240m smpte240
bt2020_ncl bt2020
row-mt boolean
Enable row based multi-threading.
For more information about libvpx see: <http://www.webmproject.org/>
libwebp
libwebp WebP Image encoder wrapper
libwebp is Google's official encoder for WebP images. It can encode in
either lossy or lossless mode. Lossy images are essentially a wrapper
around a VP8 frame. Lossless images are a separate codec developed by
Google.
Pixel Format
Currently, libwebp only supports YUV420 for lossy and RGB for lossless
due to limitations of the format and libwebp. Alpha is supported for
either mode. Because of API limitations, if RGB is passed in when
encoding lossy or YUV is passed in for encoding lossless, the pixel
format will automatically be converted using functions from libwebp.
This is not ideal and is done only for convenience.
Options
-lossless boolean
Enables/Disables use of lossless mode. Default is 0.
-compression_level integer
For lossy, this is a quality/speed tradeoff. Higher values give
better quality for a given size at the cost of increased encoding
time. For lossless, this is a size/speed tradeoff. Higher values
give smaller size at the cost of increased encoding time. More
specifically, it controls the number of extra algorithms and
compression tools used, and varies the combination of these tools.
This maps to the method option in libwebp. The valid range is 0 to
6. Default is 4.
-qscale float
For lossy encoding, this controls image quality, 0 to 100. For
lossless encoding, this controls the effort and time spent at
compressing more. The default value is 75. Note that for usage via
libavcodec, this option is called global_quality and must be
multiplied by FF_QP2LAMBDA.
-preset type
Configuration preset. This does some automatic settings based on
the general type of the image.
none
Do not use a preset.
default
Use the encoder default.
picture
Digital picture, like portrait, inner shot
photo
Outdoor photograph, with natural lighting
drawing
Hand or line drawing, with high-contrast details
icon
Small-sized colorful images
text
Text-like
libx264, libx264rgb
x264 H.264/MPEG-4 AVC encoder wrapper.
This encoder requires the presence of the libx264 headers and library
during configuration. You need to explicitly configure the build with
"--enable-libx264".
libx264 supports an impressive number of features, including 8x8 and
4x4 adaptive spatial transform, adaptive B-frame placement, CAVLC/CABAC
entropy coding, interlacing (MBAFF), lossless mode, psy optimizations
for detail retention (adaptive quantization, psy-RD, psy-trellis).
Many libx264 encoder options are mapped to FFmpeg global codec options,
while unique encoder options are provided through private options.
Additionally the x264opts and x264-params private options allows one to
pass a list of key=value tuples as accepted by the libx264
"x264_param_parse" function.
The x264 project website is at
<http://www.videolan.org/developers/x264.html>.
The libx264rgb encoder is the same as libx264, except it accepts packed
RGB pixel formats as input instead of YUV.
Supported Pixel Formats
x264 supports 8- to 10-bit color spaces. The exact bit depth is
controlled at x264's configure time. FFmpeg only supports one bit depth
in one particular build. In other words, it is not possible to build
one FFmpeg with multiple versions of x264 with different bit depths.
Options
The following options are supported by the libx264 wrapper. The
x264-equivalent options or values are listed in parentheses for easy
migration.
To reduce the duplication of documentation, only the private options
and some others requiring special attention are documented here. For
the documentation of the undocumented generic options, see the Codec
Options chapter.
To get a more accurate and extensive documentation of the libx264
options, invoke the command x264 --fullhelp or consult the libx264
documentation.
b (bitrate)
Set bitrate in bits/s. Note that FFmpeg's b option is expressed in
bits/s, while x264's bitrate is in kilobits/s.
bf (bframes)
g (keyint)
qmin (qpmin)
Minimum quantizer scale.
qmax (qpmax)
Maximum quantizer scale.
qdiff (qpstep)
Maximum difference between quantizer scales.
qblur (qblur)
Quantizer curve blur
qcomp (qcomp)
Quantizer curve compression factor
refs (ref)
Number of reference frames each P-frame can use. The range is from
0-16.
sc_threshold (scenecut)
Sets the threshold for the scene change detection.
trellis (trellis)
Performs Trellis quantization to increase efficiency. Enabled by
default.
nr (nr)
me_range (merange)
Maximum range of the motion search in pixels.
me_method (me)
Set motion estimation method. Possible values in the decreasing
order of speed:
dia (dia)
epzs (dia)
Diamond search with radius 1 (fastest). epzs is an alias for
dia.
hex (hex)
Hexagonal search with radius 2.
umh (umh)
Uneven multi-hexagon search.
esa (esa)
Exhaustive search.
tesa (tesa)
Hadamard exhaustive search (slowest).
forced-idr
Normally, when forcing a I-frame type, the encoder can select any
type of I-frame. This option forces it to choose an IDR-frame.
subq (subme)
Sub-pixel motion estimation method.
b_strategy (b-adapt)
Adaptive B-frame placement decision algorithm. Use only on first-
pass.
keyint_min (min-keyint)
Minimum GOP size.
coder
Set entropy encoder. Possible values:
ac Enable CABAC.
vlc Enable CAVLC and disable CABAC. It generates the same effect as
x264's --no-cabac option.
cmp Set full pixel motion estimation comparison algorithm. Possible
values:
chroma
Enable chroma in motion estimation.
sad Ignore chroma in motion estimation. It generates the same
effect as x264's --no-chroma-me option.
threads (threads)
Number of encoding threads.
thread_type
Set multithreading technique. Possible values:
slice
Slice-based multithreading. It generates the same effect as
x264's --sliced-threads option.
frame
Frame-based multithreading.
flags
Set encoding flags. It can be used to disable closed GOP and enable
open GOP by setting it to "-cgop". The result is similar to the
behavior of x264's --open-gop option.
rc_init_occupancy (vbv-init)
preset (preset)
Set the encoding preset.
tune (tune)
Set tuning of the encoding params.
profile (profile)
Set profile restrictions.
fastfirstpass
Enable fast settings when encoding first pass, when set to 1. When
set to 0, it has the same effect of x264's --slow-firstpass option.
crf (crf)
Set the quality for constant quality mode.
crf_max (crf-max)
In CRF mode, prevents VBV from lowering quality beyond this point.
qp (qp)
Set constant quantization rate control method parameter.
aq-mode (aq-mode)
Set AQ method. Possible values:
none (0)
Disabled.
variance (1)
Variance AQ (complexity mask).
autovariance (2)
Auto-variance AQ (experimental).
aq-strength (aq-strength)
Set AQ strength, reduce blocking and blurring in flat and textured
areas.
psy Use psychovisual optimizations when set to 1. When set to 0, it has
the same effect as x264's --no-psy option.
psy-rd (psy-rd)
Set strength of psychovisual optimization, in psy-rd:psy-trellis
format.
rc-lookahead (rc-lookahead)
Set number of frames to look ahead for frametype and ratecontrol.
weightb
Enable weighted prediction for B-frames when set to 1. When set to
0, it has the same effect as x264's --no-weightb option.
weightp (weightp)
Set weighted prediction method for P-frames. Possible values:
none (0)
Disabled
simple (1)
Enable only weighted refs
smart (2)
Enable both weighted refs and duplicates
ssim (ssim)
Enable calculation and printing SSIM stats after the encoding.
intra-refresh (intra-refresh)
Enable the use of Periodic Intra Refresh instead of IDR frames when
set to 1.
avcintra-class (class)
Configure the encoder to generate AVC-Intra. Valid values are
50,100 and 200
bluray-compat (bluray-compat)
Configure the encoder to be compatible with the bluray standard.
It is a shorthand for setting "bluray-compat=1 force-cfr=1".
b-bias (b-bias)
Set the influence on how often B-frames are used.
b-pyramid (b-pyramid)
Set method for keeping of some B-frames as references. Possible
values:
none (none)
Disabled.
strict (strict)
Strictly hierarchical pyramid.
normal (normal)
Non-strict (not Blu-ray compatible).
mixed-refs
Enable the use of one reference per partition, as opposed to one
reference per macroblock when set to 1. When set to 0, it has the
same effect as x264's --no-mixed-refs option.
8x8dct
Enable adaptive spatial transform (high profile 8x8 transform) when
set to 1. When set to 0, it has the same effect as x264's
--no-8x8dct option.
fast-pskip
Enable early SKIP detection on P-frames when set to 1. When set to
0, it has the same effect as x264's --no-fast-pskip option.
aud (aud)
Enable use of access unit delimiters when set to 1.
mbtree
Enable use macroblock tree ratecontrol when set to 1. When set to
0, it has the same effect as x264's --no-mbtree option.
deblock (deblock)
Set loop filter parameters, in alpha:beta form.
cplxblur (cplxblur)
Set fluctuations reduction in QP (before curve compression).
partitions (partitions)
Set partitions to consider as a comma-separated list of. Possible
values in the list:
p8x8
8x8 P-frame partition.
p4x4
4x4 P-frame partition.
b8x8
4x4 B-frame partition.
i8x8
8x8 I-frame partition.
i4x4
4x4 I-frame partition. (Enabling p4x4 requires p8x8 to be
enabled. Enabling i8x8 requires adaptive spatial transform
(8x8dct option) to be enabled.)
none (none)
Do not consider any partitions.
all (all)
Consider every partition.
direct-pred (direct)
Set direct MV prediction mode. Possible values:
none (none)
Disable MV prediction.
spatial (spatial)
Enable spatial predicting.
temporal (temporal)
Enable temporal predicting.
auto (auto)
Automatically decided.
slice-max-size (slice-max-size)
Set the limit of the size of each slice in bytes. If not specified
but RTP payload size (ps) is specified, that is used.
stats (stats)
Set the file name for multi-pass stats.
nal-hrd (nal-hrd)
Set signal HRD information (requires vbv-bufsize to be set).
Possible values:
none (none)
Disable HRD information signaling.
vbr (vbr)
Variable bit rate.
cbr (cbr)
Constant bit rate (not allowed in MP4 container).
x264opts (N.A.)
Set any x264 option, see x264 --fullhelp for a list.
Argument is a list of key=value couples separated by ":". In filter
and psy-rd options that use ":" as a separator themselves, use ","
instead. They accept it as well since long ago but this is kept
undocumented for some reason.
For example to specify libx264 encoding options with ffmpeg:
ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
a53cc boolean
Import closed captions (which must be ATSC compatible format) into
output. Only the mpeg2 and h264 decoders provide these. Default is
1 (on).
x264-params (N.A.)
Override the x264 configuration using a :-separated list of
key=value parameters.
This option is functionally the same as the x264opts, but is
duplicated for compatibility with the Libav fork.
For example to specify libx264 encoding options with ffmpeg:
ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\
cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\
no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT
Encoding ffpresets for common usages are provided so they can be used
with the general presets system (e.g. passing the pre option).
libx265
x265 H.265/HEVC encoder wrapper.
This encoder requires the presence of the libx265 headers and library
during configuration. You need to explicitly configure the build with
--enable-libx265.
Options
preset
Set the x265 preset.
tune
Set the x265 tune parameter.
forced-idr
Normally, when forcing a I-frame type, the encoder can select any
type of I-frame. This option forces it to choose an IDR-frame.
x265-params
Set x265 options using a list of key=value couples separated by
":". See x265 --help for a list of options.
For example to specify libx265 encoding options with -x265-params:
ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4
libxvid
Xvid MPEG-4 Part 2 encoder wrapper.
This encoder requires the presence of the libxvidcore headers and
library during configuration. You need to explicitly configure the
build with "--enable-libxvid --enable-gpl".
The native "mpeg4" encoder supports the MPEG-4 Part 2 format, so users
can encode to this format without this library.
Options
The following options are supported by the libxvid wrapper. Some of the
following options are listed but are not documented, and correspond to
shared codec options. See the Codec Options chapter for their
documentation. The other shared options which are not listed have no
effect for the libxvid encoder.
b
g
qmin
qmax
mpeg_quant
threads
bf
b_qfactor
b_qoffset
flags
Set specific encoding flags. Possible values:
mv4 Use four motion vector by macroblock.
aic Enable high quality AC prediction.
gray
Only encode grayscale.
gmc Enable the use of global motion compensation (GMC).
qpel
Enable quarter-pixel motion compensation.
cgop
Enable closed GOP.
global_header
Place global headers in extradata instead of every keyframe.
trellis
me_method
Set motion estimation method. Possible values in decreasing order
of speed and increasing order of quality:
zero
Use no motion estimation (default).
phods
x1
log Enable advanced diamond zonal search for 16x16 blocks and half-
pixel refinement for 16x16 blocks. x1 and log are aliases for
phods.
epzs
Enable all of the things described above, plus advanced diamond
zonal search for 8x8 blocks, half-pixel refinement for 8x8
blocks, and motion estimation on chroma planes.
full
Enable all of the things described above, plus extended 16x16
and 8x8 blocks search.
mbd Set macroblock decision algorithm. Possible values in the
increasing order of quality:
simple
Use macroblock comparing function algorithm (default).
bits
Enable rate distortion-based half pixel and quarter pixel
refinement for 16x16 blocks.
rd Enable all of the things described above, plus rate distortion-
based half pixel and quarter pixel refinement for 8x8 blocks,
and rate distortion-based search using square pattern.
lumi_aq
Enable lumi masking adaptive quantization when set to 1. Default is
0 (disabled).
variance_aq
Enable variance adaptive quantization when set to 1. Default is 0
(disabled).
When combined with lumi_aq, the resulting quality will not be
better than any of the two specified individually. In other words,
the resulting quality will be the worse one of the two effects.
ssim
Set structural similarity (SSIM) displaying method. Possible
values:
off Disable displaying of SSIM information.
avg Output average SSIM at the end of encoding to stdout. The
format of showing the average SSIM is:
Average SSIM: %f
For users who are not familiar with C, %f means a float number,
or a decimal (e.g. 0.939232).
frame
Output both per-frame SSIM data during encoding and average
SSIM at the end of encoding to stdout. The format of per-frame
information is:
SSIM: avg: %1.3f min: %1.3f max: %1.3f
For users who are not familiar with C, %1.3f means a float
number rounded to 3 digits after the dot (e.g. 0.932).
ssim_acc
Set SSIM accuracy. Valid options are integers within the range of
0-4, while 0 gives the most accurate result and 4 computes the
fastest.
mpeg2
MPEG-2 video encoder.
Options
seq_disp_ext integer
Specifies if the encoder should write a sequence_display_extension
to the output.
-1
auto
Decide automatically to write it or not (this is the default)
by checking if the data to be written is different from the
default or unspecified values.
0
never
Never write it.
1
always
Always write it.
png
PNG image encoder.
Private options
dpi integer
Set physical density of pixels, in dots per inch, unset by default
dpm integer
Set physical density of pixels, in dots per meter, unset by default
ProRes
Apple ProRes encoder.
FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
The used encoder can be chosen with the "-vcodec" option.
Private Options for prores-ks
profile integer
Select the ProRes profile to encode
proxy
lt
standard
hq
4444
4444xq
quant_mat integer
Select quantization matrix.
auto
default
proxy
lt
standard
hq
If set to auto, the matrix matching the profile will be picked. If
not set, the matrix providing the highest quality, default, will be
picked.
bits_per_mb integer
How many bits to allot for coding one macroblock. Different
profiles use between 200 and 2400 bits per macroblock, the maximum
is 8000.
mbs_per_slice integer
Number of macroblocks in each slice (1-8); the default value (8)
should be good in almost all situations.
vendor string
Override the 4-byte vendor ID. A custom vendor ID like apl0 would
claim the stream was produced by the Apple encoder.
alpha_bits integer
Specify number of bits for alpha component. Possible values are 0,
8 and 16. Use 0 to disable alpha plane coding.
Speed considerations
In the default mode of operation the encoder has to honor frame
constraints (i.e. not produce frames with size bigger than requested)
while still making output picture as good as possible. A frame
containing a lot of small details is harder to compress and the encoder
would spend more time searching for appropriate quantizers for each
slice.
Setting a higher bits_per_mb limit will improve the speed.
For the fastest encoding speed set the qscale parameter (4 is the
recommended value) and do not set a size constraint.
QSV encoders
The family of Intel QuickSync Video encoders (MPEG-2, H.264 and HEVC)
The ratecontrol method is selected as follows:
路 When global_quality is specified, a quality-based mode is used.
Specifically this means either
- CQP - constant quantizer scale, when the qscale codec flag is
also set (the -qscale ffmpeg option).
- LA_ICQ - intelligent constant quality with lookahead, when the
look_ahead option is also set.
- ICQ -- intelligent constant quality otherwise.
路 Otherwise, a bitrate-based mode is used. For all of those, you
should specify at least the desired average bitrate with the b
option.
- LA - VBR with lookahead, when the look_ahead option is
specified.
- VCM - video conferencing mode, when the vcm option is set.
- CBR - constant bitrate, when maxrate is specified and equal to
the average bitrate.
- VBR - variable bitrate, when maxrate is specified, but is
higher than the average bitrate.
- AVBR - average VBR mode, when maxrate is not specified. This
mode is further configured by the avbr_accuracy and
avbr_convergence options.
Note that depending on your system, a different mode than the one you
specified may be selected by the encoder. Set the verbosity level to
verbose or higher to see the actual settings used by the QSV runtime.
Additional libavcodec global options are mapped to MSDK options as
follows:
路 g/gop_size -> GopPicSize
路 bf/max_b_frames+1 -> GopRefDist
路 rc_init_occupancy/rc_initial_buffer_occupancy -> InitialDelayInKB
路 slices -> NumSlice
路 refs -> NumRefFrame
路 b_strategy/b_frame_strategy -> BRefType
路 cgop/CLOSED_GOP codec flag -> GopOptFlag
路 For the CQP mode, the i_qfactor/i_qoffset and b_qfactor/b_qoffset
set the difference between QPP and QPI, and QPP and QPB
respectively.
路 Setting the coder option to the value vlc will make the H.264
encoder use CAVLC instead of CABAC.
snow
Options
iterative_dia_size
dia size for the iterative motion estimation
VAAPI encoders
Wrappers for hardware encoders accessible via VAAPI.
These encoders only accept input in VAAPI hardware surfaces. If you
have input in software frames, use the hwupload filter to upload them
to the GPU.
The following standard libavcodec options are used:
路 g / gop_size
路 bf / max_b_frames
路 profile
路 level
路 b / bit_rate
路 maxrate / rc_max_rate
路 bufsize / rc_buffer_size
路 rc_init_occupancy / rc_initial_buffer_occupancy
路 compression_level
Speed / quality tradeoff: higher values are faster / worse quality.
路 q / global_quality
Size / quality tradeoff: higher values are smaller / worse quality.
路 qmin (only: qmax is not supported)
路 i_qfactor / i_quant_factor
路 i_qoffset / i_quant_offset
路 b_qfactor / b_quant_factor
路 b_qoffset / b_quant_offset
h264_vaapi
profile sets the value of profile_idc and the
constraint_set*_flags. level sets the value of level_idc.
low_power
Use low-power encoding mode.
coder
Set entropy encoder (default is cabac). Possible values:
ac
cabac
Use CABAC.
vlc
cavlc
Use CAVLC.
hevc_vaapi
profile and level set the values of general_profile_idc and
general_level_idc respectively.
mjpeg_vaapi
Always encodes using the standard quantisation and huffman tables -
global_quality scales the standard quantisation table (range
1-100).
mpeg2_vaapi
profile and level set the value of profile_and_level_indication.
No rate control is supported.
vp8_vaapi
B-frames are not supported.
global_quality sets the q_idx used for non-key frames (range
0-127).
loop_filter_level
loop_filter_sharpness
Manually set the loop filter parameters.
vp9_vaapi
global_quality sets the q_idx used for P-frames (range 0-255).
loop_filter_level
loop_filter_sharpness
Manually set the loop filter parameters.
B-frames are supported, but the output stream is always in encode
order rather than display order. If B-frames are enabled, it may
be necessary to use the vp9_raw_reorder bitstream filter to modify
the output stream to display frames in the correct order.
Only normal frames are produced - the vp9_superframe bitstream
filter may be required to produce a stream usable with all
decoders.
vc2
SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed
at professional broadcasting but since it supports yuv420, yuv422 and
yuv444 at 8 (limited range or full range), 10 or 12 bits, this makes it
suitable for other tasks which require low overhead and low compression
(like screen recording).
Options
b Sets target video bitrate. Usually that's around 1:6 of the
uncompressed video bitrate (e.g. for 1920x1080 50fps yuv422p10
that's around 400Mbps). Higher values (close to the uncompressed
bitrate) turn on lossless compression mode.
field_order
Enables field coding when set (e.g. to tt - top field first) for
interlaced inputs. Should increase compression with interlaced
content as it splits the fields and encodes each separately.
wavelet_depth
Sets the total amount of wavelet transforms to apply, between 1 and
5 (default). Lower values reduce compression and quality. Less
capable decoders may not be able to handle values of wavelet_depth
over 3.
wavelet_type
Sets the transform type. Currently only 5_3 (LeGall) and 9_7
(Deslauriers-Dubuc) are implemented, with 9_7 being the one with
better compression and thus is the default.
slice_width
slice_height
Sets the slice size for each slice. Larger values result in better
compression. For compatibility with other more limited decoders
use slice_width of 32 and slice_height of 8.
tolerance
Sets the undershoot tolerance of the rate control system in
percent. This is to prevent an expensive search from being run.
qm Sets the quantization matrix preset to use by default or when
wavelet_depth is set to 5
- default Uses the default quantization matrix from the
specifications, extended with values for the fifth level. This
provides a good balance between keeping detail and omitting
artifacts.
- flat Use a completely zeroed out quantization matrix. This
increases PSNR but might reduce perception. Use in bogus
benchmarks.
- color Reduces detail but attempts to preserve color at
extremely low bitrates.
SUBTITLES ENCODERS
dvdsub
This codec encodes the bitmap subtitle format that is used in DVDs.
Typically they are stored in VOBSUB file pairs (*.idx + *.sub), and
they can also be used in Matroska files.
Options
even_rows_fix
When set to 1, enable a work-around that makes the number of pixel
rows even in all subtitles. This fixes a problem with some players
that cut off the bottom row if the number is odd. The work-around
just adds a fully transparent row if needed. The overhead is low,
typically one byte per subtitle on average.
By default, this work-around is disabled.
BITSTREAM FILTERS
When you configure your FFmpeg build, all the supported bitstream
filters are enabled by default. You can list all available ones using
the configure option "--list-bsfs".
You can disable all the bitstream filters using the configure option
"--disable-bsfs", and selectively enable any bitstream filter using the
option "--enable-bsf=BSF", or you can disable a particular bitstream
filter using the option "--disable-bsf=BSF".
The option "-bsfs" of the ff* tools will display the list of all the
supported bitstream filters included in your build.
The ff* tools have a -bsf option applied per stream, taking a comma-
separated list of filters, whose parameters follow the filter name
after a '='.
ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT
Below is a description of the currently available bitstream filters,
with their parameters, if any.
aac_adtstoasc
Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration
bitstream.
This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS
header and removes the ADTS header.
This filter is required for example when copying an AAC stream from a
raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
to MOV/MP4 files and related formats such as 3GP or M4A. Please note
that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.
chomp
Remove zero padding at the end of a packet.
dca_core
Extract the core from a DCA/DTS stream, dropping extensions such as
DTS-HD.
dump_extra
Add extradata to the beginning of the filtered packets.
The additional argument specifies which packets should be filtered. It
accepts the values:
a add extradata to all key packets, but only if local_header is set
in the flags2 codec context field
k add extradata to all key packets
e add extradata to all packets
If not specified it is assumed k.
For example the following ffmpeg command forces a global header (thus
disabling individual packet headers) in the H.264 packets generated by
the "libx264" encoder, but corrects them by adding the header stored in
extradata to the key packets:
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
extract_extradata
Extract the in-band extradata.
Certain codecs allow the long-term headers (e.g. MPEG-2 sequence
headers, or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either "in-
band" (i.e. as a part of the bitstream containing the coded frames) or
"out of band" (e.g. on the container level). This latter form is called
"extradata" in FFmpeg terminology.
This bitstream filter detects the in-band headers and makes them
available as extradata.
remove
When this option is enabled, the long-term headers are removed from
the bitstream after extraction.
h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code
prefixed mode (as defined in the Annex B of the ITU-T H.264
specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format (muxer "mpegts").
For example to remux an MP4 file containing an H.264 stream to mpegts
format with ffmpeg, you can use the command:
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
Please note that this filter is auto-inserted for MPEG-TS (muxer
"mpegts") and raw H.264 (muxer "h264") output formats.
hevc_mp4toannexb
Convert an HEVC/H.265 bitstream from length prefixed mode to start code
prefixed mode (as defined in the Annex B of the ITU-T H.265
specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format (muxer "mpegts").
For example to remux an MP4 file containing an HEVC stream to mpegts
format with ffmpeg, you can use the command:
ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts
Please note that this filter is auto-inserted for MPEG-TS (muxer
"mpegts") and raw HEVC/H.265 (muxer "h265" or "hevc") output formats.
imxdump
Modifies the bitstream to fit in MOV and to be usable by the Final Cut
Pro decoder. This filter only applies to the mpeg2video codec, and is
likely not needed for Final Cut Pro 7 and newer with the appropriate
-tag:v.
For example, to remux 30 MB/sec NTSC IMX to MOV:
ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
mjpeg2jpeg
Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
MJPEG is a video codec wherein each video frame is essentially a JPEG
image. The individual frames can be extracted without loss, e.g. by
ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
Unfortunately, these chunks are incomplete JPEG images, because they
lack the DHT segment required for decoding. Quoting from
<http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml>:
Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
commented that "MJPEG, or at least the MJPEG in AVIs having the MJPG
fourcc, is restricted JPEG with a fixed -- and *omitted* -- Huffman
table. The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must
use basic Huffman encoding, not arithmetic or progressive. . . . You
can indeed extract the MJPEG frames and decode them with a regular JPEG
decoder, but you have to prepend the DHT segment to them, or else the
decoder won't have any idea how to decompress the data. The exact table
necessary is given in the OpenDML spec."
This bitstream filter patches the header of frames extracted from an
MJPEG stream (carrying the AVI1 header ID and lacking a DHT segment) to
produce fully qualified JPEG images.
ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
mjpegadump
Add an MJPEG A header to the bitstream, to enable decoding by
Quicktime.
mov2textsub
Extract a representable text file from MOV subtitles, stripping the
metadata header from each subtitle packet.
See also the text2movsub filter.
mp3decomp
Decompress non-standard compressed MP3 audio headers.
mpeg4_unpack_bframes
Unpack DivX-style packed B-frames.
DivX-style packed B-frames are not valid MPEG-4 and were only a
workaround for the broken Video for Windows subsystem. They use more
space, can cause minor AV sync issues, require more CPU power to decode
(unless the player has some decoded picture queue to compensate the
2,0,2,0 frame per packet style) and cause trouble if copied into a
standard container like mp4 or mpeg-ps/ts, because MPEG-4 decoders may
not be able to decode them, since they are not valid MPEG-4.
For example to fix an AVI file containing an MPEG-4 stream with DivX-
style packed B-frames using ffmpeg, you can use the command:
ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
noise
Damages the contents of packets or simply drops them without damaging
the container. Can be used for fuzzing or testing error
resilience/concealment.
Parameters:
amount
A numeral string, whose value is related to how often output bytes
will be modified. Therefore, values below or equal to 0 are
forbidden, and the lower the more frequent bytes will be modified,
with 1 meaning every byte is modified.
dropamount
A numeral string, whose value is related to how often packets will
be dropped. Therefore, values below or equal to 0 are forbidden,
and the lower the more frequent packets will be dropped, with 1
meaning every packet is dropped.
The following example applies the modification to every byte but does
not drop any packets.
ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
null
This bitstream filter passes the packets through unchanged.
remove_extra
Remove extradata from packets.
It accepts the following parameter:
freq
Set which frame types to remove extradata from.
k Remove extradata from non-keyframes only.
keyframe
Remove extradata from keyframes only.
e, all
Remove extradata from all frames.
text2movsub
Convert text subtitles to MOV subtitles (as used by the "mov_text"
codec) with metadata headers.
See also the mov2textsub filter.
vp9_superframe
Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This
fixes merging of split/segmented VP9 streams where the alt-ref frame
was split from its visible counterpart.
vp9_superframe_split
Split VP9 superframes into single frames.
vp9_raw_reorder
Given a VP9 stream with correct timestamps but possibly out of order,
insert additional show-existing-frame packets to correct the ordering.
FORMAT OPTIONS
The libavformat library provides some generic global options, which can
be set on all the muxers and demuxers. In addition each muxer or
demuxer 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 "AVFormatContext" options or
using the libavutil/opt.h API for programmatic use.
The list of supported options follows:
avioflags flags (input/output)
Possible values:
direct
Reduce buffering.
probesize integer (input)
Set probing size in bytes, i.e. the size of the data to analyze to
get stream information. A higher value will enable detecting more
information in case it is dispersed into the stream, but will
increase latency. Must be an integer not lesser than 32. It is
5000000 by default.
packetsize integer (output)
Set packet size.
fflags flags (input/output)
Set format flags.
Possible values:
ignidx
Ignore index.
fastseek
Enable fast, but inaccurate seeks for some formats.
genpts
Generate PTS.
nofillin
Do not fill in missing values that can be exactly calculated.
noparse
Disable AVParsers, this needs "+nofillin" too.
igndts
Ignore DTS.
discardcorrupt
Discard corrupted frames.
sortdts
Try to interleave output packets by DTS.
keepside
Do not merge side data.
latm
Enable RTP MP4A-LATM payload.
nobuffer
Reduce the latency introduced by optional buffering
bitexact
Only write platform-, build- and time-independent data. This
ensures that file and data checksums are reproducible and match
between platforms. Its primary use is for regression testing.
shortest
Stop muxing at the end of the shortest stream. It may be
needed to increase max_interleave_delta to avoid flushing the
longer streams before EOF.
seek2any integer (input)
Allow seeking to non-keyframes on demuxer level when supported if
set to 1. Default is 0.
analyzeduration integer (input)
Specify how many microseconds are analyzed to probe the input. A
higher value will enable detecting more accurate information, but
will increase latency. It defaults to 5,000,000 microseconds = 5
seconds.
cryptokey hexadecimal string (input)
Set decryption key.
indexmem integer (input)
Set max memory used for timestamp index (per stream).
rtbufsize integer (input)
Set max memory used for buffering real-time frames.
fdebug flags (input/output)
Print specific debug info.
Possible values:
ts
max_delay integer (input/output)
Set maximum muxing or demuxing delay in microseconds.
fpsprobesize integer (input)
Set number of frames used to probe fps.
audio_preload integer (output)
Set microseconds by which audio packets should be interleaved
earlier.
chunk_duration integer (output)
Set microseconds for each chunk.
chunk_size integer (output)
Set size in bytes for each chunk.
err_detect, f_err_detect flags (input)
Set error detection flags. "f_err_detect" is deprecated and should
be used only via the ffmpeg tool.
Possible values:
crccheck
Verify embedded CRCs.
bitstream
Detect bitstream specification deviations.
buffer
Detect improper bitstream length.
explode
Abort decoding on minor error detection.
careful
Consider things that violate the spec and have not been seen in
the wild as errors.
compliant
Consider all spec non compliancies as errors.
aggressive
Consider things that a sane encoder should not do as an error.
max_interleave_delta integer (output)
Set maximum buffering duration for interleaving. The duration is
expressed in microseconds, and defaults to 1000000 (1 second).
To ensure all the streams are interleaved correctly, libavformat
will wait until it has at least one packet for each stream before
actually writing any packets to the output file. When some streams
are "sparse" (i.e. there are large gaps between successive
packets), this can result in excessive buffering.
This field specifies the maximum difference between the timestamps
of the first and the last packet in the muxing queue, above which
libavformat will output a packet regardless of whether it has
queued a packet for all the streams.
If set to 0, libavformat will continue buffering packets until it
has a packet for each stream, regardless of the maximum timestamp
difference between the buffered packets.
use_wallclock_as_timestamps integer (input)
Use wallclock as timestamps if set to 1. Default is 0.
avoid_negative_ts integer (output)
Possible values:
make_non_negative
Shift timestamps to make them non-negative. Also note that
this affects only leading negative timestamps, and not non-
monotonic negative timestamps.
make_zero
Shift timestamps so that the first timestamp is 0.
auto (default)
Enables shifting when required by the target format.
disabled
Disables shifting of timestamp.
When shifting is enabled, all output timestamps are shifted by the
same amount. Audio, video, and subtitles desynching and relative
timestamp differences are preserved compared to how they would have
been without shifting.
skip_initial_bytes integer (input)
Set number of bytes to skip before reading header and frames if set
to 1. Default is 0.
correct_ts_overflow integer (input)
Correct single timestamp overflows if set to 1. Default is 1.
flush_packets integer (output)
Flush the underlying I/O stream after each packet. Default is -1
(auto), which means that the underlying protocol will decide, 1
enables it, and has the effect of reducing the latency, 0 disables
it and may increase IO throughput in some cases.
output_ts_offset offset (output)
Set the output time offset.
offset must be a time duration specification, see the Time duration
section in the ffmpeg-utils(1) manual.
The offset is added by the muxer to the output timestamps.
Specifying a positive offset means that the corresponding streams
are delayed bt the time duration specified in offset. Default value
is 0 (meaning that no offset is applied).
format_whitelist list (input)
"," separated list of allowed demuxers. By default all are allowed.
dump_separator string (input)
Separator used to separate the fields printed on the command line
about the Stream parameters. For example to separate the fields
with newlines and indention:
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
max_streams integer (input)
Specifies the maximum number of streams. This can be used to reject
files that would require too many resources due to a large number
of streams.
Format stream specifiers
Format stream specifiers allow selection of one or more streams that
match specific properties.
Possible forms of stream specifiers are:
stream_index
Matches the stream with this index.
stream_type[:stream_index]
stream_type is one of following: 'v' for video, 'a' for audio, 's'
for subtitle, 'd' for data, and 't' for attachments. If
stream_index is given, then it matches the 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
Matches the stream by a format-specific ID.
The exact semantics of stream specifiers is defined by the
"avformat_match_stream_specifier()" function declared in the
libavformat/avformat.h header.
DEMUXERS
Demuxers are configured elements in FFmpeg that can read the multimedia
streams from a particular type of file.
When you configure your FFmpeg build, all the supported demuxers are
enabled by default. You can list all available ones using the configure
option "--list-demuxers".
You can disable all the demuxers using the configure option
"--disable-demuxers", and selectively enable a single demuxer with the
option "--enable-demuxer=DEMUXER", or disable it with the option
"--disable-demuxer=DEMUXER".
The option "-demuxers" of the ff* tools will display the list of
enabled demuxers. Use "-formats" to view a combined list of enabled
demuxers and muxers.
The description of some of the currently available demuxers follows.
aa
Audible Format 2, 3, and 4 demuxer.
This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
applehttp
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams. The id
field is set to the bitrate variant index number. By setting the
discard flags on AVStreams (by pressing 'a' or 'v' in ffplay), the
caller can decide which variant streams to actually receive. The total
bitrate of the variant that the stream belongs to is available in a
metadata key named "variant_bitrate".
apng
Animated Portable Network Graphics demuxer.
This demuxer is used to demux APNG files. All headers, but the PNG
signature, up to (but not including) the first fcTL chunk are
transmitted as extradata. Frames are then split as being all the
chunks between two fcTL ones, or between the last fcTL and IEND chunks.
-ignore_loop bool
Ignore the loop variable in the file if set.
-max_fps int
Maximum framerate in frames per second (0 for no limit).
-default_fps int
Default framerate in frames per second when none is specified in
the file (0 meaning as fast as possible).
asf
Advanced Systems Format demuxer.
This demuxer is used to demux ASF files and MMS network streams.
-no_resync_search bool
Do not try to resynchronize by looking for a certain optional start
code.
concat
Virtual concatenation script demuxer.
This demuxer reads a list of files and other directives from a text
file and demuxes them one after the other, as if all their packets had
been muxed together.
The timestamps in the files are adjusted so that the first file starts
at 0 and each next file starts where the previous one finishes. Note
that it is done globally and may cause gaps if all streams do not have
exactly the same length.
All files must have the same streams (same codecs, same time base,
etc.).
The duration of each file is used to adjust the timestamps of the next
file: if the duration is incorrect (because it was computed using the
bit-rate or because the file is truncated, for example), it can cause
artifacts. The "duration" directive can be used to override the
duration stored in each file.
Syntax
The script is a text file in extended-ASCII, with one directive per
line. Empty lines, leading spaces and lines starting with '#' are
ignored. The following directive is recognized:
"file path"
Path to a file to read; special characters and spaces must be
escaped with backslash or single quotes.
All subsequent file-related directives apply to that file.
"ffconcat version 1.0"
Identify the script type and version. It also sets the safe option
to 1 if it was -1.
To make FFmpeg recognize the format automatically, this directive
must appear exactly as is (no extra space or byte-order-mark) on
the very first line of the script.
"duration dur"
Duration of the file. This information can be specified from the
file; specifying it here may be more efficient or help if the
information from the file is not available or accurate.
If the duration is set for all files, then it is possible to seek
in the whole concatenated video.
"inpoint timestamp"
In point of the file. When the demuxer opens the file it instantly
seeks to the specified timestamp. Seeking is done so that all
streams can be presented successfully at In point.
This directive works best with intra frame codecs, because for non-
intra frame ones you will usually get extra packets before the
actual In point and the decoded content will most likely contain
frames before In point too.
For each file, packets before the file In point will have
timestamps less than the calculated start timestamp of the file
(negative in case of the first file), and the duration of the files
(if not specified by the "duration" directive) will be reduced
based on their specified In point.
Because of potential packets before the specified In point, packet
timestamps may overlap between two concatenated files.
"outpoint timestamp"
Out point of the file. When the demuxer reaches the specified
decoding timestamp in any of the streams, it handles it as an end
of file condition and skips the current and all the remaining
packets from all streams.
Out point is exclusive, which means that the demuxer will not
output packets with a decoding timestamp greater or equal to Out
point.
This directive works best with intra frame codecs and formats where
all streams are tightly interleaved. For non-intra frame codecs you
will usually get additional packets with presentation timestamp
after Out point therefore the decoded content will most likely
contain frames after Out point too. If your streams are not tightly
interleaved you may not get all the packets from all streams before
Out point and you may only will be able to decode the earliest
stream until Out point.
The duration of the files (if not specified by the "duration"
directive) will be reduced based on their specified Out point.
"file_packet_metadata key=value"
Metadata of the packets of the file. The specified metadata will be
set for each file packet. You can specify this directive multiple
times to add multiple metadata entries.
"stream"
Introduce a stream in the virtual file. All subsequent stream-
related directives apply to the last introduced stream. Some
streams properties must be set in order to allow identifying the
matching streams in the subfiles. If no streams are defined in the
script, the streams from the first file are copied.
"exact_stream_id id"
Set the id of the stream. If this directive is given, the string
with the corresponding id in the subfiles will be used. This is
especially useful for MPEG-PS (VOB) files, where the order of the
streams is not reliable.
Options
This demuxer accepts the following option:
safe
If set to 1, reject unsafe file paths. A file path is considered
safe if it does not contain a protocol specification and is
relative and all components only contain characters from the
portable character set (letters, digits, period, underscore and
hyphen) and have no period at the beginning of a component.
If set to 0, any file name is accepted.
The default is 1.
-1 is equivalent to 1 if the format was automatically probed and 0
otherwise.
auto_convert
If set to 1, try to perform automatic conversions on packet data to
make the streams concatenable. The default is 1.
Currently, the only conversion is adding the h264_mp4toannexb
bitstream filter to H.264 streams in MP4 format. This is necessary
in particular if there are resolution changes.
segment_time_metadata
If set to 1, every packet will contain the lavf.concat.start_time
and the lavf.concat.duration packet metadata values which are the
start_time and the duration of the respective file segments in the
concatenated output expressed in microseconds. The duration
metadata is only set if it is known based on the concat file. The
default is 0.
Examples
路 Use absolute filenames and include some comments:
# my first filename
file /mnt/share/file-1.wav
# my second filename including whitespace
file '/mnt/share/file 2.wav'
# my third filename including whitespace plus single quote
file '/mnt/share/file 3'\''.wav'
路 Allow for input format auto-probing, use safe filenames and set the
duration of the first file:
ffconcat version 1.0
file file-1.wav
duration 20.0
file subdir/file-2.wav
flv, live_flv
Adobe Flash Video Format demuxer.
This demuxer is used to demux FLV files and RTMP network streams. In
case of live network streams, if you force format, you may use live_flv
option instead of flv to survive timestamp discontinuities.
ffmpeg -f flv -i myfile.flv ...
ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
-flv_metadata bool
Allocate the streams according to the onMetaData array content.
gif
Animated GIF demuxer.
It accepts the following options:
min_delay
Set the minimum valid delay between frames in hundredths of
seconds. Range is 0 to 6000. Default value is 2.
max_gif_delay
Set the maximum valid delay between frames in hundredth of seconds.
Range is 0 to 65535. Default value is 65535 (nearly eleven
minutes), the maximum value allowed by the specification.
default_delay
Set the default delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 10.
ignore_loop
GIF files can contain information to loop a certain number of times
(or infinitely). If ignore_loop is set to 1, then the loop setting
from the input will be ignored and looping will not occur. If set
to 0, then looping will occur and will cycle the number of times
according to the GIF. Default value is 1.
For example, with the overlay filter, place an infinitely looping GIF
over another video:
ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
Note that in the above example the shortest option for overlay filter
is used to end the output video at the length of the shortest input
file, which in this case is input.mp4 as the GIF in this example loops
infinitely.
hls
HLS demuxer
It accepts the following options:
live_start_index
segment index to start live streams at (negative values are from
the end).
allowed_extensions
',' separated list of file extensions that hls is allowed to
access.
max_reload
Maximum number of times a insufficient list is attempted to be
reloaded. Default value is 1000.
image2
Image file demuxer.
This demuxer reads from a list of image files specified by a pattern.
The syntax and meaning of the pattern is specified by the option
pattern_type.
The pattern may contain a suffix which is used to automatically
determine the format of the images contained in the files.
The size, the pixel format, and the format of each image must be the
same for all the files in the sequence.
This demuxer accepts the following options:
framerate
Set the frame rate for the video stream. It defaults to 25.
loop
If set to 1, loop over the input. Default value is 0.
pattern_type
Select the pattern type used to interpret the provided filename.
pattern_type accepts one of the following values.
none
Disable pattern matching, therefore the video will only contain
the specified image. You should use this option if you do not
want to create sequences from multiple images and your
filenames may contain special pattern characters.
sequence
Select a sequence pattern type, used to specify a sequence of
files indexed by sequential numbers.
A sequence pattern may contain the string "%d" or "%0Nd", which
specifies the position of the characters representing a
sequential number in each filename matched by the pattern. If
the form "%d0Nd" is used, the string representing the number in
each filename is 0-padded and N is the total number of 0-padded
digits representing the number. The literal character '%' can
be specified in the pattern with the string "%%".
If the sequence pattern contains "%d" or "%0Nd", the first
filename of the file list specified by the pattern must contain
a number inclusively contained between start_number and
start_number+start_number_range-1, and all the following
numbers must be sequential.
For example the pattern "img-%03d.bmp" will match a sequence of
filenames of the form img-001.bmp, img-002.bmp, ...,
img-010.bmp, etc.; the pattern "i%%m%%g-%d.jpg" will match a
sequence of filenames of the form i%m%g-1.jpg, i%m%g-2.jpg,
..., i%m%g-10.jpg, etc.
Note that the pattern must not necessarily contain "%d" or
"%0Nd", for example to convert a single image file img.jpeg you
can employ the command:
ffmpeg -i img.jpeg img.png
glob
Select a glob wildcard pattern type.
The pattern is interpreted like a "glob()" pattern. This is
only selectable if libavformat was compiled with globbing
support.
glob_sequence (deprecated, will be removed)
Select a mixed glob wildcard/sequence pattern.
If your version of libavformat was compiled with globbing
support, and the provided pattern contains at least one glob
meta character among "%*?[]{}" that is preceded by an unescaped
"%", the pattern is interpreted like a "glob()" pattern,
otherwise it is interpreted like a sequence pattern.
All glob special characters "%*?[]{}" must be prefixed with
"%". To escape a literal "%" you shall use "%%".
For example the pattern "foo-%*.jpeg" will match all the
filenames prefixed by "foo-" and terminating with ".jpeg", and
"foo-%?%?%?.jpeg" will match all the filenames prefixed with
"foo-", followed by a sequence of three characters, and
terminating with ".jpeg".
This pattern type is deprecated in favor of glob and sequence.
Default value is glob_sequence.
pixel_format
Set the pixel format of the images to read. If not specified the
pixel format is guessed from the first image file in the sequence.
start_number
Set the index of the file matched by the image file pattern to
start to read from. Default value is 0.
start_number_range
Set the index interval range to check when looking for the first
image file in the sequence, starting from start_number. Default
value is 5.
ts_from_file
If set to 1, will set frame timestamp to modification time of image
file. Note that monotonity of timestamps is not provided: images go
in the same order as without this option. Default value is 0. If
set to 2, will set frame timestamp to the modification time of the
image file in nanosecond precision.
video_size
Set the video size of the images to read. If not specified the
video size is guessed from the first image file in the sequence.
Examples
路 Use ffmpeg for creating a video from the images in the file
sequence img-001.jpeg, img-002.jpeg, ..., assuming an input frame
rate of 10 frames per second:
ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
路 As above, but start by reading from a file with index 100 in the
sequence:
ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
路 Read images matching the "*.png" glob pattern , that is all the
files terminating with the ".png" suffix:
ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
libgme
The Game Music Emu library is a collection of video game music file
emulators.
See <http://code.google.com/p/game-music-emu/> for more information.
Some files have multiple tracks. The demuxer will pick the first track
by default. The track_index option can be used to select a different
track. Track indexes start at 0. The demuxer exports the number of
tracks as tracks meta data entry.
For very large files, the max_size option may have to be adjusted.
libopenmpt
libopenmpt based module demuxer
See <https://lib.openmpt.org/libopenmpt/> for more information.
Some files have multiple subsongs (tracks) this can be set with the
subsong option.
It accepts the following options:
subsong
Set the subsong index. This can be either 'all', 'auto', or the
index of the subsong. Subsong indexes start at 0. The default is
'auto'.
The default value is to let libopenmpt choose.
layout
Set the channel layout. Valid values are 1, 2, and 4 channel
layouts. The default value is STEREO.
sample_rate
Set the sample rate for libopenmpt to output. Range is from 1000
to INT_MAX. The value default is 48000.
mov/mp4/3gp/QuickTime
QuickTime / MP4 demuxer.
This demuxer accepts the following options:
enable_drefs
Enable loading of external tracks, disabled by default. Enabling
this can theoretically leak information in some use cases.
use_absolute_path
Allows loading of external tracks via absolute paths, disabled by
default. Enabling this poses a security risk. It should only be
enabled if the source is known to be non malicious.
mpegts
MPEG-2 transport stream demuxer.
This demuxer accepts the following options:
resync_size
Set size limit for looking up a new synchronization. Default value
is 65536.
fix_teletext_pts
Override teletext packet PTS and DTS values with the timestamps
calculated from the PCR of the first program which the teletext
stream is part of and is not discarded. Default value is 1, set
this option to 0 if you want your teletext packet PTS and DTS
values untouched.
ts_packetsize
Output option carrying the raw packet size in bytes. Show the
detected raw packet size, cannot be set by the user.
scan_all_pmts
Scan and combine all PMTs. The value is an integer with value from
-1 to 1 (-1 means automatic setting, 1 means enabled, 0 means
disabled). Default value is -1.
mpjpeg
MJPEG encapsulated in multi-part MIME demuxer.
This demuxer allows reading of MJPEG, where each frame is represented
as a part of multipart/x-mixed-replace stream.
strict_mime_boundary
Default implementation applies a relaxed standard to multi-part
MIME boundary detection, to prevent regression with numerous
existing endpoints not generating a proper MIME MJPEG stream.
Turning this option on by setting it to 1 will result in a stricter
check of the boundary value.
rawvideo
Raw video demuxer.
This demuxer allows one to read raw video data. Since there is no
header specifying the assumed video parameters, the user must specify
them in order to be able to decode the data correctly.
This demuxer accepts the following options:
framerate
Set input video frame rate. Default value is 25.
pixel_format
Set the input video pixel format. Default value is "yuv420p".
video_size
Set the input video size. This value must be specified explicitly.
For example to read a rawvideo file input.raw with ffplay, assuming a
pixel format of "rgb24", a video size of "320x240", and a frame rate of
10 images per second, use the command:
ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
sbg
SBaGen script demuxer.
This demuxer reads the script language used by SBaGen
<http://uazu.net/sbagen/> to generate binaural beats sessions. A SBG
script looks like that:
-SE
a: 300-2.5/3 440+4.5/0
b: 300-2.5/0 440+4.5/3
off: -
NOW == a
+0:07:00 == b
+0:14:00 == a
+0:21:00 == b
+0:30:00 off
A SBG script can mix absolute and relative timestamps. If the script
uses either only absolute timestamps (including the script start time)
or only relative ones, then its layout is fixed, and the conversion is
straightforward. On the other hand, if the script mixes both kind of
timestamps, then the NOW reference for relative timestamps will be
taken from the current time of day at the time the script is read, and
the script layout will be frozen according to that reference. That
means that if the script is directly played, the actual times will
match the absolute timestamps up to the sound controller's clock
accuracy, but if the user somehow pauses the playback or seeks, all
times will be shifted accordingly.
tedcaptions
JSON captions used for <http://www.ted.com/>.
TED does not provide links to the captions, but they can be guessed
from the page. The file tools/bookmarklets.html from the FFmpeg source
tree contains a bookmarklet to expose them.
This demuxer accepts the following option:
start_time
Set the start time of the TED talk, in milliseconds. The default is
15000 (15s). It is used to sync the captions with the downloadable
videos, because they include a 15s intro.
Example: convert the captions to a format most players understand:
ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
MUXERS
Muxers are configured elements in FFmpeg which allow writing multimedia
streams to a particular type of file.
When you configure your FFmpeg build, all the supported muxers are
enabled by default. You can list all available muxers using the
configure option "--list-muxers".
You can disable all the muxers with the configure option
"--disable-muxers" and selectively enable / disable single muxers with
the options "--enable-muxer=MUXER" / "--disable-muxer=MUXER".
The option "-muxers" of the ff* tools will display the list of enabled
muxers. Use "-formats" to view a combined list of enabled demuxers and
muxers.
A description of some of the currently available muxers follows.
aiff
Audio Interchange File Format muxer.
Options
It accepts the following options:
write_id3v2
Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
id3v2_version
Select ID3v2 version to write. Currently only version 3 and 4 (aka.
ID3v2.3 and ID3v2.4) are supported. The default is version 4.
asf
Advanced Systems Format muxer.
Note that Windows Media Audio (wma) and Windows Media Video (wmv) use
this muxer too.
Options
It accepts the following options:
packet_size
Set the muxer packet size. By tuning this setting you may reduce
data fragmentation or muxer overhead depending on your source.
Default value is 3200, minimum is 100, maximum is 64k.
avi
Audio Video Interleaved muxer.
Options
It accepts the following options:
reserve_index_space
Reserve the specified amount of bytes for the OpenDML master index
of each stream within the file header. By default additional master
indexes are embedded within the data packets if there is no space
left in the first master index and are linked together as a chain
of indexes. This index structure can cause problems for some use
cases, e.g. third-party software strictly relying on the OpenDML
index specification or when file seeking is slow. Reserving enough
index space in the file header avoids these problems.
The required index space depends on the output file size and should
be about 16 bytes per gigabyte. When this option is omitted or set
to zero the necessary index space is guessed.
write_channel_mask
Write the channel layout mask into the audio stream header.
This option is enabled by default. Disabling the channel mask can
be useful in specific scenarios, e.g. when merging multiple audio
streams into one for compatibility with software that only supports
a single audio stream in AVI (see the "amerge" section in the
ffmpeg-filters manual).
chromaprint
Chromaprint fingerprinter
This muxer feeds audio data to the Chromaprint library, which generates
a fingerprint for the provided audio data. It takes a single signed
native-endian 16-bit raw audio stream.
Options
silence_threshold
Threshold for detecting silence, ranges from 0 to 32767. -1 for
default (required for use with the AcoustID service).
algorithm
Algorithm index to fingerprint with.
fp_format
Format to output the fingerprint as. Accepts the following options:
raw Binary raw fingerprint
compressed
Binary compressed fingerprint
base64
Base64 compressed fingerprint
crc
CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC of all the input audio
and video frames. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
CRC.
The output of the muxer consists of a single line of the form:
CRC=0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits
containing the CRC for all the decoded input frames.
See also the framecrc muxer.
Examples
For example to compute the CRC of the input, and store it in the file
out.crc:
ffmpeg -i INPUT -f crc out.crc
You can print the CRC to stdout with the command:
ffmpeg -i INPUT -f crc -
You can select the output format of each frame with ffmpeg by
specifying the audio and video codec and format. For example to compute
the CRC of the input audio converted to PCM unsigned 8-bit and the
input video converted to MPEG-2 video, use the command:
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
flv
Adobe Flash Video Format muxer.
This muxer accepts the following options:
flvflags flags
Possible values:
aac_seq_header_detect
Place AAC sequence header based on audio stream data.
no_sequence_end
Disable sequence end tag.
no_metadata
Disable metadata tag.
no_duration_filesize
Disable duration and filesize in metadata when they are equal
to zero at the end of stream. (Be used to non-seekable living
stream).
add_keyframe_index
Used to facilitate seeking; particularly for HTTP pseudo
streaming.
dash
Dynamic Adaptive Streaming over HTTP (DASH) muxer that creates segments
and manifest files according to the MPEG-DASH standard ISO/IEC
23009-1:2014.
For more information see:
路 ISO DASH Specification:
<http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip>
路 WebM DASH Specification:
<https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>
It creates a MPD manifest file and segment files for each stream.
The segment filename might contain pre-defined identifiers used with
SegmentTemplate as defined in section 5.3.9.4.4 of the standard.
Available identifiers are "$RepresentationID$", "$Number$",
"$Bandwidth$" and "$Time$".
ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264
-b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline
-profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0
-b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1
-window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a"
-f dash /path/to/out.mpd
-min_seg_duration microseconds
Set the segment length in microseconds.
-window_size size
Set the maximum number of segments kept in the manifest.
-extra_window_size size
Set the maximum number of segments kept outside of the manifest
before removing from disk.
-remove_at_exit remove
Enable (1) or disable (0) removal of all segments when finished.
-use_template template
Enable (1) or disable (0) use of SegmentTemplate instead of
SegmentList.
-use_timeline timeline
Enable (1) or disable (0) use of SegmentTimeline in
SegmentTemplate.
-single_file single_file
Enable (1) or disable (0) storing all segments in one file,
accessed using byte ranges.
-single_file_name file_name
DASH-templated name to be used for baseURL. Implies single_file set
to "1".
-init_seg_name init_name
DASH-templated name to used for the initialization segment. Default
is "init-stream$RepresentationID$.m4s"
-media_seg_name segment_name
DASH-templated name to used for the media segments. Default is
"chunk-stream$RepresentationID$-$Number%05d$.m4s"
-utc_timing_url utc_url
URL of the page that will return the UTC timestamp in ISO format.
Example: "https://time.akamai.com/?iso"
-adaptation_sets adaptation_sets
Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c
id=y,streams=d,e" with x and y being the IDs of the adaptation sets
and a,b,c,d and e are the indices of the mapped streams.
To map all video (or audio) streams to an AdaptationSet, "v" (or
"a") can be used as stream identifier instead of IDs.
When no assignment is defined, this defaults to an AdaptationSet
for each stream.
framecrc
Per-packet CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC for each audio and
video packet. By default audio frames are converted to signed 16-bit
raw audio and video frames to raw video before computing the CRC.
The output of the muxer consists of a line for each audio and video
packet of the form:
<stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, 0x<CRC>
CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of
the packet.
Examples
For example to compute the CRC of the audio and video frames in INPUT,
converted to raw audio and video packets, and store it in the file
out.crc:
ffmpeg -i INPUT -f framecrc out.crc
To print the information to stdout, use the command:
ffmpeg -i INPUT -f framecrc -
With ffmpeg, you can select the output format to which the audio and
video frames are encoded before computing the CRC for each packet by
specifying the audio and video codec. For example, to compute the CRC
of each decoded input audio frame converted to PCM unsigned 8-bit and
of each decoded input video frame converted to MPEG-2 video, use the
command:
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
See also the crc muxer.
framehash
Per-packet hash testing format.
This muxer computes and prints a cryptographic hash for each audio and
video packet. This can be used for packet-by-packet equality checks
without having to individually do a binary comparison on each.
By default audio frames are converted to signed 16-bit raw audio and
video frames to raw video before computing the hash, but the output of
explicit conversions to other codecs can also be used. It uses the
SHA-256 cryptographic hash function by default, but supports several
other algorithms.
The output of the muxer consists of a line for each audio and video
packet of the form:
<stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, <hash>
hash is a hexadecimal number representing the computed hash for the
packet.
hash algorithm
Use the cryptographic hash function specified by the string
algorithm. Supported values include "MD5", "murmur3", "RIPEMD128",
"RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256"
(default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32"
and "adler32".
Examples
To compute the SHA-256 hash of the audio and video frames in INPUT,
converted to raw audio and video packets, and store it in the file
out.sha256:
ffmpeg -i INPUT -f framehash out.sha256
To print the information to stdout, using the MD5 hash function, use
the command:
ffmpeg -i INPUT -f framehash -hash md5 -
See also the hash muxer.
framemd5
Per-packet MD5 testing format.
This is a variant of the framehash muxer. Unlike that muxer, it
defaults to using the MD5 hash function.
Examples
To compute the MD5 hash of the audio and video frames in INPUT,
converted to raw audio and video packets, and store it in the file
out.md5:
ffmpeg -i INPUT -f framemd5 out.md5
To print the information to stdout, use the command:
ffmpeg -i INPUT -f framemd5 -
See also the framehash and md5 muxers.
gif
Animated GIF muxer.
It accepts the following options:
loop
Set the number of times to loop the output. Use "-1" for no loop, 0
for looping indefinitely (default).
final_delay
Force the delay (expressed in centiseconds) after the last frame.
Each frame ends with a delay until the next frame. The default is
"-1", which is a special value to tell the muxer to re-use the
previous delay. In case of a loop, you might want to customize this
value to mark a pause for instance.
For example, to encode a gif looping 10 times, with a 5 seconds delay
between the loops:
ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif
Note 1: if you wish to extract the frames into separate GIF files, you
need to force the image2 muxer:
ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"
Note 2: the GIF format has a very large time base: the delay between
two frames can therefore not be smaller than one centi second.
hash
Hash testing format.
This muxer computes and prints a cryptographic hash of all the input
audio and video frames. This can be used for equality checks without
having to do a complete binary comparison.
By default audio frames are converted to signed 16-bit raw audio and
video frames to raw video before computing the hash, but the output of
explicit conversions to other codecs can also be used. Timestamps are
ignored. It uses the SHA-256 cryptographic hash function by default,
but supports several other algorithms.
The output of the muxer consists of a single line of the form:
algo=hash, where algo is a short string representing the hash function
used, and hash is a hexadecimal number representing the computed hash.
hash algorithm
Use the cryptographic hash function specified by the string
algorithm. Supported values include "MD5", "murmur3", "RIPEMD128",
"RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256"
(default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32"
and "adler32".
Examples
To compute the SHA-256 hash of the input converted to raw audio and
video, and store it in the file out.sha256:
ffmpeg -i INPUT -f hash out.sha256
To print an MD5 hash to stdout use the command:
ffmpeg -i INPUT -f hash -hash md5 -
See also the framehash muxer.
hls
Apple HTTP Live Streaming muxer that segments MPEG-TS according to the
HTTP Live Streaming (HLS) specification.
It creates a playlist file, and one or more segment files. The output
filename specifies the playlist filename.
By default, the muxer creates a file for each segment produced. These
files have the same name as the playlist, followed by a sequential
number and a .ts extension.
For example, to convert an input file with ffmpeg:
ffmpeg -i in.nut out.m3u8
This example will produce the playlist, out.m3u8, and segment files:
out0.ts, out1.ts, out2.ts, etc.
See also the segment muxer, which provides a more generic and flexible
implementation of a segmenter, and can be used to perform HLS
segmentation.
Options
This muxer supports the following options:
hls_init_time seconds
Set the initial target segment length in seconds. Default value is
0. Segment will be cut on the next key frame after this time has
passed on the first m3u8 list. After the initial playlist is
filled ffmpeg will cut segments at duration equal to "hls_time"
hls_time seconds
Set the target segment length in seconds. Default value is 2.
Segment will be cut on the next key frame after this time has
passed.
hls_list_size size
Set the maximum number of playlist entries. If set to 0 the list
file will contain all the segments. Default value is 5.
hls_ts_options options_list
Set output format options using a :-separated list of key=value
parameters. Values containing ":" special characters must be
escaped.
hls_wrap wrap
This is a deprecated option, you can use "hls_list_size" and
"hls_flags delete_segments" instead it
This option is useful to avoid to fill the disk with many segment
files, and limits the maximum number of segment files written to
disk to wrap.
hls_start_number_source
Start the playlist sequence number ("#EXT-X-MEDIA-SEQUENCE")
according to the specified source. Unless "hls_flags single_file"
is set, it also specifies source of starting sequence numbers of
segment and subtitle filenames. In any case, if "hls_flags
append_list" is set and read playlist sequence number is greater
than the specified start sequence number, then that value will be
used as start value.
It accepts the following values:
generic (default)
Set the starting sequence numbers according to start_number
option value.
epoch
The start number will be the seconds since epoch (1970-01-01
00:00:00)
datetime
The start number will be based on the current date/time as
YYYYmmddHHMMSS. e.g. 20161231235759.
start_number number
Start the playlist sequence number ("#EXT-X-MEDIA-SEQUENCE") from
the specified number when hls_start_number_source value is generic.
(This is the default case.) Unless "hls_flags single_file" is set,
it also specifies starting sequence numbers of segment and subtitle
filenames. Default value is 0.
hls_allow_cache allowcache
Explicitly set whether the client MAY (1) or MUST NOT (0) cache
media segments.
hls_base_url baseurl
Append baseurl to every entry in the playlist. Useful to generate
playlists with absolute paths.
Note that the playlist sequence number must be unique for each
segment and it is not to be confused with the segment filename
sequence number which can be cyclic, for example if the wrap option
is specified.
hls_segment_filename filename
Set the segment filename. Unless "hls_flags single_file" is set,
filename is used as a string format with the segment number:
ffmpeg -i in.nut -hls_segment_filename 'file%03d.ts' out.m3u8
This example will produce the playlist, out.m3u8, and segment
files: file000.ts, file001.ts, file002.ts, etc.
filename may contain full path or relative path specification, but
only the file name part without any path info will be contained in
the m3u8 segment list. Should a relative path be specified, the
path of the created segment files will be relative to the current
working directory. When use_localtime_mkdir is set, the whole
expanded value of filename will be written into the m3u8 segment
list.
use_localtime
Use strftime() on filename to expand the segment filename with
localtime. The segment number is also available in this mode, but
to use it, you need to specify second_level_segment_index hls_flag
and %%d will be the specifier.
ffmpeg -i in.nut -use_localtime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8
This example will produce the playlist, out.m3u8, and segment
files: file-20160215-1455569023.ts, file-20160215-1455569024.ts,
etc. Note: On some systems/environments, the %s specifier is not
available. See
"strftime()" documentation.
ffmpeg -i in.nut -use_localtime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8
This example will produce the playlist, out.m3u8, and segment
files: file-20160215-0001.ts, file-20160215-0002.ts, etc.
use_localtime_mkdir
Used together with -use_localtime, it will create all
subdirectories which is expanded in filename.
ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8
This example will create a directory 201560215 (if it does not
exist), and then produce the playlist, out.m3u8, and segment files:
20160215/file-20160215-1455569023.ts,
20160215/file-20160215-1455569024.ts, etc.
ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8
This example will create a directory hierarchy 2016/02/15 (if any
of them do not exist), and then produce the playlist, out.m3u8, and
segment files: 2016/02/15/file-20160215-1455569023.ts,
2016/02/15/file-20160215-1455569024.ts, etc.
hls_key_info_file key_info_file
Use the information in key_info_file for segment encryption. The
first line of key_info_file specifies the key URI written to the
playlist. The key URL is used to access the encryption key during
playback. The second line specifies the path to the key file used
to obtain the key during the encryption process. The key file is
read as a single packed array of 16 octets in binary format. The
optional third line specifies the initialization vector (IV) as a
hexadecimal string to be used instead of the segment sequence
number (default) for encryption. Changes to key_info_file will
result in segment encryption with the new key/IV and an entry in
the playlist for the new key URI/IV if "hls_flags periodic_rekey"
is enabled.
Key info file format:
<key URI>
<key file path>
<IV> (optional)
Example key URIs:
http://server/file.key
/path/to/file.key
file.key
Example key file paths:
file.key
/path/to/file.key
Example IV:
0123456789ABCDEF0123456789ABCDEF
Key info file example:
http://server/file.key
/path/to/file.key
0123456789ABCDEF0123456789ABCDEF
Example shell script:
#!/bin/sh
BASE_URL=${1:-'.'}
openssl rand 16 > file.key
echo $BASE_URL/file.key > file.keyinfo
echo file.key >> file.keyinfo
echo $(openssl rand -hex 16) >> file.keyinfo
ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
-hls_key_info_file file.keyinfo out.m3u8
-hls_enc enc
Enable (1) or disable (0) the AES128 encryption. When enabled
every segment generated is encrypted and the encryption key is
saved as playlist name.key.
-hls_enc_key key
Hex-coded 16byte key to encrypt the segments, by default it is
randomly generated.
-hls_enc_key_url keyurl
If set, keyurl is prepended instead of baseurl to the key filename
in the playlist.
-hls_enc_iv iv
Hex-coded 16byte initialization vector for every segment instead of
the autogenerated ones.
hls_segment_type flags
Possible values:
mpegts
If this flag is set, the hls segment files will format to
mpegts. the mpegts files is used in all hls versions.
fmp4
If this flag is set, the hls segment files will format to
fragment mp4 looks like dash. the fmp4 files is used in hls
after version 7.
hls_fmp4_init_filename filename
set filename to the fragment files header file, default filename is
init.mp4.
hls_flags flags
Possible values:
single_file
If this flag is set, the muxer will store all segments in a
single MPEG-TS file, and will use byte ranges in the playlist.
HLS playlists generated with this way will have the version
number 4. For example:
ffmpeg -i in.nut -hls_flags single_file out.m3u8
Will produce the playlist, out.m3u8, and a single segment file,
out.ts.
delete_segments
Segment files removed from the playlist are deleted after a
period of time equal to the duration of the segment plus the
duration of the playlist.
append_list
Append new segments into the end of old segment list, and
remove the "#EXT-X-ENDLIST" from the old segment list.
round_durations
Round the duration info in the playlist file segment info to
integer values, instead of using floating point.
discont_start
Add the "#EXT-X-DISCONTINUITY" tag to the playlist, before the
first segment's information.
omit_endlist
Do not append the "EXT-X-ENDLIST" tag at the end of the
playlist.
periodic_rekey
The file specified by "hls_key_info_file" will be checked
periodically and detect updates to the encryption info. Be sure
to replace this file atomically, including the file containing
the AES encryption key.
split_by_time
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. This flag should be
used with the "hls_time" option.
program_date_time
Generate "EXT-X-PROGRAM-DATE-TIME" tags.
second_level_segment_index
Makes it possible to use segment indexes as %%d in
hls_segment_filename expression besides date/time values when
use_localtime is on. To get fixed width numbers with trailing
zeroes, %%0xd format is available where x is the required
width.
second_level_segment_size
Makes it possible to use segment sizes (counted in bytes) as
%%s in hls_segment_filename expression besides date/time values
when use_localtime is on. To get fixed width numbers with
trailing zeroes, %%0xs format is available where x is the
required width.
second_level_segment_duration
Makes it possible to use segment duration (calculated in
microseconds) as %%t in hls_segment_filename expression besides
date/time values when use_localtime is on. To get fixed width
numbers with trailing zeroes, %%0xt format is available where x
is the required width.
ffmpeg -i sample.mpeg \
-f hls -hls_time 3 -hls_list_size 5 \
-hls_flags second_level_segment_index+second_level_segment_size+second_level_segment_duration \
-use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8
This will produce segments like this:
segment_20170102194334_0003_00122200_0000003000000.ts,
segment_20170102194334_0004_00120072_0000003000000.ts etc.
temp_file
Write segment data to filename.tmp and rename to filename only
once the segment is complete. A webserver serving up segments
can be configured to reject requests to *.tmp to prevent access
to in-progress segments before they have been added to the m3u8
playlist.
hls_playlist_type event
Emit "#EXT-X-PLAYLIST-TYPE:EVENT" in the m3u8 header. Forces
hls_list_size to 0; the playlist can only be appended to.
hls_playlist_type vod
Emit "#EXT-X-PLAYLIST-TYPE:VOD" in the m3u8 header. Forces
hls_list_size to 0; the playlist must not change.
method
Use the given HTTP method to create the hls files.
ffmpeg -re -i in.ts -f hls -method PUT http://example.com/live/out.m3u8
This example will upload all the mpegts segment files to the HTTP
server using the HTTP PUT method, and update the m3u8 files every
"refresh" times using the same method. Note that the HTTP server
must support the given method for uploading files.
http_user_agent
Override User-Agent field in HTTP header. Applicable only for HTTP
output.
ico
ICO file muxer.
Microsoft's icon file format (ICO) has some strict limitations that
should be noted:
路 Size cannot exceed 256 pixels in any dimension
路 Only BMP and PNG images can be stored
路 If a BMP image is used, it must be one of the following pixel
formats:
BMP Bit Depth FFmpeg Pixel Format
1bit pal8
4bit pal8
8bit pal8
16bit rgb555le
24bit bgr24
32bit bgra
路 If a BMP image is used, it must use the BITMAPINFOHEADER DIB header
路 If a PNG image is used, it must use the rgba pixel format
image2
Image file muxer.
The image file muxer writes video frames to image files.
The output filenames are specified by a pattern, which can be used to
produce sequentially numbered series of files. The pattern may contain
the string "%d" or "%0Nd", this string specifies the position of the
characters representing a numbering in the filenames. If the form
"%0Nd" is used, the string representing the number in each filename is
0-padded to N digits. The literal character '%' can be specified in the
pattern with the string "%%".
If the pattern contains "%d" or "%0Nd", the first filename of the file
list specified will contain the number 1, all the following numbers
will be sequential.
The pattern may contain a suffix which is used to automatically
determine the format of the image files to write.
For example the pattern "img-%03d.bmp" will specify a sequence of
filenames of the form img-001.bmp, img-002.bmp, ..., img-010.bmp, etc.
The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
form img%-1.jpg, img%-2.jpg, ..., img%-10.jpg, etc.
Examples
The following example shows how to use ffmpeg for creating a sequence
of files img-001.jpeg, img-002.jpeg, ..., taking one image every second
from the input video:
ffmpeg -i in.avi -vsync cfr -r 1 -f image2 'img-%03d.jpeg'
Note that with ffmpeg, if the format is not specified with the "-f"
option and the output filename specifies an image file format, the
image2 muxer is automatically selected, so the previous command can be
written as:
ffmpeg -i in.avi -vsync cfr -r 1 'img-%03d.jpeg'
Note also that the pattern must not necessarily contain "%d" or "%0Nd",
for example to create a single image file img.jpeg from the start of
the input video you can employ the command:
ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
The strftime option allows you to expand the filename with date and
time information. Check the documentation of the "strftime()" function
for the syntax.
For example to generate image files from the "strftime()"
"%Y-%m-%d_%H-%M-%S" pattern, the following ffmpeg command can be used:
ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
Options
start_number
Start the sequence from the specified number. Default value is 1.
update
If set to 1, the filename will always be interpreted as just a
filename, not a pattern, and the corresponding file will be
continuously overwritten with new images. Default value is 0.
strftime
If set to 1, expand the filename with date and time information
from "strftime()". Default value is 0.
The image muxer supports the .Y.U.V image file format. This format is
special in that that each image frame consists of three files, for each
of the YUV420P components. To read or write this image file format,
specify the name of the '.Y' file. The muxer will automatically open
the '.U' and '.V' files as required.
matroska
Matroska container muxer.
This muxer implements the matroska and webm container specs.
Metadata
The recognized metadata settings in this muxer are:
title
Set title name provided to a single track.
language
Specify the language of the track in the Matroska languages form.
The language can be either the 3 letters bibliographic ISO-639-2
(ISO 639-2/B) form (like "fre" for French), or a language code
mixed with a country code for specialities in languages (like "fre-
ca" for Canadian French).
stereo_mode
Set stereo 3D video layout of two views in a single video track.
The following values are recognized:
mono
video is not stereo
left_right
Both views are arranged side by side, Left-eye view is on the
left
bottom_top
Both views are arranged in top-bottom orientation, Left-eye
view is at bottom
top_bottom
Both views are arranged in top-bottom orientation, Left-eye
view is on top
checkerboard_rl
Each view is arranged in a checkerboard interleaved pattern,
Left-eye view being first
checkerboard_lr
Each view is arranged in a checkerboard interleaved pattern,
Right-eye view being first
row_interleaved_rl
Each view is constituted by a row based interleaving, Right-eye
view is first row
row_interleaved_lr
Each view is constituted by a row based interleaving, Left-eye
view is first row
col_interleaved_rl
Both views are arranged in a column based interleaving manner,
Right-eye view is first column
col_interleaved_lr
Both views are arranged in a column based interleaving manner,
Left-eye view is first column
anaglyph_cyan_red
All frames are in anaglyph format viewable through red-cyan
filters
right_left
Both views are arranged side by side, Right-eye view is on the
left
anaglyph_green_magenta
All frames are in anaglyph format viewable through green-
magenta filters
block_lr
Both eyes laced in one Block, Left-eye view is first
block_rl
Both eyes laced in one Block, Right-eye view is first
For example a 3D WebM clip can be created using the following command
line:
ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
Options
This muxer supports the following options:
reserve_index_space
By default, this muxer writes the index for seeking (called cues in
Matroska terms) at the end of the file, because it cannot know in
advance how much space to leave for the index at the beginning of
the file. However for some use cases -- e.g. streaming where
seeking is possible but slow -- it is useful to put the index at
the beginning of the file.
If this option is set to a non-zero value, the muxer will reserve a
given amount of space in the file header and then try to write the
cues there when the muxing finishes. If the available space does
not suffice, muxing will fail. A safe size for most use cases
should be about 50kB per hour of video.
Note that cues are only written if the output is seekable and this
option will have no effect if it is not.
md5
MD5 testing format.
This is a variant of the hash muxer. Unlike that muxer, it defaults to
using the MD5 hash function.
Examples
To compute the MD5 hash of the input converted to raw audio and video,
and store it in the file out.md5:
ffmpeg -i INPUT -f md5 out.md5
You can print the MD5 to stdout with the command:
ffmpeg -i INPUT -f md5 -
See also the hash and framemd5 muxers.
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).
-write_tmcd
Specify "on" to force writing a timecode track, "off" to disable it
and "auto" to write a timecode track only for mov and mp4 output
(default).
-movflags negative_cts_offsets
Enables utilization of version 1 of the CTTS box, in which the CTS
offsets can be negative. This enables the initial sample to have
DTS/CTS of zero, and reduces the need for edit lists for some cases
such as video tracks with B-frames. Additionally, eases conformance
with the DASH-IF interoperability guidelines.
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:
路 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.
路 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.
路 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_transport_stream_id integer
Set the transport_stream_id. This identifies a transponder in DVB.
Default is 0x0001.
mpegts_original_network_id integer
Set the original_network_id. 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. Default
is 0x0001.
mpegts_service_id integer
Set the service_id, also known as program in DVB. Default is
0x0001.
mpegts_service_type integer
Set the program service_type. Default is "digital_tv". Accepts the
following options:
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.
mpegts_pmt_start_pid integer
Set the first PID for PMT. Default is 0x1000. Max is 0x1f00.
mpegts_start_pid integer
Set the first PID for data packets. Default is 0x0100. Max is
0x0f00.
mpegts_m2ts_mode boolean
Enable m2ts mode if set to 1. Default value is "-1" which disables
m2ts mode.
muxrate integer
Set a constant muxrate. Default is VBR.
pes_payload_size integer
Set minimum PES packet payload in bytes. Default is 2930.
mpegts_flags flags
Set mpegts flags. Accepts the following options:
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.
system_b
Conform to System B (DVB) instead of System A (ATSC).
initial_discontinuity
Mark the initial packet of each stream as discontinuity.
resend_headers integer
Reemit PAT/PMT before writing the next packet. This option is
deprecated: use mpegts_flags instead.
mpegts_copyts boolean
Preserve original timestamps, if value is set to 1. Default value
is "-1", which results in shifting timestamps so that they start
from 0.
omit_video_pes_length boolean
Omit the PES packet length for video packets. Default is 1 (true).
pcr_period integer
Override the default PCR retransmission time in milliseconds.
Ignored if variable muxrate is selected. Default is 20.
pat_period double
Maximum time in seconds between PAT/PMT tables.
sdt_period double
Maximum time in seconds between SDT tables.
tables_version integer
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
...
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" \
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:
increment_tc 1|0
if set to 1, increment timecode between each segment If this is
selected, the input need to have a timecode in the first video
stream. Default value is 0.
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_clocktime_offset duration
Delay the segment splitting times with the specified duration when
using segment_atclocktime.
For example with segment_time set to "900" and
segment_clocktime_offset set to "300" this makes it possible to
create files at 12:05, 12:20, 12:35, etc.
Default value is "0".
segment_clocktime_wrap_duration duration
Force the segmenter to only start a new segment if a packet reaches
the muxer within the specified duration after the segmenting clock
time. This way you can make the segmenter more resilient to
backward local time jumps, such as leap seconds or transition to
standard time from daylight savings time.
Default is the maximum possible duration which means starting a new
segment regardless of the elapsed time since the last clock time.
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 beginning 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.
write_empty_segments 1|0
If enabled, write an empty segment if there are no packets during
the period a segment would usually span. Otherwise, the segment
will be filled with the next packet written. Defaults to 0.
Examples
路 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
路 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
路 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
路 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.
路 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
路 Convert the in.mkv to TS segments using the "libx264" and "aac"
encoders:
ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment -segment_list out.list out%03d.ts
路 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).
fifo
The fifo pseudo-muxer allows the separation of encoding and muxing by
using first-in-first-out queue and running the actual muxer in a
separate thread. This is especially useful in combination with the tee
muxer and can be used to send data to several destinations with
different reliability/writing speed/latency.
API users should be aware that callback functions (interrupt_callback,
io_open and io_close) used within its AVFormatContext must be thread-
safe.
The behavior of the fifo muxer if the queue fills up or if the output
fails is selectable,
路 output can be transparently restarted with configurable delay
between retries based on real time or time of the processed stream.
路 encoding can be blocked during temporary failure, or continue
transparently dropping packets in case fifo queue fills up.
fifo_format
Specify the format name. Useful if it cannot be guessed from the
output name suffix.
queue_size
Specify size of the queue (number of packets). Default value is 60.
format_opts
Specify format options for the underlying muxer. Muxer options can
be specified as a list of key=value pairs separated by ':'.
drop_pkts_on_overflow bool
If set to 1 (true), in case the fifo queue fills up, packets will
be dropped rather than blocking the encoder. This makes it possible
to continue streaming without delaying the input, at the cost of
omitting part of the stream. By default this option is set to 0
(false), so in such cases the encoder will be blocked until the
muxer processes some of the packets and none of them is lost.
attempt_recovery bool
If failure occurs, attempt to recover the output. This is
especially useful when used with network output, since it makes it
possible to restart streaming transparently. By default this
option is set to 0 (false).
max_recovery_attempts
Sets maximum number of successive unsuccessful recovery attempts
after which the output fails permanently. By default this option is
set to 0 (unlimited).
recovery_wait_time duration
Waiting time before the next recovery attempt after previous
unsuccessful recovery attempt. Default value is 5 seconds.
recovery_wait_streamtime bool
If set to 0 (false), the real time is used when waiting for the
recovery attempt (i.e. the recovery will be attempted after at
least recovery_wait_time seconds). If set to 1 (true), the time of
the processed stream is taken into account instead (i.e. the
recovery will be attempted after at least recovery_wait_time
seconds of the stream is omitted). By default, this option is set
to 0 (false).
recover_any_error bool
If set to 1 (true), recovery will be attempted regardless of type
of the error causing the failure. By default this option is set to
0 (false) and in case of certain (usually permanent) errors the
recovery is not attempted even when attempt_recovery is set to 1.
restart_with_keyframe bool
Specify whether to wait for the keyframe after recovering from
queue overflow or failure. This option is set to 0 (false) by
default.
Examples
路 Stream something to rtmp server, continue processing the stream at
real-time rate even in case of temporary failure (network outage)
and attempt to recover streaming every second indefinitely.
ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0:a
-drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 rtmp://example.com/live/stream_name
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.
use_fifo bool
If set to 1, slave outputs will be processed in separate thread
using fifo muxer. This allows to compensate for different
speed/latency/reliability of outputs and setup transparent
recovery. By default this feature is turned off.
fifo_options
Options to pass to fifo pseudo-muxer instances. See fifo.
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.
use_fifo bool
This allows to override tee muxer use_fifo option for individual
slave muxer.
fifo_options
This allows to override tee muxer fifo_options for individual slave
muxer. See fifo.
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. You may use multiple stream specifiers
separated by commas (",") e.g.: "a:0,v"
onfail
Specify behaviour on output failure. This can be set to either
"abort" (which is default) or "ignore". "abort" will cause whole
process to fail in case of failure on this slave output. "ignore"
will ignore failure on this output, so other outputs will continue
without being affected.
Examples
路 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/"
路 As above, but continue streaming even if output to local file fails
(for example local drive fills up):
ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
"[onfail=ignore]archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
路 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
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
路 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
-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:
路 WebM DASH Specification:
<https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>
路 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
PROTOCOL OPTIONS
The libavformat library provides some generic global options, which can
be set on all the protocols. In addition each protocol 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 "AVFormatContext" options or
using the libavutil/opt.h API for programmatic use.
The list of supported options follows:
protocol_whitelist list (input)
Set a ","-separated list of allowed protocols. "ALL" matches all
protocols. Protocols prefixed by "-" are disabled. All protocols
are allowed by default but protocols used by an another protocol
(nested protocols) are restricted to a per protocol subset.
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.
All protocols accept the following options:
rw_timeout
Maximum time to wait for (network) read/write operations to
complete, in microseconds.
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.
This protocol accepts the following options:
follow
If set to 1, the protocol will retry reading at the end of the
file, allowing reading files that still are being written. In order
for this to terminate, you either need to use the rw_timeout
option, or use the interrupt callback (for API users).
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 or for listen
mode.
http_proxy
set HTTP proxy to tunnel through e.g. http://example.com:1234
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
Override the User-Agent header. If not specified the protocol will
use a string describing the libavformat build. ("Lavf/<version>")
user-agent
This is a deprecated option, you can use user_agent instead it.
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.
reconnect_at_eof
If set then eof is treated like an error and causes reconnection,
this is useful for live / endless streams.
reconnect_streamed
If set then even streamed/non seekable streams will be reconnected
on errors.
reconnect_delay_max
Sets the maximum delay in seconds after which to give up
reconnecting
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 multi-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:
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 if data transmission is slow.
Note that some formats (typically MOV), require the output protocol to
be seekable, so they will fail with the pipe output protocol.
prompeg
Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction
mechanism for MPEG-2 Transport Streams sent over RTP.
This protocol must be used in conjunction with the "rtp_mpegts" muxer
and the "rtp" protocol.
The required syntax is:
-f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>
The destination UDP ports are "port + 2" for the column FEC stream and
"port + 4" for the row FEC stream.
This protocol accepts the following options:
l=n The number of columns (4-20, LxD <= 100)
d=n The number of rows (4-20, LxD <= 100)
Example usage:
-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>
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 milliseconds 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.
路 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
路 Watch a stream tunneled over HTTP:
ffplay -rtsp_transport http rtsp://server/video.mp4
路 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
路 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
tee
Writes the output to multiple protocols. The individual outputs are
separated by |
tee:file://path/to/local/this.avi|file://path/to/local/that.avi
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.
recv_buffer_size=bytes
Set receive buffer size, expressed bytes.
send_buffer_size=bytes
Set send buffer size, expressed bytes.
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.
bitrate=bitrate
If set to nonzero, the output will have the specified constant
bitrate if the input has enough packets to sustain it.
burst_bits=bits
When using bitrate this specifies the maximum number of bits in
packet bursts.
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
路 Use ffmpeg to stream over UDP to a remote endpoint:
ffmpeg -i <input> -f <format> udp://<hostname>:<port>
路 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
路 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 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 and 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
路 Print the list of AVFoundation supported devices and exit:
$ ffmpeg -f avfoundation -list_devices true -i ""
路 Record video from video device 0 and audio from audio device 0 into
out.avi:
$ ffmpeg -f avfoundation -i "0:0" out.avi
路 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
路 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 of
the input can be set with raw_format. 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. Note that
all audio channels are bundled in one single audio track.
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.
format_code <FourCC>
This sets the input video format to the format given by the FourCC.
To see the supported values of your device(s) use list_formats.
Note that there is a FourCC 'pal ' that can also be used as pal (3
letters).
bm_v210
This is a deprecated option, you can use raw_format instead. If
set to 1, video is captured in 10 bit v210 instead of uyvy422. Not
all Blackmagic devices support this option.
raw_format
Set the pixel format of the captured video. Available values are:
uyvy422
yuv422p10
argb
bgra
rgb10
teletext_lines
If set to nonzero, an additional teletext stream will be captured
from the vertical ancillary data. Both SD PAL (576i) and HD (1080i
or 1080p) sources are supported. In case of HD sources, OP47
packets are decoded.
This option is a bitmask of the SD PAL VBI lines captured,
specifically lines 6 to 22, and lines 318 to 335. Line 6 is the LSB
in the mask. Selected lines which do not contain teletext
information will be ignored. You can use the special all constant
to select all possible lines, or standard to skip lines 6, 318 and
319, which are not compatible with all receivers.
For SD sources, ffmpeg needs to be compiled with
"--enable-libzvbi". For HD sources, on older (pre-4K) DeckLink card
models you have to capture in 10 bit mode.
channels
Defines number of audio channels to capture. Must be 2, 8 or 16.
Defaults to 2.
duplex_mode
Sets the decklink device duplex mode. Must be unset, half or full.
Defaults to unset.
video_input
Sets the video input source. Must be unset, sdi, hdmi, optical_sdi,
component, composite or s_video. Defaults to unset.
audio_input
Sets the audio input source. Must be unset, embedded, aes_ebu,
analog, analog_xlr, analog_rca or microphone. Defaults to unset.
video_pts
Sets the video packet timestamp source. Must be video, audio,
reference or wallclock. Defaults to video.
audio_pts
Sets the audio packet timestamp source. Must be video, audio,
reference or wallclock. Defaults to audio.
draw_bars
If set to true, color bars are drawn in the event of a signal loss.
Defaults to true.
queue_size
Sets maximum input buffer size in bytes. If the buffering reaches
this value, incoming frames will be dropped. Defaults to
1073741824.
Examples
路 List input devices:
ffmpeg -f decklink -list_devices 1 -i dummy
路 List supported formats:
ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
路 Capture video clip at 1080i50:
ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi
路 Capture video clip at 1080i50 10 bit:
ffmpeg -bm_v210 1 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
路 Capture video clip at 1080i50 with 16 audio channels:
ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
kmsgrab
KMS video input device.
Captures the KMS scanout framebuffer associated with a specified CRTC
or plane as a DRM object that can be passed to other hardware
functions.
Requires either DRM master or CAP_SYS_ADMIN to run.
If you don't understand what all of that means, you probably don't want
this. Look at x11grab instead.
Options
device
DRM device to capture on. Defaults to /dev/dri/card0.
format
Pixel format of the framebuffer. Defaults to bgr0.
format_modifier
Format modifier to signal on output frames. This is necessary to
import correctly into some APIs, but can't be autodetected. See
the libdrm documentation for possible values.
crtc_id
KMS CRTC ID to define the capture source. The first active plane
on the given CRTC will be used.
plane_id
KMS plane ID to define the capture source. Defaults to the first
active plane found if neither crtc_id nor plane_id are specified.
framerate
Framerate to capture at. This is not synchronised to any page
flipping or framebuffer changes - it just defines the interval at
which the framebuffer is sampled. Sampling faster than the
framebuffer update rate will generate independent frames with the
same content. Defaults to 30.
Examples
路 Capture from the first active plane, download the result to normal
frames and encode. This will only work if the framebuffer is both
linear and mappable - if not, the result may be scrambled or fail
to download.
ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
路 Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert
to NV12 and encode as H.264.
ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
libndi_newtek
The libndi_newtek input device provides capture capabilities for using
NDI (Network Device Interface, standard created by NewTek).
Input filename is a NDI source name that could be found by sending
-find_sources 1 to command line - it has no specific syntax but human-
readable formatted.
To enable this input device, you need the NDI SDK and you need to
configure with the appropriate "--extra-cflags" and "--extra-ldflags".
Options
find_sources
If set to true, print a list of found/available NDI sources and
exit. Defaults to false.
wait_sources
Override time to wait until the number of online sources have
changed. Defaults to 0.5.
allow_video_fields
When this flag is false, all video that you receive will be
progressive. Defaults to true.
Examples
路 List input devices:
ffmpeg -f libndi_newtek -find_sources 1 -i dummy
路 Restream to NDI:
ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
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
路 Print the list of DirectShow supported devices and exit:
$ ffmpeg -list_devices true -f dshow -i dummy
路 Open video device Camera:
$ ffmpeg -f dshow -i video="Camera"
路 Open second video device with name Camera:
$ ffmpeg -f dshow -video_device_number 1 -i video="Camera"
路 Open video device Camera and audio device Microphone:
$ ffmpeg -f dshow -i video="Camera":audio="Microphone"
路 Print the list of supported options in selected device and exit:
$ ffmpeg -list_options true -f dshow -i video="Camera"
路 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"
路 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"
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
or
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 its 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
路 Grab and show the input of a FireWire DV/HDV device.
ffplay -f iec61883 -i auto
路 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
路 Create a color video stream and play it back with ffplay:
ffplay -f lavfi -graph "color=c=pink [out0]" dummy
路 As the previous example, but use filename for specifying the graph
description, and omit the "out0" label:
ffplay -f lavfi color=c=pink
路 Create three different video test filtered sources and play them:
ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3
路 Read an audio stream from a file using the amovie source and play
it back with ffplay:
ffplay -f lavfi "amovie=test.wav"
路 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]"
路 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
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:
路 List supported formats for a video4linux2 device:
ffplay -f video4linux2 -list_formats all /dev/video0
路 Grab and show the input of a video4linux2 device:
ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
路 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.
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 specifies
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".
grab_x
grab_y
Set the grabbing region coordinates. They are expressed as offset
from the top left corner of the X11 window and correspond to the
x_offset and y_offset parameters in the device name. The default
value for both options 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
路 Play a file on default ALSA device:
ffmpeg -i INPUT -f alsa default
路 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
路 The following command shows the ffmpeg output is an CACA window,
forcing its size to 80x25:
ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -
路 Show the list of available drivers and exit:
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -
路 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, field order 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
路 List output devices:
ffmpeg -i test.avi -f decklink -list_devices 1 dummy
路 List supported formats:
ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'
路 Play video clip:
ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'
路 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'
libndi_newtek
The libndi_newtek output device provides playback capabilities for
using NDI (Network Device Interface, standard created by NewTek).
Output filename is a NDI name.
To enable this output device, you need the NDI SDK and you need to
configure with the appropriate "--extra-cflags" and "--extra-ldflags".
NDI uses uyvy422 pixel format natively, but also supports bgra, bgr0,
rgba and rgb0.
Options
reference_level
The audio reference level in dB. This specifies how many dB above
the reference level (+4dBU) is the full range of 16 bit audio.
Defaults to 0.
clock_video
These specify whether video "clock" themselves. Defaults to false.
clock_audio
These specify whether audio "clock" themselves. Defaults to false.
Examples
路 Play video clip:
ffmpeg -i "udp://@239.1.1.1:10480?fifo_size=1000000&overrun_nonfatal=1" -vf "scale=720:576,fps=fps=25,setdar=dar=16/9,format=pix_fmts=uyvy422" -f libndi_newtek NEW_NDI1
fbdev
Linux framebuffer output 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.
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 -c:v 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 -c:v 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
路 Decode, display and encode video input with ffmpeg at the same
time:
ffmpeg -i INPUT OUTPUT -f xv display
路 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 volume reduction. A value of 1.0
prevents clipping.
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, exact_rational,
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 when enabled (the default). Disable it if
you want to preserve speed instead of quality when exact_rational
fails.
exact_rational
For swr only, when enabled, try to use exact phase_count based on
input and output sample rate. However, if it is larger than "1 <<
phase_shift", the phase_count will be "1 << phase_shift" as
fallback. Default is enabled.
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 a double float
value 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. Default
value is bicubic.
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@id=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 optionally followed by "@id". 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:
路 A ':'-separated list of key=value pairs.
路 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.
路 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 '_'
<FILTER_NAME> ::= <NAME>["@"<NAME>]
<LINKLABEL> ::= "[" <NAME> "]"
<LINKLABELS> ::= <LINKLABEL> [<LINKLABELS>]
<FILTER_ARGUMENTS> ::= sequence of chars (possibly quoted)
<FILTER> ::= [<LINKLABELS>] <FILTER_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
See "ffmpeg -filters" to view which filters have timeline support.
OPTIONS FOR FILTERS WITH SEVERAL INPUTS
Some filters with several inputs support a common set of options.
These options can only be set by name, not with the short notation.
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.
shortest
If set to 1, force the output to terminate when the shortest input
terminates. Default value is 0.
repeatlast
If set to 1, force the filter to extend the last frame of secondary
streams until the end of the primary stream. A value of 0 disables
this behavior. Default value is 1.
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.
acompressor
A compressor is mainly used to reduce the dynamic range of a signal.
Especially modern music is mostly compressed at a high ratio to improve
the overall loudness. It's done to get the highest attention of a
listener, "fatten" the sound and bring more "power" to the track. If a
signal is compressed too much it may sound dull or "dead" afterwards or
it may start to "pump" (which could be a powerful effect but can also
destroy a track completely). The right compression is the key to reach
a professional sound and is the high art of mixing and mastering.
Because of its complex settings it may take a long time to get the
right feeling for this kind of effect.
Compression is done by detecting the volume above a chosen level
"threshold" and dividing it by the factor set with "ratio". So if you
set the threshold to -12dB and your signal reaches -6dB a ratio of 2:1
will result in a signal at -9dB. Because an exact manipulation of the
signal would cause distortion of the waveform the reduction can be
levelled over the time. This is done by setting "Attack" and "Release".
"attack" determines how long the signal has to rise above the threshold
before any reduction will occur and "release" sets the time the signal
has to fall below the threshold to reduce the reduction again. Shorter
signals than the chosen attack time will be left untouched. The
overall reduction of the signal can be made up afterwards with the
"makeup" setting. So compressing the peaks of a signal about 6dB and
raising the makeup to this level results in a signal twice as loud than
the source. To gain a softer entry in the compression the "knee"
flattens the hard edge at the threshold in the range of the chosen
decibels.
The filter accepts the following options:
level_in
Set input gain. Default is 1. Range is between 0.015625 and 64.
threshold
If a signal of stream rises above this level it will affect the
gain reduction. By default it is 0.125. Range is between
0.00097563 and 1.
ratio
Set a ratio by which the signal is reduced. 1:2 means that if the
level rose 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 1. Range is from 1 to 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 input stream
or the louder("maximum") channel of input 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 mostly smoother.
mix How much to use compressed signal in output. Default is 1. Range
is between 0 and 1.
acopy
Copy the input audio source unchanged to the output. This is mainly
useful for testing purposes.
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
路 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
路 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
acrusher
Reduce audio bit resolution.
This filter is bit crusher with enhanced functionality. A bit crusher
is used to audibly reduce number of bits an audio signal is sampled
with. This doesn't change the bit depth at all, it just produces the
effect. Material reduced in bit depth sounds more harsh and "digital".
This filter is able to even round to continuous values instead of
discrete bit depths. Additionally it has a D/C offset which results in
different crushing of the lower and the upper half of the signal. An
Anti-Aliasing setting is able to produce "softer" crushing sounds.
Another feature of this filter is the logarithmic mode. This setting
switches from linear distances between bits to logarithmic ones. The
result is a much more "natural" sounding crusher which doesn't gate low
signals for example. The human ear has a logarithmic perception, too so
this kind of crushing is much more pleasant. Logarithmic crushing is
also able to get anti-aliased.
The filter accepts the following options:
level_in
Set level in.
level_out
Set level out.
bits
Set bit reduction.
mix Set mixing amount.
mode
Can be linear: "lin" or logarithmic: "log".
dc Set DC.
aa Set anti-aliasing.
samples
Set sample reduction.
lfo Enable LFO. By default disabled.
lforange
Set LFO range.
lforate
Set LFO rate.
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
'|'. Unused delays will be silently ignored. If number of given
delays is smaller than number of channels all remaining channels
will not be delayed. If you want to delay exact number of samples,
append 'S' to number.
Examples
路 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
路 Delay second channel by 500 samples, the third channel by 700
samples and leave the first channel (and any other channels that
may be present) unchanged.
adelay=0|500S|700S
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 loudness of reflected signals separated by '|'.
Allowed range for each "decay" is "(0 - 1.0]". Default is 0.5.
Examples
路 Make it sound as if there are twice as many instruments as are
actually playing:
aecho=0.8:0.88:60:0.4
路 If delay is very short, then it sound like a (metallic) robot
playing music:
aecho=0.8:0.88:6:0.4
路 A longer delay will sound like an open air concert in the
mountains:
aecho=0.8:0.9:1000:0.3
路 Same as above but with one more mountain:
aecho=0.8:0.9:1000|1800:0.3|0.25
aemphasis
Audio emphasis filter creates or restores material directly taken from
LPs or emphased CDs with different filter curves. E.g. to store music
on vinyl the signal has to be altered by a filter first to even out the
disadvantages of this recording medium. Once the material is played
back the inverse filter has to be applied to restore the distortion of
the frequency response.
The filter accepts the following options:
level_in
Set input gain.
level_out
Set output gain.
mode
Set filter mode. For restoring material use "reproduction" mode,
otherwise use "production" mode. Default is "reproduction" mode.
type
Set filter type. Selects medium. Can be one of the following:
col select Columbia.
emi select EMI.
bsi select BSI (78RPM).
riaa
select RIAA.
cd select Compact Disc (CD).
50fm
select 50Xs (FM).
75fm
select 75Xs (FM).
50kf
select 50Xs (FM-KF).
75kf
select 75Xs (FM-KF).
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
路 Half volume:
aeval=val(ch)/2:c=same
路 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
路 Fade in first 15 seconds of audio:
afade=t=in:ss=0:d=15
路 Fade out last 25 seconds of a 900 seconds audio:
afade=t=out:st=875:d=25
afftfilt
Apply arbitrary expressions to samples in frequency domain.
real
Set frequency domain real expression for each separate channel
separated by '|'. Default is "1". If the number of input channels
is greater than the number of expressions, the last specified
expression is used for the remaining output channels.
imag
Set frequency domain imaginary expression for each separate channel
separated by '|'. If not set, real option is used.
Each expression in real and imag can contain the following
constants:
sr sample rate
b current frequency bin number
nb number of available bins
ch channel number of the current expression
chs number of channels
pts current frame pts
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 "w4096"
win_func
Set window function. Default is "hann".
overlap
Set window overlap. If set to 1, the recommended overlap for
selected window function will be picked. Default is 0.75.
Examples
路 Leave almost only low frequencies in audio:
afftfilt="1-clip((b/nb)*b,0,1)"
afir
Apply an arbitrary Frequency Impulse Response filter.
This filter is designed for applying long FIR filters, up to 30 seconds
long.
It can be used as component for digital crossover filters, room
equalization, cross talk cancellation, wavefield synthesis,
auralization, ambiophonics and ambisonics.
This filter uses second stream as FIR coefficients. If second stream
holds single channel, it will be used for all input channels in first
stream, otherwise number of channels in second stream must be same as
number of channels in first stream.
It accepts the following parameters:
dry Set dry gain. This sets input gain.
wet Set wet gain. This sets final output gain.
length
Set Impulse Response filter length. Default is 1, which means whole
IR is processed.
again
Enable applying gain measured from power of IR.
Examples
路 Apply reverb to stream using mono IR file as second input, complete
command using ffmpeg:
ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav
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
agate
A gate is mainly used to reduce lower parts of a signal. This kind of
signal processing reduces disturbing noise between useful signals.
Gating is done by detecting the volume below a chosen level threshold
and dividing it by the factor set with ratio. The bottom of the noise
floor is set via range. Because an exact manipulation of the signal
would cause distortion of the waveform the reduction can be levelled
over time. This is done by setting attack and release.
attack determines how long the signal has to fall below the threshold
before any reduction will occur and release sets the time the signal
has to rise above the threshold to reduce the reduction again. Shorter
signals than the chosen attack time will be left untouched.
level_in
Set input level before filtering. Default is 1. Allowed range is
from 0.015625 to 64.
range
Set the level of gain reduction when the signal is below the
threshold. Default is 0.06125. Allowed range is from 0 to 1.
threshold
If a signal rises above this level the gain reduction is released.
Default is 0.125. Allowed range is from 0 to 1.
ratio
Set a ratio by which the signal is reduced. Default is 2. Allowed
range is from 1 to 9000.
attack
Amount of milliseconds the signal has to rise above the threshold
before gain reduction stops. Default is 20 milliseconds. Allowed
range is from 0.01 to 9000.
release
Amount of milliseconds the signal has to fall below the threshold
before the reduction is increased again. Default is 250
milliseconds. Allowed range is from 0.01 to 9000.
makeup
Set amount of amplification of signal after processing. Default is
1. Allowed range is from 1 to 64.
knee
Curve the sharp knee around the threshold to enter gain reduction
more softly. Default is 2.828427125. Allowed range is from 1 to 8.
detection
Choose if exact signal should be taken for detection or an RMS like
one. Default is "rms". Can be "peak" or "rms".
link
Choose if the average level between all channels or the louder
channel affects the reduction. Default is "average". Can be
"average" or "maximum".
alimiter
The limiter prevents an input signal from rising over a desired
threshold. This limiter uses lookahead technology to prevent your
signal from distorting. It means that there is a small delay after the
signal is processed. Keep in mind that the delay it produces is the
attack time you set.
The filter accepts the following options:
level_in
Set input gain. Default is 1.
level_out
Set output gain. Default is 1.
limit
Don't let signals above this level pass the limiter. Default is 1.
attack
The limiter will reach its attenuation level in this amount of time
in milliseconds. Default is 5 milliseconds.
release
Come back from limiting to attenuation 1.0 in this amount of
milliseconds. Default is 50 milliseconds.
asc When gain reduction is always needed ASC takes care of releasing to
an average reduction level rather than reaching a reduction of 0 in
the release time.
asc_level
Select how much the release time is affected by ASC, 0 means nearly
no changes in release time while 1 produces higher release times.
level
Auto level output signal. Default is enabled. This normalizes
audio back to 0dB if enabled.
Depending on picked setting it is recommended to upsample input 2x or
4x times with aresample before applying this filter.
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, t
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.
channels, c
Specify which channels to filter, by default all available are
filtered.
aloop
Loop audio samples.
The filter accepts the following options:
loop
Set the number of loops.
size
Set maximal number of samples.
start
Set first sample of loop.
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
路 Merge two mono files into a stereo stream:
amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
路 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.
anequalizer
High-order parametric multiband equalizer for each channel.
It accepts the following parameters:
params
This option string is in format: "cchn f=cf w=w g=g t=f | ..."
Each equalizer band is separated by '|'.
chn Set channel number to which equalization will be applied. If
input doesn't have that channel the entry is ignored.
f Set central frequency for band. If input doesn't have that
frequency the entry is ignored.
w Set band width in hertz.
g Set band gain in dB.
t Set filter type for band, optional, can be:
0 Butterworth, this is default.
1 Chebyshev type 1.
2 Chebyshev type 2.
curves
With this option activated frequency response of anequalizer is
displayed in video stream.
size
Set video stream size. Only useful if curves option is activated.
mgain
Set max gain that will be displayed. Only useful if curves option
is activated. Setting this to a reasonable value makes it possible
to display gain which is derived from neighbour bands which are too
close to each other and thus produce higher gain when both are
activated.
fscale
Set frequency scale used to draw frequency response in video
output. Can be linear or logarithmic. Default is logarithmic.
colors
Set color for each channel curve which is going to be displayed in
video stream. This is list of color names separated by space or by
'|'. Unrecognised or missing colors will be replaced by white
color.
Examples
路 Lower gain by 10 of central frequency 200Hz and width 100 Hz for
first 2 channels using Chebyshev type 1 filter:
anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1
Commands
This filter supports the following commands:
change
Alter existing filter parameters. Syntax for the commands is :
"fN|f=freq|w=width|g=gain"
fN is existing filter number, starting from 0, if no such filter is
available error is returned. freq set new frequency parameter.
width set new width parameter in herz. gain set new gain parameter
in dB.
Full filter invocation with asendcmd may look like this:
asendcmd=c='4.0 anequalizer change
0|f=200|w=50|g=1',anequalizer=...
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
路 Add 1024 samples of silence to the end of the input:
apad=pad_len=1024
路 Make sure the audio output will contain at least 10000 samples, pad
the input with silence if required:
apad=whole_len=10000
路 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
apulsator
Audio pulsator is something between an autopanner and a tremolo. But
it can produce funny stereo effects as well. Pulsator changes the
volume of the left and right channel based on a LFO (low frequency
oscillator) with different waveforms and shifted phases. This filter
have the ability to define an offset between left and right channel. An
offset of 0 means that both LFO shapes match each other. The left and
right channel are altered equally - a conventional tremolo. An offset
of 50% means that the shape of the right channel is exactly shifted in
phase (or moved backwards about half of the frequency) - pulsator acts
as an autopanner. At 1 both curves match again. Every setting in
between moves the phase shift gapless between all stages and produces
some "bypassing" sounds with sine and triangle waveforms. The more you
set the offset near 1 (starting from the 0.5) the faster the signal
passes from the left to the right speaker.
The filter accepts the following options:
level_in
Set input gain. By default it is 1. Range is [0.015625 - 64].
level_out
Set output gain. By default it is 1. Range is [0.015625 - 64].
mode
Set waveform shape the LFO will use. Can be one of: sine, triangle,
square, sawup or sawdown. Default is sine.
amount
Set modulation. Define how much of original signal is affected by
the LFO.
offset_l
Set left channel offset. Default is 0. Allowed range is [0 - 1].
offset_r
Set right channel offset. Default is 0.5. Allowed range is [0 - 1].
width
Set pulse width. Default is 1. Allowed range is [0 - 2].
timing
Set possible timing mode. Can be one of: bpm, ms or hz. Default is
hz.
bpm Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if
timing is set to bpm.
ms Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if
timing is set to ms.
hz Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100].
Only used if timing is set to hz.
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 the "Resampler Options"
section in the ffmpeg-resampler(1) manual for the complete list of
supported options.
Examples
路 Resample the input audio to 44100Hz:
aresample=44100
路 Stretch/squeeze samples to the given timestamps, with a maximum of
1000 samples per second compensation:
aresample=async=1000
areverse
Reverse an audio clip.
Warning: This filter requires memory to buffer the entire clip, so
trimming is suggested.
Examples
路 Take the first 5 seconds of a clip, and reverse it.
atrim=end=5,areverse
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
signals 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 RMS_difference
Peak_level RMS_peak RMS_trough Crest_factor Flat_factor Peak_count
Bit_depth Dynamic_range
and for Overall: DC_offset Min_level Max_level Min_difference
Max_difference Mean_difference RMS_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.
RMS difference
Root Mean Square 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.
Dynamic range
Measured dynamic range of audio in dB.
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
路 Slow down audio to 80% tempo:
atempo=0.8
路 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:
路 Drop everything except the second minute of input:
ffmpeg -i INPUT -af atrim=60:120
路 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, t
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.
channels, c
Specify which channels to filter, by default all available are
filtered.
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, t
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.
channels, c
Specify which channels to filter, by default all available are
filtered.
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, t
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.
channels, c
Specify which channels to filter, by default all available are
filtered.
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. and channels, c specify which channels to filter, by
default all available are filtered.
bs2b
Bauer stereo to binaural transformation, which improves headphone
listening of stereo audio records.
To enable compilation of this filter you need to configure FFmpeg with
"--enable-libbs2b".
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:
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.
channel_layout
The channel layout of the output stream.
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
路 A single delay:
chorus=0.7:0.9:55:0.4:0.25:2
路 Two delays:
chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
路 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|1/0".
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
路 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
路 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
路 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
路 2:1 compression starting at -6dB:
compand=points=-80/-80|-6/-6|0/-3.8|20/3.5
路 2:1 compression starting at -9dB:
compand=points=-80/-80|-9/-9|0/-5.3|20/2.9
路 2:1 compression starting at -12dB:
compand=points=-80/-80|-12/-12|0/-6.8|20/1.9
路 2:1 compression starting at -18dB:
compand=points=-80/-80|-18/-18|0/-9.8|20/0.7
路 3:1 compression starting at -15dB:
compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2
路 Compressor/Gate:
compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6
路 Expander:
compand=attacks=0:points=-80/-169|-54/-80|-49.5/-64.6|-41.1/-41.1|-25.8/-15|-10.8/-4.5|0/0|20/8.3
路 Hard limiter at -6dB:
compand=attacks=0:points=-80/-80|-6/-6|20/-6
路 Hard limiter at -12dB:
compand=attacks=0:points=-80/-80|-12/-12|20/-12
路 Hard noise gate at -35 dB:
compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20
路 Soft limiter:
compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8
compensationdelay
Compensation Delay Line is a metric based delay to compensate differing
positions of microphones or speakers.
For example, you have recorded guitar with two microphones placed in
different location. Because the front of sound wave has fixed speed in
normal conditions, the phasing of microphones can vary and depends on
their location and interposition. The best sound mix can be achieved
when these microphones are in phase (synchronized). Note that distance
of ~30 cm between microphones makes one microphone to capture signal in
antiphase to another microphone. That makes the final mix sounding
moody. This filter helps to solve phasing problems by adding different
delays to each microphone track and make them synchronized.
The best result can be reached when you take one track as base and
synchronize other tracks one by one with it. Remember that
synchronization/delay tolerance depends on sample rate, too. Higher
sample rates will give more tolerance.
It accepts the following parameters:
mm Set millimeters distance. This is compensation distance for fine
tuning. Default is 0.
cm Set cm distance. This is compensation distance for tightening
distance setup. Default is 0.
m Set meters distance. This is compensation distance for hard
distance setup. Default is 0.
dry Set dry amount. Amount of unprocessed (dry) signal. Default is 0.
wet Set wet amount. Amount of processed (wet) signal. Default is 1.
temp
Set temperature degree in Celsius. This is the temperature of the
environment. Default is 20.
crossfeed
Apply headphone crossfeed filter.
Crossfeed is the process of blending the left and right channels of
stereo audio recording. It is mainly used to reduce extreme stereo
separation of low frequencies.
The intent is to produce more speaker like sound to the listener.
The filter accepts the following options:
strength
Set strength of crossfeed. Default is 0.2. Allowed range is from 0
to 1. This sets gain of low shelf filter for side part of stereo
image. Default is -6dB. Max allowed is -30db when strength is set
to 1.
range
Set soundstage wideness. Default is 0.5. Allowed range is from 0 to
1. This sets cut off frequency of low shelf filter. Default is cut
off near 1550 Hz. With range set to 1 cut off frequency is set to
2100 Hz.
level_in
Set input gain. Default is 0.9.
level_out
Set output gain. Default is 1.
crystalizer
Simple algorithm to expand audio dynamic range.
The filter accepts the following options:
i Sets the intensity of effect (default: 2.0). Must be in range
between 0.0 (unchanged sound) to 10.0 (maximum effect).
c Enable clipping. By default is enabled.
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, t
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.
channels, c
Specify which channels to filter, by default all available are
filtered.
Examples
路 Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:
equalizer=f=1000:t=h:width=200:g=-10
路 Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz
with Q 2:
equalizer=f=1000:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5
extrastereo
Linearly increases the difference between left and right channels which
adds some sort of "live" effect to playback.
The filter accepts the following options:
m Sets the difference coefficient (default: 2.5). 0.0 means mono
sound (average of both channels), with 1.0 sound will be unchanged,
with -1.0 left and right channels will be swapped.
c Enable clipping. By default is enabled.
firequalizer
Apply FIR Equalization using arbitrary frequency response.
The filter accepts the following option:
gain
Set gain curve equation (in dB). The expression can contain
variables:
f the evaluated frequency
sr sample rate
ch channel number, set to 0 when multichannels evaluation is
disabled
chid
channel id, see libavutil/channel_layout.h, set to the first
channel id when multichannels evaluation is disabled
chs number of channels
chlayout
channel_layout, see libavutil/channel_layout.h
and functions:
gain_interpolate(f)
interpolate gain on frequency f based on gain_entry
cubic_interpolate(f)
same as gain_interpolate, but smoother
This option is also available as command. Default is
gain_interpolate(f).
gain_entry
Set gain entry for gain_interpolate function. The expression can
contain functions:
entry(f, g)
store gain entry at frequency f with value g
This option is also available as command.
delay
Set filter delay in seconds. Higher value means more accurate.
Default is 0.01.
accuracy
Set filter accuracy in Hz. Lower value means more accurate.
Default is 5.
wfunc
Set window function. Acceptable values are:
rectangular
rectangular window, useful when gain curve is already smooth
hann
hann window (default)
hamming
hamming window
blackman
blackman window
nuttall3
3-terms continuous 1st derivative nuttall window
mnuttall3
minimum 3-terms discontinuous nuttall window
nuttall
4-terms continuous 1st derivative nuttall window
bnuttall
minimum 4-terms discontinuous nuttall (blackman-nuttall) window
bharris
blackman-harris window
tukey
tukey window
fixed
If enabled, use fixed number of audio samples. This improves speed
when filtering with large delay. Default is disabled.
multi
Enable multichannels evaluation on gain. Default is disabled.
zero_phase
Enable zero phase mode by subtracting timestamp to compensate
delay. Default is disabled.
scale
Set scale used by gain. Acceptable values are:
linlin
linear frequency, linear gain
linlog
linear frequency, logarithmic (in dB) gain (default)
loglin
logarithmic (in octave scale where 20 Hz is 0) frequency,
linear gain
loglog
logarithmic frequency, logarithmic gain
dumpfile
Set file for dumping, suitable for gnuplot.
dumpscale
Set scale for dumpfile. Acceptable values are same with scale
option. Default is linlog.
fft2
Enable 2-channel convolution using complex FFT. This improves speed
significantly. Default is disabled.
min_phase
Enable minimum phase impulse response. Default is disabled.
Examples
路 lowpass at 1000 Hz:
firequalizer=gain='if(lt(f,1000), 0, -INF)'
路 lowpass at 1000 Hz with gain_entry:
firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'
路 custom equalization:
firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'
路 higher delay with zero phase to compensate delay:
firequalizer=delay=0.1:fixed=on:zero_phase=on
路 lowpass on left channel, highpass on right channel:
firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
:gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on
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 sweep 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.
haas
Apply Haas effect to audio.
Note that this makes most sense to apply on mono signals. With this
filter applied to mono signals it give some directionality and
stretches its stereo image.
The filter accepts the following options:
level_in
Set input level. By default is 1, or 0dB
level_out
Set output level. By default is 1, or 0dB.
side_gain
Set gain applied to side part of signal. By default is 1.
middle_source
Set kind of middle source. Can be one of the following:
left
Pick left channel.
right
Pick right channel.
mid Pick middle part signal of stereo image.
side
Pick side part signal of stereo image.
middle_phase
Change middle phase. By default is disabled.
left_delay
Set left channel delay. By default is 2.05 milliseconds.
left_balance
Set left channel balance. By default is -1.
left_gain
Set left channel gain. By default is 1.
left_phase
Change left phase. By default is disabled.
right_delay
Set right channel delay. By defaults is 2.12 milliseconds.
right_balance
Set right channel balance. By default is 1.
right_gain
Set right channel gain. By default is 1.
right_phase
Change right phase. By default is enabled.
hdcd
Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM
stream with embedded HDCD codes is expanded into a 20-bit PCM stream.
The filter supports the Peak Extend and Low-level Gain Adjustment
features of HDCD, and detects the Transient Filter flag.
ffmpeg -i HDCD16.flac -af hdcd OUT24.flac
When using the filter with wav, note the default encoding for wav is
16-bit, so the resulting 20-bit stream will be truncated back to
16-bit. Use something like -acodec pcm_s24le after the filter to get
24-bit PCM output.
ffmpeg -i HDCD16.wav -af hdcd OUT16.wav
ffmpeg -i HDCD16.wav -af hdcd -c:a pcm_s24le OUT24.wav
The filter accepts the following options:
disable_autoconvert
Disable any automatic format conversion or resampling in the filter
graph.
process_stereo
Process the stereo channels together. If target_gain does not match
between channels, consider it invalid and use the last valid
target_gain.
cdt_ms
Set the code detect timer period in ms.
force_pe
Always extend peaks above -3dBFS even if PE isn't signaled.
analyze_mode
Replace audio with a solid tone and adjust the amplitude to signal
some specific aspect of the decoding process. The output file can
be loaded in an audio editor alongside the original to aid
analysis.
"analyze_mode=pe:force_pe=true" can be used to see all samples
above the PE level.
Modes are:
0, off
Disabled
1, lle
Gain adjustment level at each sample
2, pe
Samples where peak extend occurs
3, cdt
Samples where the code detect timer is active
4, tgm
Samples where the target gain does not match between channels
headphone
Apply head-related transfer functions (HRTFs) to create virtual
loudspeakers around the user for binaural listening via headphones.
The HRIRs are provided via additional streams, for each channel one
stereo input stream is needed.
The filter accepts the following options:
map Set mapping of input streams for convolution. The argument is a
'|'-separated list of channel names in order as they are given as
additional stream inputs for filter. This also specify number of
input streams. Number of input streams must be not less than number
of channels in first stream plus one.
gain
Set gain applied to audio. Value is in dB. Default is 0.
type
Set processing type. Can be time or freq. time is processing audio
in time domain which is slow. freq is processing audio in
frequency domain which is fast. Default is freq.
lfe Set custom gain for LFE channels. Value is in dB. Default is 0.
Examples
路 Full example using wav files as coefficients with amovie filters
for 7.1 downmix, each amovie filter use stereo file with IR
coefficients as input. The files give coefficients for each
position of virtual loudspeaker:
ffmpeg -i input.wav -lavfi-complex "amovie=azi_270_ele_0_DFC.wav[sr],amovie=azi_90_ele_0_DFC.wav[sl],amovie=azi_225_ele_0_DFC.wav[br],amovie=azi_135_ele_0_DFC.wav[bl],amovie=azi_0_ele_0_DFC.wav,asplit[fc][lfe],amovie=azi_35_ele_0_DFC.wav[fl],amovie=azi_325_ele_0_DFC.wav[fr],[a:0][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"
output.wav
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, t
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.
channels, c
Specify which channels to filter, by default all available are
filtered.
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. Alternatively they can be also defined using the
following syntax: value0|value1|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
路 List all available plugins within amp (LADSPA example plugin)
library:
ladspa=file=amp
路 List all available controls and their valid ranges for "vcf_notch"
plugin from "VCF" library:
ladspa=f=vcf:p=vcf_notch:c=help
路 Simulate low quality audio equipment using "Computer Music Toolkit"
(CMT) plugin library:
ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12
路 Add reverberation to the audio using TAP-plugins (Tom's Audio
Processing plugins):
ladspa=file=tap_reverb:tap_reverb
路 Generate white noise, with 0.2 amplitude:
ladspa=file=cmt:noise_source_white:c=c0=.2
路 Generate 20 bpm clicks using plugin "C* Click - Metronome" from the
"C* Audio Plugin Suite" (CAPS) library:
ladspa=file=caps:Click:c=c1=20'
路 Apply "C* Eq10X2 - Stereo 10-band equaliser" effect:
ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2
路 Increase volume by 20dB using fast lookahead limiter from Steve
Harris "SWH Plugins" collection:
ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2
路 Attenuate low frequencies using Multiband EQ from Steve Harris "SWH
Plugins" collection:
ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0
路 Reduce stereo image using "Narrower" from the "C* Audio Plugin
Suite" (CAPS) library:
ladspa=caps:Narrower
路 Another white noise, now using "C* Audio Plugin Suite" (CAPS)
library:
ladspa=caps:White:.2
路 Some fractal noise, using "C* Audio Plugin Suite" (CAPS) library:
ladspa=caps:Fractal:c=c1=1
路 Dynamic volume normalization using "VLevel" plugin:
ladspa=vlevel-ladspa:vlevel_mono
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.
loudnorm
EBU R128 loudness normalization. Includes both dynamic and linear
normalization modes. Support for both single pass (livestreams, files)
and double pass (files) modes. This algorithm can target IL, LRA, and
maximum true peak. To accurately detect true peaks, the audio stream
will be upsampled to 192 kHz unless the normalization mode is linear.
Use the "-ar" option or "aresample" filter to explicitly set an output
sample rate.
The filter accepts the following options:
I, i
Set integrated loudness target. Range is -70.0 - -5.0. Default
value is -24.0.
LRA, lra
Set loudness range target. Range is 1.0 - 20.0. Default value is
7.0.
TP, tp
Set maximum true peak. Range is -9.0 - +0.0. Default value is
-2.0.
measured_I, measured_i
Measured IL of input file. Range is -99.0 - +0.0.
measured_LRA, measured_lra
Measured LRA of input file. Range is 0.0 - 99.0.
measured_TP, measured_tp
Measured true peak of input file. Range is -99.0 - +99.0.
measured_thresh
Measured threshold of input file. Range is -99.0 - +0.0.
offset
Set offset gain. Gain is applied before the true-peak limiter.
Range is -99.0 - +99.0. Default is +0.0.
linear
Normalize linearly if possible. measured_I, measured_LRA,
measured_TP, and measured_thresh must also to be specified in order
to use this mode. Options are true or false. Default is true.
dual_mono
Treat mono input files as "dual-mono". If a mono file is intended
for playback on a stereo system, its EBU R128 measurement will be
perceptually incorrect. If set to "true", this option will
compensate for this effect. Multi-channel input files are not
affected by this option. Options are true or false. Default is
false.
print_format
Set print format for stats. Options are summary, json, or none.
Default value is none.
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, t
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.
channels, c
Specify which channels to filter, by default all available are
filtered.
Examples
路 Lowpass only LFE channel, it LFE is not present it does nothing:
lowpass=c=LFE
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.
rubberband
Apply time-stretching and pitch-shifting with librubberband.
The filter accepts the following options:
tempo
Set tempo scale factor.
pitch
Set pitch scale factor.
transients
Set transients detector. Possible values are:
crisp
mixed
smooth
detector
Set detector. Possible values are:
compound
percussive
soft
phase
Set phase. Possible values are:
laminar
independent
window
Set processing window size. Possible values are:
standard
short
long
smoothing
Set smoothing. Possible values are:
off
on
formant
Enable formant preservation when shift pitching. Possible values
are:
shifted
preserved
pitchq
Set pitch quality. Possible values are:
quality
speed
consistency
channels
Set channels. Possible values are:
apart
together
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:
level_in
Set input gain. Default is 1. Range is between 0.015625 and 64.
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 1. Range is from 1 to 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.
level_sc
Set sidechain gain. Default is 1. Range is between 0.015625 and 64.
mix How much to use compressed signal in output. Default is 1. Range
is between 0 and 1.
Examples
路 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"
sidechaingate
A sidechain gate acts like a normal (wideband) gate but has the ability
to filter the detected signal before sending it to the gain reduction
stage. Normally a gate uses the full range signal to detect a level
above the threshold. For example: If you cut all lower frequencies
from your sidechain signal the gate will decrease the volume of your
track only if not enough highs appear. With this technique you are able
to reduce the resonation of a natural drum or remove "rumbling" of
muted strokes from a heavily distorted guitar. It needs two input
streams and returns one output stream. First input stream will be
processed depending on second stream signal.
The filter accepts the following options:
level_in
Set input level before filtering. Default is 1. Allowed range is
from 0.015625 to 64.
range
Set the level of gain reduction when the signal is below the
threshold. Default is 0.06125. Allowed range is from 0 to 1.
threshold
If a signal rises above this level the gain reduction is released.
Default is 0.125. Allowed range is from 0 to 1.
ratio
Set a ratio about which the signal is reduced. Default is 2.
Allowed range is from 1 to 9000.
attack
Amount of milliseconds the signal has to rise above the threshold
before gain reduction stops. Default is 20 milliseconds. Allowed
range is from 0.01 to 9000.
release
Amount of milliseconds the signal has to fall below the threshold
before the reduction is increased again. Default is 250
milliseconds. Allowed range is from 0.01 to 9000.
makeup
Set amount of amplification of signal after processing. Default is
1. Allowed range is from 1 to 64.
knee
Curve the sharp knee around the threshold to enter gain reduction
more softly. Default is 2.828427125. Allowed range is from 1 to 8.
detection
Choose if exact signal should be taken for detection or an RMS like
one. Default is rms. Can be peak or rms.
link
Choose if the average level between all channels or the louder
channel affects the reduction. Default is average. Can be average
or maximum.
level_sc
Set sidechain gain. Default is 1. Range is from 0.015625 to 64.
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
路 Detect 5 seconds of silence with -50dB noise tolerance:
silencedetect=n=-50dB:d=5
路 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 indicates 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.
detection
Set how is silence detected. Can be "rms" or "peak". Second is
faster and works better with digital silence which is exactly 0.
Default value is "rms".
window
Set ratio used to calculate size of window for detecting silence.
Default value is 0.02. Allowed range is from 0 to 10.
Examples
路 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
路 Trim all silence encountered from beginning to end where there is
more than 1 second of silence in audio:
silenceremove=0:0:0:-1:1:-90dB
sofalizer
SOFAlizer uses head-related transfer functions (HRTFs) to create
virtual loudspeakers around the user for binaural listening via
headphones (audio formats up to 9 channels supported). The HRTFs are
stored in SOFA files (see <http://www.sofacoustics.org/> for a
database). SOFAlizer is developed at the Acoustics Research Institute
(ARI) of the Austrian Academy of Sciences.
To enable compilation of this filter you need to configure FFmpeg with
"--enable-libmysofa".
The filter accepts the following options:
sofa
Set the SOFA file used for rendering.
gain
Set gain applied to audio. Value is in dB. Default is 0.
rotation
Set rotation of virtual loudspeakers in deg. Default is 0.
elevation
Set elevation of virtual speakers in deg. Default is 0.
radius
Set distance in meters between loudspeakers and the listener with
near-field HRTFs. Default is 1.
type
Set processing type. Can be time or freq. time is processing audio
in time domain which is slow. freq is processing audio in
frequency domain which is fast. Default is freq.
speakers
Set custom positions of virtual loudspeakers. Syntax for this
option is: <CH> <AZIM> <ELEV>[|<CH> <AZIM> <ELEV>|...]. Each
virtual loudspeaker is described with short channel name following
with azimuth and elevation in degrees. Each virtual loudspeaker
description is separated by '|'. For example to override front
left and front right channel positions use: 'speakers=FL 45 15|FR
345 15'. Descriptions with unrecognised channel names are ignored.
lfegain
Set custom gain for LFE channels. Value is in dB. Default is 0.
Examples
路 Using ClubFritz6 sofa file:
sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1
路 Using ClubFritz12 sofa file and bigger radius with small rotation:
sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5
路 Similar as above but with custom speaker positions for front left,
front right, back left and back right and also with custom gain:
"sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28"
stereotools
This filter has some handy utilities to manage stereo signals, for
converting M/S stereo recordings to L/R signal while having control
over the parameters or spreading the stereo image of master track.
The filter accepts the following options:
level_in
Set input level before filtering for both channels. Defaults is 1.
Allowed range is from 0.015625 to 64.
level_out
Set output level after filtering for both channels. Defaults is 1.
Allowed range is from 0.015625 to 64.
balance_in
Set input balance between both channels. Default is 0. Allowed
range is from -1 to 1.
balance_out
Set output balance between both channels. Default is 0. Allowed
range is from -1 to 1.
softclip
Enable softclipping. Results in analog distortion instead of harsh
digital 0dB clipping. Disabled by default.
mutel
Mute the left channel. Disabled by default.
muter
Mute the right channel. Disabled by default.
phasel
Change the phase of the left channel. Disabled by default.
phaser
Change the phase of the right channel. Disabled by default.
mode
Set stereo mode. Available values are:
lr>lr
Left/Right to Left/Right, this is default.
lr>ms
Left/Right to Mid/Side.
ms>lr
Mid/Side to Left/Right.
lr>ll
Left/Right to Left/Left.
lr>rr
Left/Right to Right/Right.
lr>l+r
Left/Right to Left + Right.
lr>rl
Left/Right to Right/Left.
ms>ll
Mid/Side to Left/Left.
ms>rr
Mid/Side to Right/Right.
slev
Set level of side signal. Default is 1. Allowed range is from
0.015625 to 64.
sbal
Set balance of side signal. Default is 0. Allowed range is from -1
to 1.
mlev
Set level of the middle signal. Default is 1. Allowed range is
from 0.015625 to 64.
mpan
Set middle signal pan. Default is 0. Allowed range is from -1 to 1.
base
Set stereo base between mono and inversed channels. Default is 0.
Allowed range is from -1 to 1.
delay
Set delay in milliseconds how much to delay left from right channel
and vice versa. Default is 0. Allowed range is from -20 to 20.
sclevel
Set S/C level. Default is 1. Allowed range is from 1 to 100.
phase
Set the stereo phase in degrees. Default is 0. Allowed range is
from 0 to 360.
bmode_in, bmode_out
Set balance mode for balance_in/balance_out option.
Can be one of the following:
balance
Classic balance mode. Attenuate one channel at time. Gain is
raised up to 1.
amplitude
Similar as classic mode above but gain is raised up to 2.
power
Equal power distribution, from -6dB to +6dB range.
Examples
路 Apply karaoke like effect:
stereotools=mlev=0.015625
路 Convert M/S signal to L/R:
"stereotools=mode=ms>lr"
stereowiden
This filter enhance the stereo effect by suppressing signal common to
both channels and by delaying the signal of left into right and vice
versa, thereby widening the stereo effect.
The filter accepts the following options:
delay
Time in milliseconds of the delay of left signal into right and
vice versa. Default is 20 milliseconds.
feedback
Amount of gain in delayed signal into right and vice versa. Gives a
delay effect of left signal in right output and vice versa which
gives widening effect. Default is 0.3.
crossfeed
Cross feed of left into right with inverted phase. This helps in
suppressing the mono. If the value is 1 it will cancel all the
signal common to both channels. Default is 0.3.
drymix
Set level of input signal of original channel. Default is 0.8.
superequalizer
Apply 18 band equalizer.
The filter accepts the following options:
1b Set 65Hz band gain.
2b Set 92Hz band gain.
3b Set 131Hz band gain.
4b Set 185Hz band gain.
5b Set 262Hz band gain.
6b Set 370Hz band gain.
7b Set 523Hz band gain.
8b Set 740Hz band gain.
9b Set 1047Hz band gain.
10b Set 1480Hz band gain.
11b Set 2093Hz band gain.
12b Set 2960Hz band gain.
13b Set 4186Hz band gain.
14b Set 5920Hz band gain.
15b Set 8372Hz band gain.
16b Set 11840Hz band gain.
17b Set 16744Hz band gain.
18b Set 20000Hz band gain.
surround
Apply audio surround upmix filter.
This filter allows to produce multichannel output from audio stream.
The filter accepts the following options:
chl_out
Set output channel layout. By default, this is 5.1.
See the Channel Layout section in the ffmpeg-utils(1) manual for
the required syntax.
chl_in
Set input channel layout. By default, this is stereo.
See the Channel Layout section in the ffmpeg-utils(1) manual for
the required syntax.
level_in
Set input volume level. By default, this is 1.
level_out
Set output volume level. By default, this is 1.
lfe Enable LFE channel output if output channel layout has it. By
default, this is enabled.
lfe_low
Set LFE low cut off frequency. By default, this is 128 Hz.
lfe_high
Set LFE high cut off frequency. By default, this is 256 Hz.
fc_in
Set front center input volume. By default, this is 1.
fc_out
Set front center output volume. By default, this is 1.
lfe_in
Set LFE input volume. By default, this is 1.
lfe_out
Set LFE output volume. By default, this is 1.
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, t
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.
channels, c
Specify which channels to filter, by default all available are
filtered.
tremolo
Sinusoidal amplitude modulation.
The filter accepts the following options:
f Modulation frequency in Hertz. Modulation frequencies in the
subharmonic range (20 Hz or lower) will result in a tremolo effect.
This filter may also be used as a ring modulator by specifying a
modulation frequency higher than 20 Hz. Range is 0.1 - 20000.0.
Default value is 5.0 Hz.
d Depth of modulation as a percentage. Range is 0.0 - 1.0. Default
value is 0.5.
vibrato
Sinusoidal phase modulation.
The filter accepts the following options:
f Modulation frequency in Hertz. Range is 0.1 - 20000.0. Default
value is 5.0 Hz.
d Depth of modulation as a percentage. Range is 0.0 - 1.0. Default
value is 0.5.
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
路 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
路 Increase input audio power by 6 decibels using fixed-point
precision:
volume=volume=6dB:precision=fixed
路 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:
路 The mean square energy is approximately -27 dB, or 10^-2.7.
路 The largest sample is at -4 dB, or more precisely between -4 dB and
-5 dB.
路 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.
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.
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
路 Generate silence:
aevalsrc=0
路 Generate a sin signal with frequency of 440 Hz, set sample rate to
8000 Hz:
aevalsrc="sin(440*2*PI*t):s=8000"
路 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"
路 Generate white noise:
aevalsrc="-2+random(0)"
路 Generate an amplitude modulated signal:
aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"
路 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
路 Set the sample rate to 48000 Hz and the channel layout to
AV_CH_LAYOUT_MONO.
anullsrc=r=48000:cl=4
路 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
路 Read from file speech.txt, and synthesize the text using the
standard flite voice:
flite=textfile=speech.txt
路 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
路 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
路 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/>
anoisesrc
Generate a noise audio signal.
The filter accepts the following options:
sample_rate, r
Specify the sample rate. Default value is 48000 Hz.
amplitude, a
Specify the amplitude (0.0 - 1.0) of the generated audio stream.
Default value is 1.0.
duration, d
Specify the duration of the generated audio stream. Not specifying
this option results in noise with an infinite length.
color, colour, c
Specify the color of noise. Available noise colors are white, pink,
brown, blue and violet. Default color is white.
seed, s
Specify a value used to seed the PRNG.
nb_samples, n
Set the number of samples per each output frame, default is 1024.
Examples
路 Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate
and an amplitude of 0.5:
anoisesrc=d=60:c=pink:r=44100:a=0.5
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.
The expression can contain the following constants:
n The (sequential) number of the output audio frame, starting
from 0.
pts The PTS (Presentation TimeStamp) of the output audio frame,
expressed in TB units.
t The PTS of the output audio frame, expressed in seconds.
TB The timebase of the output audio frames.
Default is 1024.
Examples
路 Generate a simple 440 Hz sine wave:
sine
路 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
路 Generate a 1 kHz sine wave following "1602,1601,1602,1601,1602"
NTSC pattern:
sine=1000:samples_per_frame='st(0,mod(n,5)); 1602-not(not(eq(ld(0),1)+eq(ld(0),3)))'
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].
p Set what planes of frame filter will use for averaging. Default is
all.
avgblur
Apply average blur filter.
The filter accepts the following options:
sizeX
Set horizontal kernel size.
planes
Set which planes to filter. By default all planes are filtered.
sizeY
Set vertical kernel size, if zero it will be same as "sizeX".
Default is 0.
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.
bitplanenoise
Show and measure bit plane noise.
The filter accepts the following options:
bitplane
Set which plane to analyze. Default is 1.
filter
Filter out noisy pixels from "bitplane" set above. Default is
disabled.
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.
This filter exports frame metadata "lavfi.blackframe.pblack". The
value represents the percentage of pixels in the picture that are below
the threshold 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. By
default, the output terminates when the longest 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
grainmerge
and
average
burn
darken
difference
grainextract
divide
dodge
freeze
exclusion
extremity
glow
hardlight
hardmix
heat
lighten
linearlight
multiply
multiply128
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).
The "blend" filter also supports the framesync options.
Examples
路 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)))'
路 Apply linear horizontal transition from top layer to bottom layer:
blend=all_expr='A*(X/W)+B*(1-X/W)'
路 Apply 1x1 checkerboard effect:
blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'
路 Apply uncover left effect:
blend=all_expr='if(gte(N*SW+X,W),A,B)'
路 Apply uncover down effect:
blend=all_expr='if(gte(Y-N*SH,0),A,B)'
路 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)'
路 Split diagonally video and shows top and bottom layer on each side:
blend=all_expr='if(gt(X,Y*(W/H)),A,B)'
路 Display differences between the current and the previous frame:
tblend=all_mode=grainextract
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
路 Apply a boxblur filter with the luma, chroma, and alpha radii set
to 2:
boxblur=luma_radius=2:luma_power=1
boxblur=2:1
路 Set the luma radius to 2, and alpha and chroma radius to 0:
boxblur=2:1:cr=0:ar=0
路 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
bwdif
Deinterlace the input video ("bwdif" stands for "Bob Weaver
Deinterlacing Filter").
Motion adaptive deinterlacing based on yadif with the use of w3fdif and
cubic interpolation algorithms. 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.
The default value is "send_field".
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".
chromakey
YUV colorspace color/chroma 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.
yuv Signals that the color passed is already in YUV instead of RGB.
Literal colors like "green" or "red" don't make sense with this
enabled anymore. This can be used to pass exact YUV values as
hexadecimal numbers.
Examples
路 Make every green pixel in the input image transparent:
ffmpeg -i input.png -vf chromakey=green out.png
路 Overlay a greenscreen-video on top of a static black background.
ffmpeg -f lavfi -i color=c=black:s=1280x720 -i video.mp4 -shortest -filter_complex "[1:v]chromakey=0x70de77:0.1:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.mkv
ciescope
Display CIE color diagram with pixels overlaid onto it.
The filter accepts the following options:
system
Set color system.
ntsc, 470m
ebu, 470bg
smpte
240m
apple
widergb
cie1931
rec709, hdtv
uhdtv, rec2020
cie Set CIE system.
xyy
ucs
luv
gamuts
Set what gamuts to draw.
See "system" option for available values.
size, s
Set ciescope size, by default set to 512.
intensity, i
Set intensity used to map input pixel values to CIE diagram.
contrast
Set contrast used to draw tongue colors that are out of active
color system gamut.
corrgamma
Correct gamma displayed on scope, by default enabled.
showwhite
Show white point on CIE diagram, by default disabled.
gamma
Set input gamma. Used only with XYZ input color space.
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
qp Display quantization parameters using the chroma planes.
mv_type, mvt
Set motion vectors type to visualize. Includes MVs from all frames
unless specified by frame_type option.
Available flags for mv_type are:
fp forward predicted MVs
bp backward predicted MVs
frame_type, ft
Set frame type to visualize motion vectors of.
Available flags for frame_type are:
if intra-coded frames (I-frames)
pf predicted frames (P-frames)
bf bi-directionally predicted frames (B-frames)
Examples
路 Visualize forward predicted MVs of all frames using ffplay:
ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv_type=fp
路 Visualize multi-directionals MVs of P and B-Frames using ffplay:
ffplay -flags2 +export_mvs input.mp4 -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
路 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
路 Make every green pixel in the input image transparent:
ffmpeg -i input.png -vf colorkey=green out.png
路 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
路 Make video output darker:
colorlevels=rimin=0.058:gimin=0.058:bimin=0.058
路 Increase contrast:
colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96
路 Make video output lighter:
colorlevels=rimax=0.902:gimax=0.902:bimax=0.902
路 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:
<red>=<red>*<rr> + <blue>*<rb> + <green>*<rg> + <alpha>*<ra>
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
路 Convert source to grayscale:
colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3
路 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
fcc FCC
bt601
BT.601
bt470
BT.470
bt470bg
BT.470BG
smpte170m
SMPTE-170M
smpte240m
SMPTE-240M
bt2020
BT.2020
For example to convert from BT.601 to SMPTE-240M, use the command:
colormatrix=bt601:smpte240m
colorspace
Convert colorspace, transfer characteristics or color primaries. Input
video needs to have an even size.
The filter accepts the following options:
all Specify all color properties at once.
The accepted values are:
bt470m
BT.470M
bt470bg
BT.470BG
bt601-6-525
BT.601-6 525
bt601-6-625
BT.601-6 625
bt709
BT.709
smpte170m
SMPTE-170M
smpte240m
SMPTE-240M
bt2020
BT.2020
space
Specify output colorspace.
The accepted values are:
bt709
BT.709
fcc FCC
bt470bg
BT.470BG or BT.601-6 625
smpte170m
SMPTE-170M or BT.601-6 525
smpte240m
SMPTE-240M
ycgco
YCgCo
bt2020ncl
BT.2020 with non-constant luminance
trc Specify output transfer characteristics.
The accepted values are:
bt709
BT.709
bt470m
BT.470M
bt470bg
BT.470BG
gamma22
Constant gamma of 2.2
gamma28
Constant gamma of 2.8
smpte170m
SMPTE-170M, BT.601-6 625 or BT.601-6 525
smpte240m
SMPTE-240M
srgb
SRGB
iec61966-2-1
iec61966-2-1
iec61966-2-4
iec61966-2-4
xvycc
xvycc
bt2020-10
BT.2020 for 10-bits content
bt2020-12
BT.2020 for 12-bits content
primaries
Specify output color primaries.
The accepted values are:
bt709
BT.709
bt470m
BT.470M
bt470bg
BT.470BG or BT.601-6 625
smpte170m
SMPTE-170M or BT.601-6 525
smpte240m
SMPTE-240M
film
film
smpte431
SMPTE-431
smpte432
SMPTE-432
bt2020
BT.2020
jedec-p22
JEDEC P22 phosphors
range
Specify output color range.
The accepted values are:
tv TV (restricted) range
mpeg
MPEG (restricted) range
pc PC (full) range
jpeg
JPEG (full) range
format
Specify output color format.
The accepted values are:
yuv420p
YUV 4:2:0 planar 8-bits
yuv420p10
YUV 4:2:0 planar 10-bits
yuv420p12
YUV 4:2:0 planar 12-bits
yuv422p
YUV 4:2:2 planar 8-bits
yuv422p10
YUV 4:2:2 planar 10-bits
yuv422p12
YUV 4:2:2 planar 12-bits
yuv444p
YUV 4:4:4 planar 8-bits
yuv444p10
YUV 4:4:4 planar 10-bits
yuv444p12
YUV 4:4:4 planar 12-bits
fast
Do a fast conversion, which skips gamma/primary correction. This
will take significantly less CPU, but will be mathematically
incorrect. To get output compatible with that produced by the
colormatrix filter, use fast=1.
dither
Specify dithering mode.
The accepted values are:
none
No dithering
fsb Floyd-Steinberg dithering
wpadapt
Whitepoint adaptation mode.
The accepted values are:
bradford
Bradford whitepoint adaptation
vonkries
von Kries whitepoint adaptation
identity
identity whitepoint adaptation (i.e. no whitepoint adaptation)
iall
Override all input properties at once. Same accepted values as all.
ispace
Override input colorspace. Same accepted values as space.
iprimaries
Override input color primaries. Same accepted values as primaries.
itrc
Override input transfer characteristics. Same accepted values as
trc.
irange
Override input color range. Same accepted values as range.
The filter converts the transfer characteristics, color space and color
primaries to the specified user values. The output value, if not
specified, is set to a default value based on the "all" property. If
that property is also not specified, the filter will log an error. The
output color range and format default to the same value as the input
color range and format. The input transfer characteristics, color
space, color primaries and color range should be set on the input data.
If any of these are missing, the filter will log an error and no
conversion will take place.
For example to convert the input to SMPTE-240M, use the command:
colorspace=smpte240m
convolution
Apply convolution 3x3 or 5x5 filter.
The filter accepts the following options:
0m
1m
2m
3m Set matrix for each plane. Matrix is sequence of 9 or 25 signed
integers.
0rdiv
1rdiv
2rdiv
3rdiv
Set multiplier for calculated value for each plane.
0bias
1bias
2bias
3bias
Set bias for each plane. This value is added to the result of the
multiplication. Useful for making the overall image brighter or
darker. Default is 0.0.
Examples
路 Apply sharpen:
convolution="0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0"
路 Apply blur:
convolution="1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1/9:1/9:1/9:1/9"
路 Apply edge enhance:
convolution="0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:5:1:1:1:0:128:128:128"
路 Apply edge detect:
convolution="0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:5:5:5:1:0:128:128:128"
路 Apply laplacian edge detector which includes diagonals:
convolution="1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:5:5:5:1:0:128:128:0"
路 Apply emboss:
convolution="-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2"
convolve
Apply 2D convolution of video stream in frequency domain using second
stream as impulse.
The filter accepts the following options:
planes
Set which planes to process.
impulse
Set which impulse video frames will be processed, can be first or
all. Default is all.
The "convolve" filter also supports the framesync options.
copy
Copy the input video source unchanged to the output. This is mainly
useful for testing purposes.
coreimage
Video filtering on GPU using Apple's CoreImage API on OSX.
Hardware acceleration is based on an OpenGL context. Usually, this
means it is processed by video hardware. However, software-based OpenGL
implementations exist which means there is no guarantee for hardware
processing. It depends on the respective OSX.
There are many filters and image generators provided by Apple that come
with a large variety of options. The filter has to be referenced by its
name along with its options.
The coreimage filter accepts the following options:
list_filters
List all available filters and generators along with all their
respective options as well as possible minimum and maximum values
along with the default values.
list_filters=true
filter
Specify all filters by their respective name and options. Use
list_filters to determine all valid filter names and options.
Numerical options are specified by a float value and are
automatically clamped to their respective value range. Vector and
color options have to be specified by a list of space separated
float values. Character escaping has to be done. A special option
name "default" is available to use default options for a filter.
It is required to specify either "default" or at least one of the
filter options. All omitted options are used with their default
values. The syntax of the filter string is as follows:
filter=<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...][#<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...]][#...]
output_rect
Specify a rectangle where the output of the filter chain is copied
into the input image. It is given by a list of space separated
float values:
output_rect=x\ y\ width\ height
If not given, the output rectangle equals the dimensions of the
input image. The output rectangle is automatically cropped at the
borders of the input image. Negative values are valid for each
component.
output_rect=25\ 25\ 100\ 100
Several filters can be chained for successive processing without GPU-
HOST transfers allowing for fast processing of complex filter chains.
Currently, only filters with zero (generators) or exactly one (filters)
input image and one output image are supported. Also, transition
filters are not yet usable as intended.
Some filters generate output images with additional padding depending
on the respective filter kernel. The padding is automatically removed
to ensure the filter output has the same size as the input image.
For image generators, the size of the output image is determined by the
previous output image of the filter chain or the input image of the
whole filterchain, respectively. The generators do not use the pixel
information of this image to generate their output. However, the
generated output is blended onto this image, resulting in partial or
complete coverage of the output image.
The coreimagesrc video source can be used for generating input images
which are directly fed into the filter chain. By using it, providing
input images by another video source or an input video is not required.
Examples
路 List all filters available:
coreimage=list_filters=true
路 Use the CIBoxBlur filter with default options to blur an image:
coreimage=filter=CIBoxBlur@default
路 Use a filter chain with CISepiaTone at default values and
CIVignetteEffect with its center at 100x100 and a radius of 50
pixels:
coreimage=filter=CIBoxBlur@default#CIVignetteEffect@inputCenter=100\ 100@inputRadius=50
路 Use nullsrc and CIQRCodeGenerator to create a QR code for the
FFmpeg homepage, given as complete and escaped command-line for
Apple's standard bash shell:
ffmpeg -f lavfi -i nullsrc=s=100x100,coreimage=filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png
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.
exact
Enable exact cropping. If enabled, subsampled videos will be
cropped at exact width/height/x/y as specified and will not be
rounded to nearest smaller value. 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
路 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
路 Crop the central input area with size 100x100:
crop=100:100
路 Crop the central input area with size 2/3 of the input video:
crop=2/3*in_w:2/3*in_h
路 Crop the input video central square:
crop=out_w=in_h
crop=in_h
路 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
路 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
路 Keep only the bottom right quarter of the input image:
crop=in_w/2:in_h/2:in_w/2:in_h/2
路 Crop height for getting Greek harmony:
crop=in_w:1/PHI*in_w
路 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)
路 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)"
路 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 8-bit 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.
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 (".acv") to import the settings
from.
plot
Save Gnuplot script of the curves in specified file.
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
路 Increase slightly the middle level of blue:
curves=blue='0/0 0.5/0.58 1/1'
路 Vintage effect:
curves=r='0/0.11 .42/.51 1/0.95':g='0/0 0.50/0.48 1/1':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)"
路 The previous example can also be achieved with the associated
built-in preset:
curves=preset=vintage
路 Or simply:
curves=vintage
路 Use a Photoshop preset and redefine the points of the green
component:
curves=psfile='MyCurvesPresets/purple.acv':green='0/0 0.45/0.53 1/1'
路 Check out the curves of the "cross_process" profile using ffmpeg
and gnuplot:
ffmpeg -f lavfi -i color -vf curves=cross_process:plot=/tmp/curves.plt -frames:v 1 -f null -
gnuplot -p /tmp/curves.plt
datascope
Video data analysis filter.
This filter shows hexadecimal pixel values of part of video.
The filter accepts the following options:
size, s
Set output video size.
x Set x offset from where to pick pixels.
y Set y offset from where to pick pixels.
mode
Set scope mode, can be one of the following:
mono
Draw hexadecimal pixel values with white color on black
background.
color
Draw hexadecimal pixel values with input video pixel color on
black background.
color2
Draw hexadecimal pixel values on color background picked from
input video, the text color is picked in such way so its always
visible.
axis
Draw rows and columns numbers on left and top of video.
opacity
Set background opacity.
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, b
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.
coupling, c
If enabled, current pixel is changed if and only if all pixel
components are banded, e.g. banding detection threshold is
triggered for all color components. The default is disabled.
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.
deflicker
Remove temporal frame luminance variations.
It accepts the following options:
size, s
Set moving-average filter size in frames. Default is 5. Allowed
range is 2 - 129.
mode, m
Set averaging mode to smooth temporal luminance variations.
Available values are:
am Arithmetic mean
gm Geometric mean
hm Harmonic mean
qm Quadratic mean
cm Cubic mean
pm Power mean
median
Median
bypass
Do not actually modify frame. Useful when one only wants metadata.
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 1. This option is deprecated,
setting higher values should no longer be necessary and is not
recommended.
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
路 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.
despill
Remove unwanted contamination of foreground colors, caused by reflected
color of greenscreen or bluescreen.
This filter accepts the following options:
type
Set what type of despill to use.
mix Set how spillmap will be generated.
expand
Set how much to get rid of still remaining spill.
red Controls amount of red in spill area.
green
Controls amount of green in spill area. Should be -1 for
greenscreen.
blue
Controls amount of blue in spill area. Should be -1 for
bluescreen.
brightness
Controls brightness of spill area, preserving colors.
alpha
Modify alpha from generated spillmap.
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
displace
Displace pixels as indicated by second and third input stream.
It takes three input streams and outputs one stream, the first input is
the source, and second and third input are displacement maps.
The second input specifies how much to displace pixels along the
x-axis, while the third input specifies how much to displace pixels
along the y-axis. If one of displacement map streams terminates, last
frame from that displacement map will be used.
Note that once generated, displacements maps can be reused over and
over again.
A description of the accepted options follows.
edge
Set displace behavior for pixels that are out of range.
Available values are:
blank
Missing pixels are replaced by black pixels.
smear
Adjacent pixels will spread out to replace missing pixels.
wrap
Out of range pixels are wrapped so they point to pixels of
other side.
mirror
Out of range pixels will be replaced with mirrored pixels.
Default is smear.
Examples
路 Add ripple effect to rgb input of video size hd720:
ffmpeg -i INPUT -f lavfi -i nullsrc=s=hd720,lutrgb=128:128:128 -f lavfi -i nullsrc=s=hd720,geq='r=128+30*sin(2*PI*X/400+T):g=128+30*sin(2*PI*X/400+T):b=128+30*sin(2*PI*X/400+T)' -lavfi '[0][1][2]displace' OUTPUT
路 Add wave effect to rgb input of video size hd720:
ffmpeg -i INPUT -f lavfi -i nullsrc=hd720,geq='r=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):g=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):b=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T))' -lavfi '[1]split[x][y],[0][x][y]displace' OUTPUT
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
路 Draw a black box around the edge of the input image:
drawbox
路 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
路 Fill the box with pink color:
drawbox=x=10:y=10:w=100:h=100:color=pink@0.5:t=max
路 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
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
路 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
路 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".
line_spacing
Set the line spacing in pixels of the border to be drawn around the
box using box. The default value of line_spacing is 0.
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.
basetime
Set a start time for the count. Value is in microseconds. Only
applied in the deprecated strftime expansion mode. To emulate in
normal expansion mode use the "pts" function, supplying the start
time (in seconds) as the second argument.
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.
alpha
Draw the text applying alpha blending. The value can be a number
between 0.0 and 1.0. The expression accepts the same variables x,
y as well. 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).
tc24hmax
If set to 1, the output of the timecode option will wrap around at
24 hours. Default is 0 (disabled).
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.
Sequences 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. Takes one or two arguments.
The first argument is mandatory and specifies the metadata key.
The second argument is optional and specifies a default value, used
when the metadata key is not found or empty.
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 three
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. "gmtime" stands for the timestamp of the
frame formatted as UTC time; "localtime" stands for the timestamp
of the frame formatted as local time zone time.
The second argument is an offset added to the timestamp.
If the format is set to "localtime" or "gmtime", a third argument
may be supplied: a strftime() format string. By default, YYYY-MM-
DD HH:MM:SS format will be used.
Examples
路 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'"
路 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.
路 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"
路 Show the text at a random position, switching to a new position
every 30 seconds:
drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=if(eq(mod(t\,30)\,0)\,rand(0\,(w-text_w))\,x):y=if(eq(mod(t\,30)\,0)\,rand(0\,(h-text_h))\,y)"
路 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"
路 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"
路 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"
路 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'"
路 Use fontconfig to set the font. Note that the colons need to be
escaped.
drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg'
路 Print the date of a real-time encoding (see strftime(3)):
drawtext='fontfile=FreeSans.ttf:text=%{localtime\:%a %b %d %Y}'
路 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 }"
路 Horizontally align multiple separate texts. Note that max_glyph_a
and the fontsize value are included in the y offset.
drawtext=fontfile=FreeSans.ttf:text=DOG:fontsize=24:x=10:y=20+24-max_glyph_a,
drawtext=fontfile=FreeSans.ttf:text=cow:fontsize=24:x=80:y=20+24-max_glyph_a
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
路 Standard edge detection with custom values for the hysteresis
thresholding:
edgedetect=low=0.1:high=0.4
路 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 "1".
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
路 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
路 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
路 Fade out the last 45 frames of a 200-frame video:
fade=out:155:45
fade=type=out:start_frame=155:nb_frames=45
路 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
路 Make the first 5 frames yellow, then fade in from frame 5-24:
fade=in:5:20:color=yellow
路 Fade in alpha over first 25 frames of video:
fade=in:0:25:alpha=1
路 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.
eval
Set when the expressions are evaluated.
It accepts the following values:
init
Only evaluate expressions once during the filter
initialization.
frame
Evaluate expressions for each incoming frame.
Default value is init.
The filter accepts the following variables:
X
Y The coordinates of the current sample.
W
H The width and height of the image.
N The number of input frame, starting from 0.
Examples
路 High-pass:
fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'
路 Low-pass:
fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'
路 Sharpen:
fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'
路 Blur:
fftfilt=dc_Y=0:weight_Y='exp(-4 * ((Y+X)/(W+H)))'
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").
fieldhint
Create new frames by copying the top and bottom fields from surrounding
frames supplied as numbers by the hint file.
hint
Set file containing hints: absolute/relative frame numbers.
There must be one line for each frame in a clip. Each line must
contain two numbers separated by the comma, optionally followed by
"-" or "+". Numbers supplied on each line of file can not be out
of [N-1,N+1] where N is current frame number for "absolute" mode or
out of [-1, 1] range for "relative" mode. First number tells from
which frame to pick up top field and second number tells from which
frame to pick up bottom field.
If optionally followed by "+" output frame will be marked as
interlaced, else if followed by "-" output frame will be marked as
progressive, else it will be marked same as input frame. If line
starts with "#" or ";" that line is skipped.
mode
Can be item "absolute" or "relative". Default is "absolute".
Example of first several lines of "hint" file for "relative" mode:
0,0 - # first frame
1,0 - # second frame, use third's frame top field and second's frame bottom field
1,0 - # third frame, use fourth's frame top field and third's frame bottom field
1,0 -
0,0 -
0,0 -
1,0 -
1,0 -
1,0 -
0,0 -
0,0 -
1,0 -
1,0 -
1,0 -
0,0 -
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, afifo
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
路 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
路 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
floodfill
Flood area with values of same pixel components with another values.
It accepts the following options:
x Set pixel x coordinate.
y Set pixel y coordinate.
s0 Set source #0 component value.
s1 Set source #1 component value.
s2 Set source #2 component value.
s3 Set source #3 component value.
d0 Set destination #0 component value.
d1 Set destination #1 component value.
d2 Set destination #2 component value.
d3 Set destination #3 component value.
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
路 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.
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.
round
Timestamp (PTS) rounding method.
Possible values are:
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".
eof_action
Action performed when reading the last frame.
Possible values are:
round
Use same timestamp rounding method as used for other frames.
pass
Pass through last frame if input duration has not been reached
yet.
The default is "round".
Alternatively, the options can be specified as a flat string:
fps[:start_time[:round]].
See also the setpts filter.
Examples
路 A typical usage in order to set the fps to 25:
fps=fps=25
路 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
FREI0R_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
路 Apply the distort0r effect, setting the first two double
parameters:
frei0r=filter_name=distort0r:filter_params=0.5|0.01
路 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
路 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).
gblur
Apply Gaussian blur filter.
The filter accepts the following options:
sigma
Set horizontal sigma, standard deviation of Gaussian blur. Default
is 0.5.
steps
Set number of steps for Gaussian approximation. Defauls is 1.
planes
Set which planes to filter. By default all planes are filtered.
sigmaV
Set vertical sigma, if negative it will be same as "sigma".
Default is "-1".
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
路 Flip the image horizontally:
geq=p(W-X\,Y)
路 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
路 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
路 Generate a quick emboss effect:
format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2'
路 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)'
路 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
gradfun
Fix the banding artifacts that are sometimes introduced into nearly
flat regions by truncation to 8-bit 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
路 Apply the filter with a 3.5 strength and radius of 8:
gradfun=3.5:8
路 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.
Standard histogram 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.
The filter accepts the following options:
level_height
Set height of level. Default value is 200. Allowed range is [50,
2048].
scale_height
Set height of color scale. Default value is 12. Allowed range is
[0, 40].
display_mode
Set display mode. It accepts the following values:
stack
Per color component graphs are placed below each other.
parade
Per color component graphs are placed side by side.
overlay
Presents information identical to that in the "parade", except
that the graphs representing color components are superimposed
directly over one another.
Default is "stack".
levels_mode
Set mode. Can be either "linear", or "logarithmic". Default is
"linear".
components
Set what color components to display. Default is 7.
fgopacity
Set foreground opacity. Default is 0.7.
bgopacity
Set background opacity. Default is 0.5.
Examples
路 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.
hwdownload
Download hardware frames to system memory.
The input must be in hardware frames, and the output a non-hardware
format. Not all formats will be supported on the output - it may be
necessary to insert an additional format filter immediately following
in the graph to get the output in a supported format.
hwmap
Map hardware frames to system memory or to another device.
This filter has several different modes of operation; which one is used
depends on the input and output formats:
路 Hardware frame input, normal frame output
Map the input frames to system memory and pass them to the output.
If the original hardware frame is later required (for example,
after overlaying something else on part of it), the hwmap filter
can be used again in the next mode to retrieve it.
路 Normal frame input, hardware frame output
If the input is actually a software-mapped hardware frame, then
unmap it - that is, return the original hardware frame.
Otherwise, a device must be provided. Create new hardware surfaces
on that device for the output, then map them back to the software
format at the input and give those frames to the preceding filter.
This will then act like the hwupload filter, but may be able to
avoid an additional copy when the input is already in a compatible
format.
路 Hardware frame input and output
A device must be supplied for the output, either directly or with
the derive_device option. The input and output devices must be of
different types and compatible - the exact meaning of this is
system-dependent, but typically it means that they must refer to
the same underlying hardware context (for example, refer to the
same graphics card).
If the input frames were originally created on the output device,
then unmap to retrieve the original frames.
Otherwise, map the frames to the output device - create new
hardware frames on the output corresponding to the frames on the
input.
The following additional parameters are accepted:
mode
Set the frame mapping mode. Some combination of:
read
The mapped frame should be readable.
write
The mapped frame should be writeable.
overwrite
The mapping will always overwrite the entire frame.
This may improve performance in some cases, as the original
contents of the frame need not be loaded.
direct
The mapping must not involve any copying.
Indirect mappings to copies of frames are created in some cases
where either direct mapping is not possible or it would have
unexpected properties. Setting this flag ensures that the
mapping is direct and will fail if that is not possible.
Defaults to read+write if not specified.
derive_device type
Rather than using the device supplied at initialisation, instead
derive a new device of type type from the device the input frames
exist on.
reverse
In a hardware to hardware mapping, map in reverse - create frames
in the sink and map them back to the source. This may be necessary
in some cases where a mapping in one direction is required but only
the opposite direction is supported by the devices being used.
This option is dangerous - it may break the preceding filter in
undefined ways if there are any additional constraints on that
filter's output. Do not use it without fully understanding the
implications of its use.
hwupload
Upload system memory frames to hardware surfaces.
The device to upload to must be supplied when the filter is
initialised. If using ffmpeg, select the appropriate device with the
-filter_hw_device option.
hwupload_cuda
Upload system memory frames to a CUDA device.
It accepts the following optional parameters:
device
The number of the CUDA device to use
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:
inputs
Set number of input streams. Default is 2.
shortest
If set to 1, force the output to terminate when the shortest input
terminates. Default value is 0.
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
路 Set the hue to 90 degrees and the saturation to 1.0:
hue=h=90:s=1
路 Same command but expressing the hue in radians:
hue=H=PI/2:s=1
路 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"
路 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))"
路 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.
hysteresis
Grow first stream into second stream by connecting components. This
makes it possible to build more robust edge masks.
This filter accepts the following options:
planes
Set which planes will be processed as bitmap, unprocessed planes
will be copied from first stream. By default value 0xf, all planes
will be processed.
threshold
Set threshold which is used in filtering. If pixel component value
is higher than this value filter algorithm for connecting
components is activated. By default value is 0.
idet
Detect video interlacing type.
This filter tries to detect if the input frames are interlaced,
progressive, top or bottom field first. It will also try to 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.
rep_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 its
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
Vertical lowpass filter to avoid twitter interlacing and reduce
moire patterns.
0, off
Disable vertical lowpass filter
1, linear
Enable linear filter (default)
2, complex
Enable complex filter. This will slightly less reduce twitter
and moire but better retain detail and subjective sharpness
impression.
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
路 Apply default values:
kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0
路 Enable additional sharpening:
kerndeint=sharp=1
路 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.
libvmaf
Obtain the average VMAF (Video Multi-Method Assessment Fusion) score
between two input videos.
This filter takes 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 obtained average VMAF score is printed through the logging system.
It requires Netflix's vmaf library (libvmaf) as a pre-requisite. After
installing the library it can be enabled using: "./configure
--enable-libvmaf". If no model path is specified it uses the default
model: "vmaf_v0.6.1.pkl".
On the below examples the input file main.mpg being processed is
compared with the reference file ref.mpg.
The filter has following options:
model_path
Set the model path which is to be used for SVM. Default value:
"vmaf_v0.6.1.pkl"
log_path
Set the file path to be used to store logs.
log_fmt
Set the format of the log file (xml or json).
enable_transform
Enables transform for computing vmaf.
phone_model
Invokes the phone model which will generate VMAF scores higher than
in the regular model, which is more suitable for laptop, TV, etc.
viewing conditions.
psnr
Enables computing psnr along with vmaf.
ssim
Enables computing ssim along with vmaf.
ms_ssim
Enables computing ms_ssim along with vmaf.
pool
Set the pool method to be used for computing vmaf.
This filter also supports the framesync options.
For example:
ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf -f null -
Example with options:
ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf="psnr=1:enable-transform=1" -f null -
limiter
Limits the pixel components values to the specified range [min, max].
The filter accepts the following options:
min Lower bound. Defaults to the lowest allowed value for the input.
max Upper bound. Defaults to the highest allowed value for the input.
planes
Specify which planes will be processed. Defaults to all available.
loop
Loop video frames.
The filter accepts the following options:
loop
Set the number of loops.
size
Set maximal size in number of frames.
start
Set first frame of loop.
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.
This filter also supports the framesync options.
lumakey
Turn certain luma values into transparency.
The filter accepts the following options:
threshold
Set the luma which will be used as base for transparency. Default
value is 0.
tolerance
Set the range of luma values to be keyed out. Default value is 0.
softness
Set the range of softness. Default value is 0. Use this to control
gradual transition from zero to full transparency.
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
路 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"
路 Negate luminance:
lutyuv=y=negval
路 Remove chroma components, turning the video into a graytone image:
lutyuv="u=128:v=128"
路 Apply a luma burning effect:
lutyuv="y=2*val"
路 Remove green and blue components:
lutrgb="g=0:b=0"
路 Set a constant alpha channel value on input:
format=rgba,lutrgb=a="maxval-minval/2"
路 Correct luminance gamma by a factor of 0.5:
lutyuv=y=gammaval(0.5)
路 Discard least significant bits of luma:
lutyuv=y='bitand(val, 128+64+32)'
路 Technicolor like effect:
lutyuv=u='(val-maxval/2)*2+maxval/2':v='(val-maxval/2)*2+maxval/2'
lut2, tlut2
The "lut2" filter takes two input streams and outputs one stream.
The "tlut2" (time lut2) filter takes two consecutive frames from one
single stream.
This filter accepts 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
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 inputs.
The expressions can contain the following constants:
w
h The input width and height.
x The first input value for the pixel component.
y The second input value for the pixel component.
bdx The first input video bit depth.
bdy The second input video bit depth.
All expressions default to "x".
Examples
路 Highlight differences between two RGB video streams:
lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1)'
路 Highlight differences between two YUV video streams:
lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1)'
路 Show max difference between two video streams:
lut2='if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1)))'
maskedclamp
Clamp the first input stream with the second input and third input
stream.
Returns the value of first stream to be between second input stream -
"undershoot" and third input stream + "overshoot".
This filter accepts the following options:
undershoot
Default value is 0.
overshoot
Default value is 0.
planes
Set which planes will be processed as bitmap, unprocessed planes
will be copied from first stream. By default value 0xf, all planes
will be processed.
maskedmerge
Merge the first input stream with the second input stream using per
pixel weights in the third input stream.
A value of 0 in the third stream pixel component means that pixel
component from first stream is returned unchanged, while maximum value
(eg. 255 for 8-bit videos) means that pixel component from second
stream is returned unchanged. Intermediate values define the amount of
merging between both input stream's pixel components.
This filter accepts the following options:
planes
Set which planes will be processed as bitmap, unprocessed planes
will be copied from first stream. By default value 0xf, all planes
will be processed.
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.
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
路 Merge three gray video streams of same width and height into single
video stream:
[a0][a1][a2]mergeplanes=0x001020:yuv444p
路 Merge 1st yuv444p stream and 2nd gray video stream into yuva444p
video stream:
[a0][a1]mergeplanes=0x00010210:yuva444p
路 Swap Y and A plane in yuva444p stream:
format=yuva444p,mergeplanes=0x03010200:yuva444p
路 Swap U and V plane in yuv420p stream:
format=yuv420p,mergeplanes=0x000201:yuv420p
路 Cast a rgb24 clip to yuv444p:
format=rgb24,mergeplanes=0x000102:yuv444p
mestimate
Estimate and export motion vectors using block matching algorithms.
Motion vectors are stored in frame side data to be used by other
filters.
This filter accepts the following options:
method
Specify the motion estimation method. Accepts one of the following
values:
esa Exhaustive search algorithm.
tss Three step search algorithm.
tdls
Two dimensional logarithmic search algorithm.
ntss
New three step search algorithm.
fss Four step search algorithm.
ds Diamond search algorithm.
hexbs
Hexagon-based search algorithm.
epzs
Enhanced predictive zonal search algorithm.
umh Uneven multi-hexagon search algorithm.
Default value is esa.
mb_size
Macroblock size. Default 16.
search_param
Search parameter. Default 7.
midequalizer
Apply Midway Image Equalization effect using two video streams.
Midway Image Equalization adjusts a pair of images to have the same
histogram, while maintaining their dynamics as much as possible. It's
useful for e.g. matching exposures from a pair of stereo cameras.
This filter has two inputs and one output, which must be of same pixel
format, but may be of different sizes. The output of filter is first
input adjusted with midway histogram of both inputs.
This filter accepts the following option:
planes
Set which planes to process. Default is 15, which is all available
planes.
minterpolate
Convert the video to specified frame rate using motion interpolation.
This filter accepts the following options:
fps Specify the output frame rate. This can be rational e.g.
"60000/1001". Frames are dropped if fps is lower than source fps.
Default 60.
mi_mode
Motion interpolation mode. Following values are accepted:
dup Duplicate previous or next frame for interpolating new ones.
blend
Blend source frames. Interpolated frame is mean of previous and
next frames.
mci Motion compensated interpolation. Following options are
effective when this mode is selected:
mc_mode
Motion compensation mode. Following values are accepted:
obmc
Overlapped block motion compensation.
aobmc
Adaptive overlapped block motion compensation. Window
weighting coefficients are controlled adaptively
according to the reliabilities of the neighboring
motion vectors to reduce oversmoothing.
Default mode is obmc.
me_mode
Motion estimation mode. Following values are accepted:
bidir
Bidirectional motion estimation. Motion vectors are
estimated for each source frame in both forward and
backward directions.
bilat
Bilateral motion estimation. Motion vectors are
estimated directly for interpolated frame.
Default mode is bilat.
me The algorithm to be used for motion estimation. Following
values are accepted:
esa Exhaustive search algorithm.
tss Three step search algorithm.
tdls
Two dimensional logarithmic search algorithm.
ntss
New three step search algorithm.
fss Four step search algorithm.
ds Diamond search algorithm.
hexbs
Hexagon-based search algorithm.
epzs
Enhanced predictive zonal search algorithm.
umh Uneven multi-hexagon search algorithm.
Default algorithm is epzs.
mb_size
Macroblock size. Default 16.
search_param
Motion estimation search parameter. Default 32.
vsbmc
Enable variable-size block motion compensation. Motion
estimation is applied with smaller block sizes at object
boundaries in order to make the them less blur. Default is
0 (disabled).
scd Scene change detection method. Scene change leads motion vectors to
be in random direction. Scene change detection replace interpolated
frames by duplicate ones. May not be needed for other modes.
Following values are accepted:
none
Disable scene change detection.
fdiff
Frame difference. Corresponding pixel values are compared and
if it satisfies scd_threshold scene change is detected.
Default method is fdiff.
scd_threshold
Scene change detection threshold. Default is 5.0.
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 disregarding 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.
nlmeans
Denoise frames using Non-Local Means algorithm.
Each pixel is adjusted by looking for other pixels with similar
contexts. This context similarity is defined by comparing their
surrounding patches of size pxp. Patches are searched in an area of rxr
around the pixel.
Note that the research area defines centers for patches, which means
some patches will be made of pixels outside that research area.
The filter accepts the following options.
s Set denoising strength.
p Set patch size.
pc Same as p but for chroma planes.
The default value is 0 and means automatic.
r Set research size.
rc Same as r but for chroma planes.
The default value is 0 and means automatic.
nnedi
Deinterlace video using neural network edge directed interpolation.
This filter accepts the following options:
weights
Mandatory option, without binary file filter can not work.
Currently file can be found here:
https://github.com/dubhater/vapoursynth-nnedi3/blob/master/src/nnedi3_weights.bin
deint
Set which frames to deinterlace, by default it is "all". Can be
"all" or "interlaced".
field
Set mode of operation.
Can be one of the following:
af Use frame flags, both fields.
a Use frame flags, single field.
t Use top field only.
b Use bottom field only.
tf Use both fields, top first.
bf Use both fields, bottom first.
planes
Set which planes to process, by default filter process all frames.
nsize
Set size of local neighborhood around each pixel, used by the
predictor neural network.
Can be one of the following:
s8x6
s16x6
s32x6
s48x6
s8x4
s16x4
s32x4
nns Set the number of neurons in predictor neural network. Can be one
of the following:
n16
n32
n64
n128
n256
qual
Controls the number of different neural network predictions that
are blended together to compute the final output value. Can be
"fast", default or "slow".
etype
Set which set of weights to use in the predictor. Can be one of
the following:
a weights trained to minimize absolute error
s weights trained to minimize squared error
pscrn
Controls whether or not the prescreener neural network is used to
decide which pixels should be processed by the predictor neural
network and which can be handled by simple cubic interpolation.
The prescreener is trained to know whether cubic interpolation will
be sufficient for a pixel or whether it should be predicted by the
predictor nn. The computational complexity of the prescreener nn
is much less than that of the predictor nn. Since most pixels can
be handled by cubic interpolation, using the prescreener generally
results in much faster processing. The prescreener is pretty
accurate, so the difference between using it and not using it is
almost always unnoticeable.
Can be one of the following:
none
original
new
Default is "new".
fapprox
Set various debugging flags.
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
pix_fmts=yuv420p|monow|rgb24".
Examples
路 Force libavfilter to use a format different from yuv420p for the
input to the vflip filter:
noformat=pix_fmts=yuv420p,vflip
路 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.
ocr
Optical Character Recognition
This filter uses Tesseract for optical character recognition.
It accepts the following options:
datapath
Set datapath to tesseract data. Default is to use whatever was set
at installation.
language
Set language, default is "eng".
whitelist
Set character whitelist.
blacklist
Set character blacklist.
The filter exports recognized text as the frame metadata
"lavfi.ocr.text".
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".
oscilloscope
2D Video Oscilloscope.
Useful to measure spatial impulse, step responses, chroma delays, etc.
It accepts the following parameters:
x Set scope center x position.
y Set scope center y position.
s Set scope size, relative to frame diagonal.
t Set scope tilt/rotation.
o Set trace opacity.
tx Set trace center x position.
ty Set trace center y position.
tw Set trace width, relative to width of frame.
th Set trace height, relative to height of frame.
c Set which components to trace. By default it traces first three
components.
g Draw trace grid. By default is enabled.
st Draw some statistics. By default is enabled.
sc Draw scope. By default is enabled.
Examples
路 Inspect full first row of video frame.
oscilloscope=x=0.5:y=0:s=1
路 Inspect full last row of video frame.
oscilloscope=x=0.5:y=1:s=1
路 Inspect full 5th line of video frame of height 1080.
oscilloscope=x=0.5:y=5/1080:s=1
路 Inspect full last column of video frame.
oscilloscope=x=1:y=0.5:s=1:t=1
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
See framesync.
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
See framesync.
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 packed RGB output
gbrp
force planar RGB output
auto
automatically pick format
Default value is yuv420.
repeatlast
See framesync.
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.
This filter also supports the framesync options.
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
路 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
路 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
路 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
路 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]
路 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'
路 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
路 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]
"
路 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
路 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.
If x or y evaluate to a negative number, they'll be changed so the
input image is centered on the padded area.
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".
eval
Specify when to evaluate width, height, x and y expression.
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.
aspect
Pad to aspect instead to a resolution.
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
路 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
路 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"
路 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"
路 Pad the input to get a final w/h ratio of 16:9:
pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"
路 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"
路 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.
single
Compute new histogram for each frame.
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
路 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.
new Take new palette for each output frame.
Examples
路 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.
in Input frame count.
on Output frame count.
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.
eval
Set when the expressions for coordinates x0,y0,...x3,y3 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.
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.
pixscope
Display sample values of color channels. Mainly useful for checking
color and levels. Minimum supported resolution is 640x480.
The filters accept the following options:
x Set scope X position, relative offset on X axis.
y Set scope Y position, relative offset on Y axis.
w Set scope width.
h Set scope height.
o Set window opacity. This window also holds statistics about pixel
area.
wx Set window X position, relative offset on X axis.
wy Set window Y position, relative offset on Y axis.
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
路 Apply horizontal and vertical deblocking, deringing and automatic
brightness/contrast:
pp=hb/vb/dr/al
路 Apply default filters without brightness/contrast correction:
pp=de/-al
路 Apply default filters and temporal denoiser:
pp=default/tmpnoise|1|2|3
路 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).
premultiply
Apply alpha premultiply effect to input video stream using first plane
of second stream as alpha.
Both streams must have same dimensions and same pixel format.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be
copied. By default value 0xf, all planes will be processed.
inplace
Do not require 2nd input for processing, instead use alpha plane
from input stream.
prewitt
Apply prewitt operator to input video stream.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be
copied. By default value 0xf, all planes will be processed.
scale
Set value which will be multiplied with filtered result.
delta
Set value which will be added to filtered result.
pseudocolor
Alter frame colors in video with pseudocolors.
This filter accept the following options:
c0 set pixel first component expression
c1 set pixel second component expression
c2 set pixel third component expression
c3 set pixel fourth component expression, corresponds to the alpha
component
i set component to use as base for altering colors
Each of them specifies the expression to use for computing the lookup
table for the corresponding pixel component values.
The expressions can contain the following constants and functions:
w
h The input width and height.
val The input value for the pixel component.
ymin, umin, vmin, amin
The minimum allowed component value.
ymax, umax, vmax, amax
The maximum allowed component value.
All expressions default to "val".
Examples
路 Change too high luma values to gradient:
pseudocolor="'if(between(val,ymax,amax),lerp(ymin,ymax,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(umax,umin,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(vmin,vmax,(val-ymax)/(amax-ymax)),-1):-1'"
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. When filename equals "-" the data is sent to
standard output.
stats_version
Specifies which version of the stats file format to use. Details of
each format are written below. Default value is 1.
stats_add_max
Determines whether the max value is output to the stats log.
Default value is 0. Requires stats_version >= 2. If this is set
and stats_version < 2, the filter will return an error.
This filter also supports the framesync options.
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.
If a stats_version greater than 1 is specified, a header line precedes
the list of per-frame-pair stats, with key value pairs following the
frame format with the following parameters:
psnr_log_version
The version of the log file format. Will match stats_version.
fields
A comma separated list of the per-frame-pair parameters included in
the log.
A description of each shown per-frame-pair 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.
max_avg, max_y, max_u, max_v
Maximum allowed value for each channel, and average over all
channels.
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 Sequential index starting from -129 to 128.
Examples
路 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.
readeia608
Read closed captioning (EIA-608) information from the top lines of a
video frame.
This filter adds frame metadata for "lavfi.readeia608.X.cc" and
"lavfi.readeia608.X.line", where "X" is the number of the identified
line with EIA-608 data (starting from 0). A description of each
metadata value follows:
lavfi.readeia608.X.cc
The two bytes stored as EIA-608 data (printed in hexadecimal).
lavfi.readeia608.X.line
The number of the line on which the EIA-608 data was identified and
read.
This filter accepts the following options:
scan_min
Set the line to start scanning for EIA-608 data. Default is 0.
scan_max
Set the line to end scanning for EIA-608 data. Default is 29.
mac Set minimal acceptable amplitude change for sync codes detection.
Default is 0.2. Allowed range is "[0.001 - 1]".
spw Set the ratio of width reserved for sync code detection. Default
is 0.27. Allowed range is "[0.01 - 0.7]".
mhd Set the max peaks height difference for sync code detection.
Default is 0.1. Allowed range is "[0.0 - 0.5]".
mpd Set max peaks period difference for sync code detection. Default
is 0.1. Allowed range is "[0.0 - 0.5]".
msd Set the first two max start code bits differences. Default is
0.02. Allowed range is "[0.0 - 0.5]".
bhd Set the minimum ratio of bits height compared to 3rd start code
bit. Default is 0.75. Allowed range is "[0.01 - 1]".
th_w
Set the white color threshold. Default is 0.35. Allowed range is
"[0.1 - 1]".
th_b
Set the black color threshold. Default is 0.15. Allowed range is
"[0.0 - 0.5]".
chp Enable checking the parity bit. In the event of a parity error, the
filter will output 0x00 for that character. Default is false.
Examples
路 Output a csv with presentation time and the first two lines of
identified EIA-608 captioning data.
ffprobe -f lavfi -i movie=captioned_video.mov,readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.cc,lavfi.readeia608.1.cc -of csv
readvitc
Read vertical interval timecode (VITC) information from the top lines
of a video frame.
The filter adds frame metadata key "lavfi.readvitc.tc_str" with the
timecode value, if a valid timecode has been detected. Further metadata
key "lavfi.readvitc.found" is set to 0/1 depending on whether timecode
data has been found or not.
This filter accepts the following options:
scan_max
Set the maximum number of lines to scan for VITC data. If the value
is set to "-1" the full video frame is scanned. Default is 45.
thr_b
Set the luma threshold for black. Accepts float numbers in the
range [0.0,1.0], default value is 0.2. The value must be equal or
less than "thr_w".
thr_w
Set the luma threshold for white. Accepts float numbers in the
range [0.0,1.0], default value is 0.6. The value must be equal or
greater than "thr_b".
Examples
路 Detect and draw VITC data onto the video frame; if no valid VITC is
detected, draw "--:--:--:--" as a placeholder:
ffmpeg -i input.avi -filter:v 'readvitc,drawtext=fontfile=FreeMono.ttf:text=%{metadata\\:lavfi.readvitc.tc_str\\:--\\\\\\:--\\\\\\:--\\\\\\:--}:x=(w-tw)/2:y=400-ascent'
remap
Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.
Destination pixel at position (X, Y) will be picked from source (x, y)
position where x = Xmap(X, Y) and y = Ymap(X, Y). If mapping values are
out of range, zero value for pixel will be used for destination pixel.
Xmap and Ymap input video streams must be of same dimensions. Output
video stream will have Xmap/Ymap video stream dimensions. Xmap and
Ymap input video streams are 16bit depth, single channel.
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
Reverse a video clip.
Warning: This filter requires memory to buffer the entire clip, so
trimming is suggested.
Examples
路 Take the first 5 seconds of a clip, and reverse it.
trim=end=5,reverse
roberts
Apply roberts cross operator to input video stream.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be
copied. By default value 0xf, all planes will be processed.
scale
Set value which will be multiplied with filtered result.
delta
Set value which will be added to filtered result.
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
路 Rotate the input by PI/6 radians clockwise:
rotate=PI/6
路 Rotate the input by PI/6 radians counter-clockwise:
rotate=-PI/6
路 Rotate the input by 45 degrees clockwise:
rotate=45*PI/180
路 Apply a constant rotation with period T, starting from an angle of
PI/3:
rotate=PI/3+2*PI*t/T
路 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)
路 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'
路 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 -0.9-4.0.
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 -0.9-2.0
range.
chroma_strength, cs
Set chroma maximum difference between pixels to still be
considered, must be a value in the -0.9-100.0 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 width or w value is 0, the input width is used for the
output. If the height or h value is 0, the input height is used for
the output.
If one and only one of the values is -n with n >= 1, the scale
filter will 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.
If both values are -n with n >= 1, the behavior will be identical
to both values being set to 0 as previously detailed.
See below for the list of accepted constants for use in the
dimension expression.
eval
Specify when to evaluate width and height expression. 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.
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.
param0, param1
Set libswscale input parameters for scaling algorithms that need
them. See the ffmpeg-scaler manual for the complete documentation.
If not explicitly specified the filter applies empty parameters.
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:
路 ITU Radiocommunication Sector (ITU-R) Recommendation BT.601
路 ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G
路 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
路 Scale the input video to a size of 200x100
scale=w=200:h=100
This is equivalent to:
scale=200:100
or:
scale=200x100
路 Specify a size abbreviation for the output size:
scale=qcif
which can also be written as:
scale=size=qcif
路 Scale the input to 2x:
scale=w=2*iw:h=2*ih
路 The above is the same as:
scale=2*in_w:2*in_h
路 Scale the input to 2x with forced interlaced scaling:
scale=2*iw:2*ih:interl=1
路 Scale the input to half size:
scale=w=iw/2:h=ih/2
路 Increase the width, and set the height to the same size:
scale=3/2*iw:ow
路 Seek Greek harmony:
scale=iw:1/PHI*iw
scale=ih*PHI:ih
路 Increase the height, and set the width to 3/2 of the height:
scale=w=3/2*oh:h=3/5*ih
路 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"
路 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.
scale_npp
Use the NVIDIA Performance Primitives (libnpp) to perform scaling
and/or pixel format conversion on CUDA video frames. Setting the output
width and height works in the same way as for the scale filter.
The following additional options are accepted:
format
The pixel format of the output CUDA frames. If set to the string
"same" (the default), the input format will be kept. Note that
automatic format negotiation and conversion is not yet supported
for hardware frames
interp_algo
The interpolation algorithm used for resizing. One of the
following:
nn Nearest neighbour.
linear
cubic
cubic2p_bspline
2-parameter cubic (B=1, C=0)
cubic2p_catmullrom
2-parameter cubic (B=0, C=1/2)
cubic2p_b05c03
2-parameter cubic (B=1/2, C=3/10)
super
Supersampling
lanczos
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.
scale2ref also supports the following additional constants for the w
and h options:
main_w
main_h
The main input video's width and height
main_a
The same as main_w / main_h
main_sar
The main input video's sample aspect ratio
main_dar, mdar
The main input video's display aspect ratio. Calculated from
"(main_w / main_h) * main_sar".
main_hsub
main_vsub
The main input video's horizontal and vertical chroma subsample
values. For example for the pixel format "yuv422p" hsub is 2 and
vsub is 1.
Examples
路 Scale a subtitle stream (b) to match the main video (a) in size
before overlaying
'scale2ref[b][a];[a][b]overlay'
selectivecolor
Adjust cyan, magenta, yellow and black (CMYK) to certain ranges of
colors (such as "reds", "yellows", "greens", "cyans", ...). The
adjustment range is defined by the "purity" of the color (that is, how
saturated it already is).
This filter is similar to the Adobe Photoshop Selective Color tool.
The filter accepts the following options:
correction_method
Select color correction method.
Available values are:
absolute
Specified adjustments are applied "as-is" (added/subtracted to
original pixel component value).
relative
Specified adjustments are relative to the original component
value.
Default is "absolute".
reds
Adjustments for red pixels (pixels where the red component is the
maximum)
yellows
Adjustments for yellow pixels (pixels where the blue component is
the minimum)
greens
Adjustments for green pixels (pixels where the green component is
the maximum)
cyans
Adjustments for cyan pixels (pixels where the red component is the
minimum)
blues
Adjustments for blue pixels (pixels where the blue component is the
maximum)
magentas
Adjustments for magenta pixels (pixels where the green component is
the minimum)
whites
Adjustments for white pixels (pixels where all components are
greater than 128)
neutrals
Adjustments for all pixels except pure black and pure white
blacks
Adjustments for black pixels (pixels where all components are
lesser than 128)
psfile
Specify a Photoshop selective color file (".asv") to import the
settings from.
All the adjustment settings (reds, yellows, ...) accept up to 4 space
separated floating point adjustment values in the [-1,1] range,
respectively to adjust the amount of cyan, magenta, yellow and black
for the pixels of its range.
Examples
路 Increase cyan by 50% and reduce yellow by 33% in every green areas,
and increase magenta by 27% in blue areas:
selectivecolor=greens=.5 0 -.33 0:blues=0 .27
路 Use a Photoshop selective color preset:
selectivecolor=psfile=MySelectiveColorPresets/Misty.asv
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:
<DAR> = <HORIZONTAL_RESOLUTION> / <VERTICAL_RESOLUTION> * <SAR>
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
路 To change the display aspect ratio to 16:9, specify one of the
following:
setdar=dar=1.77777
setdar=dar=16/9
路 To change the sample aspect ratio to 10:11, specify:
setsar=sar=10/11
路 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).
shuffleframes
Reorder and/or duplicate and/or drop video frames.
It accepts the following parameters:
mapping
Set the destination indexes of input frames. This is space or '|'
separated list of indexes that maps input frames to output frames.
Number of indexes also sets maximal value that each index may have.
'-1' index have special meaning and that is to drop frame.
The first frame has the index 0. The default is to keep the input
unchanged.
Examples
路 Swap second and third frame of every three frames of the input:
ffmpeg -i INPUT -vf "shuffleframes=0 2 1" OUTPUT
路 Swap 10th and 1st frame of every ten frames of the input:
ffmpeg -i INPUT -vf "shuffleframes=9 1 2 3 4 5 6 7 8 0" OUTPUT
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.
Examples
路 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].
YBITDEPTH
Display bit depth of Y plane in current frame. Expressed in range
of [0-16].
UBITDEPTH
Display bit depth of U plane in current frame. Expressed in range
of [0-16].
VBITDEPTH
Display bit depth of V plane in current frame. Expressed in range
of [0-16].
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
路 Output data of various video metrics:
ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames
路 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
路 Playback video while highlighting pixels that are outside of
broadcast range in red.
ffplay example.mov -vf signalstats="out=brng:color=red"
路 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}
signature
Calculates the MPEG-7 Video Signature. The filter can handle more than
one input. In this case the matching between the inputs can be
calculated additionally. The filter always passes through the first
input. The signature of each stream can be written into a file.
It accepts the following options:
detectmode
Enable or disable the matching process.
Available values are:
off Disable the calculation of a matching (default).
full
Calculate the matching for the whole video and output whether
the whole video matches or only parts.
fast
Calculate only until a matching is found or the video ends.
Should be faster in some cases.
nb_inputs
Set the number of inputs. The option value must be a non negative
integer. Default value is 1.
filename
Set the path to which the output is written. If there is more than
one input, the path must be a prototype, i.e. must contain %d or
%0nd (where n is a positive integer), that will be replaced with
the input number. If no filename is specified, no output will be
written. This is the default.
format
Choose the output format.
Available values are:
binary
Use the specified binary representation (default).
xml Use the specified xml representation.
th_d
Set threshold to detect one word as similar. The option value must
be an integer greater than zero. The default value is 9000.
th_dc
Set threshold to detect all words as similar. The option value must
be an integer greater than zero. The default value is 60000.
th_xh
Set threshold to detect frames as similar. The option value must be
an integer greater than zero. The default value is 116.
th_di
Set the minimum length of a sequence in frames to recognize it as
matching sequence. The option value must be a non negative integer
value. The default value is 0.
th_it
Set the minimum relation, that matching frames to all frames must
have. The option value must be a double value between 0 and 1. The
default value is 0.5.
Examples
路 To calculate the signature of an input video and store it in
signature.bin:
ffmpeg -i input.mkv -vf signature=filename=signature.bin -map 0:v -f null -
路 To detect whether two videos match and store the signatures in XML
format in signature0.xml and signature1.xml:
ffmpeg -i input1.mkv -i input2.mkv -filter_complex "[0:v][1:v] signature=nb_inputs=2:detectmode=full:format=xml:filename=signature%d.xml" -map :v -f null -
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
luma_radius.
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 luma_strength.
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
luma_threshold.
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.
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 SSIM.
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. When filename equals "-" the data is sent to
standard output.
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.
This filter also supports the framesync options.
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)
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)
icl interleaved columns, left eye first
icr interleaved columns, right eye first
Default value is sbsl.
out Set stereoscopic image format of output.
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)
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)
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)
ml mono output (left eye only)
mr mono output (right eye only)
chl checkerboard, left eye first
chr checkerboard, right eye first
icl interleaved columns, left eye first
icr interleaved columns, right eye first
hdmi
HDMI frame pack
Default value is arcd.
Examples
路 Convert input video from side by side parallel to anaglyph
yellow/blue dubois:
stereo3d=sbsl:aybd
路 Convert input video from above below (left eye above, right eye
below) to side by side crosseye.
stereo3d=abl:sbsr
streamselect, astreamselect
Select video or audio streams.
The filter accepts the following options:
inputs
Set number of inputs. Default is 2.
map Set input indexes to remap to outputs.
Commands
The "streamselect" and "astreamselect" filter supports the following
commands:
map Set input indexes to remap to outputs.
Examples
路 Select first 5 seconds 1st stream and rest of time 2nd stream:
sendcmd='5.0 streamselect map 1',streamselect=inputs=2:map=0
路 Same as above, but for audio:
asendcmd='5.0 astreamselect map 1',astreamselect=inputs=2:map=0
sobel
Apply sobel operator to input video stream.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be
copied. By default value 0xf, all planes will be processed.
scale
Set value which will be multiplied with filtered result.
delta
Set value which will be added to filtered result.
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.
alpha
Process alpha channel, by default alpha channel is untouched.
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.
swaprect
Swap two rectangular objects in video.
This filter accepts the following options:
w Set object width.
h Set object height.
x1 Set 1st rect x coordinate.
y1 Set 1st rect y coordinate.
x2 Set 2nd rect x coordinate.
y2 Set 2nd rect y coordinate.
All expressions are evaluated once for each frame.
The all options are expressions containing the following constants:
w
h The input width and height.
a same as w / h
sar input sample aspect ratio
dar input display aspect ratio, it is the same as (w / h) * sar
n The number of the input frame, starting from 0.
t The timestamp expressed in seconds. It's NAN if the input timestamp
is unknown.
pos the position in the file of the input frame, NAN if unknown
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
threshold
Apply threshold effect to video stream.
This filter needs four video streams to perform thresholding. First
stream is stream we are filtering. Second stream is holding threshold
values, third stream is holding min values, and last, fourth stream is
holding max values.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be
copied. By default value 0xf, all planes will be processed.
For example if first stream pixel's component value is less then
threshold value of pixel component from 2nd threshold stream, third
stream value will picked, otherwise fourth stream pixel component value
will be picked.
Using color source filter one can perform various types of
thresholding:
Examples
路 Binary threshold, using gray color as threshold:
ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=black -f lavfi -i color=white -lavfi threshold output.avi
路 Inverted binary threshold, using gray color as threshold:
ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -f lavfi -i color=black -lavfi threshold output.avi
路 Truncate binary threshold, using gray color as threshold:
ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=gray -lavfi threshold output.avi
路 Threshold to zero, using gray color as threshold:
ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -i 320x240.avi -lavfi threshold output.avi
路 Inverted threshold to zero, using gray color as threshold:
ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=white -lavfi threshold output.avi
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
路 Extract one picture each 50 frames:
thumbnail=50
路 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
路 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.
路 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_even, 1
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
drop_odd, 2
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
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
mergex2, 7
Move odd frames into the upper field, even into the lower
field, generating a double height frame at same 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 33333 55555
22222 22222 44444 44444
11111 33333 33333 55555
22222 22222 44444 44444
11111 33333 33333 55555
22222 22222 44444 44444
11111 33333 33333 55555
22222 22222 44444 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 linear 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.
complex_filter, cvlfp
Enable complex vertical low-pass filtering. This will slightly
less reduce interlace 'twitter' and Moire patterning but better
retain detail and subjective sharpness impression.
Vertical low-pass filtering can only be enabled for mode
interleave_top and interleave_bottom.
tonemap
Tone map colors from different dynamic ranges.
This filter expects data in single precision floating point, as it
needs to operate on (and can output) out-of-range values. Another
filter, such as zscale, is needed to convert the resulting frame to a
usable format.
The tonemapping algorithms implemented only work on linear light, so
input data should be linearized beforehand (and possibly correctly
tagged).
ffmpeg -i INPUT -vf zscale=transfer=linear,tonemap=clip,zscale=transfer=bt709,format=yuv420p OUTPUT
Options
The filter accepts the following options.
tonemap
Set the tone map algorithm to use.
Possible values are:
none
Do not apply any tone map, only desaturate overbright pixels.
clip
Hard-clip any out-of-range values. Use it for perfect color
accuracy for in-range values, while distorting out-of-range
values.
linear
Stretch the entire reference gamut to a linear multiple of the
display.
gamma
Fit a logarithmic transfer between the tone curves.
reinhard
Preserve overall image brightness with a simple curve, using
nonlinear contrast, which results in flattening details and
degrading color accuracy.
hable
Preserve both dark and bright details better than reinhard, at
the cost of slightly darkening everything. Use it when detail
preservation is more important than color and brightness
accuracy.
mobius
Smoothly map out-of-range values, while retaining contrast and
colors for in-range material as much as possible. Use it when
color accuracy is more important than detail preservation.
Default is none.
param
Tune the tone mapping algorithm.
This affects the following algorithms:
none
Ignored.
linear
Specifies the scale factor to use while stretching. Default to
1.0.
gamma
Specifies the exponent of the function. Default to 1.8.
clip
Specify an extra linear coefficient to multiply into the signal
before clipping. Default to 1.0.
reinhard
Specify the local contrast coefficient at the display peak.
Default to 0.5, which means that in-gamut values will be about
half as bright as when clipping.
hable
Ignored.
mobius
Specify the transition point from linear to mobius transform.
Every value below this point is guaranteed to be mapped 1:1.
The higher the value, the more accurate the result will be, at
the cost of losing bright details. Default to 0.3, which due
to the steep initial slope still preserves in-range colors
fairly accurately.
desat
Apply desaturation for highlights that exceed this level of
brightness. The higher the parameter, the more color information
will be preserved. This setting helps prevent unnaturally blown-out
colors for super-highlights, by (smoothly) turning into white
instead. This makes images feel more natural, at the cost of
reducing information about out-of-range colors.
The default of 2.0 is somewhat conservative and will mostly just
apply to skies or directly sunlit surfaces. A setting of 0.0
disables this option.
This option works only if the input frame has a supported color
tag.
peak
Override signal/nominal/reference peak with this value. Useful when
the embedded peak information in display metadata is not reliable
or when tone mapping from a lower range to a higher range.
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:
路 Drop everything except the second minute of input:
ffmpeg -i INPUT -vf trim=60:120
路 Keep only the first second:
ffmpeg -i INPUT -vf trim=duration=1
unpremultiply
Apply alpha unpremultiply effect to input video stream using first
plane of second stream as alpha.
Both streams must have same dimensions and same pixel format.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be
copied. By default value 0xf, all planes will be processed.
If the format has 1 or 2 components, then luma is bit 0. If the
format has 3 or 4 components: for RGB formats bit 0 is green, bit 1
is blue and bit 2 is red; for YUV formats bit 0 is luma, bit 1 is
chroma-U and bit 2 is chroma-V. If present, the alpha channel is
always the last bit.
inplace
Do not require 2nd input for processing, instead use alpha plane
from input stream.
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 23. The default value is 5.
luma_msize_y, ly
Set the luma matrix vertical size. It must be an odd integer
between 3 and 23. 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 23. The default value is 5.
chroma_msize_y, cy
Set the chroma matrix vertical size. It must be an odd integer
between 3 and 23. 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
路 Apply strong luma sharpen effect:
unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5
路 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).
vaguedenoiser
Apply a wavelet based denoiser.
It transforms each frame from the video input into the wavelet domain,
using Cohen-Daubechies-Feauveau 9/7. Then it applies some filtering to
the obtained coefficients. It does an inverse wavelet transform after.
Due to wavelet properties, it should give a nice smoothed result, and
reduced noise, without blurring picture features.
This filter accepts the following options:
threshold
The filtering strength. The higher, the more filtered the video
will be. Hard thresholding can use a higher threshold than soft
thresholding before the video looks overfiltered. Default value is
2.
method
The filtering method the filter will use.
It accepts the following values:
hard
All values under the threshold will be zeroed.
soft
All values under the threshold will be zeroed. All values above
will be reduced by the threshold.
garrote
Scales or nullifies coefficients - intermediary between (more)
soft and (less) hard thresholding.
Default is garrote.
nsteps
Number of times, the wavelet will decompose the picture. Picture
can't be decomposed beyond a particular point (typically, 8 for a
640x480 frame - as 2^9 = 512 > 480). Valid values are integers
between 1 and 32. Default value is 6.
percent
Partial of full denoising (limited coefficients shrinking), from 0
to 100. Default value is 85.
planes
A list of the planes to process. By default all planes are
processed.
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". The 3rd
color component is static.
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.
color5
Gray values are displayed on graph. Similar to "color" but with
3rd color component picked from radial gradient.
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, color3 and color5 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.
graticule, g
Set what kind of graticule to draw.
none
green
color
opacity, o
Set graticule opacity.
flags, f
Set graticule flags.
white
Draw graticule for white point.
black
Draw graticule for black point.
name
Draw color points short names.
bgopacity, b
Set background opacity.
lthreshold, l
Set low threshold for color component not represented on X or Y
axis. Values lower than this value will be ignored. Default is 0.
Note this value is multiplied with actual max possible value one
pixel component can have. So for 8-bit input and low threshold
value of 0.1 actual threshold is 0.1 * 255 = 25.
hthreshold, h
Set high threshold for color component not represented on X or Y
axis. Values higher than this value will be ignored. Default is 1.
Note this value is multiplied with actual max possible value one
pixel component can have. So for 8-bit input and high threshold
value of 0.9 actual threshold is 0.9 * 255 = 230.
colorspace, c
Set what kind of colorspace to use when drawing graticule.
auto
601
709
Default is auto.
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
路 Use default values:
vidstabdetect
路 Analyze strongly shaky movie and put the results in file
mytransforms.trf:
vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"
路 Visualize the result of internal transformations in the resulting
video:
vidstabdetect=show=1
路 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
路 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.
路 Zoom in a bit more and load transform data from a given file:
vidstabtransform=zoom=5:input="mytransforms.trf"
路 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
路 Apply simple strong vignetting effect:
vignette=PI/4
路 Make a flickering vignetting:
vignette='PI/4+random(1)*PI/50':eval=frame
vmafmotion
Obtain the average vmaf motion score of a video. It is one of the
component filters of VMAF.
The obtained average motion score is printed through the logging
system.
In the below example the input file ref.mpg is being processed and
score is computed.
ffmpeg -i ref.mpg -lavfi vmafmotion -f null -
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:
inputs
Set number of input streams. Default is 2.
shortest
If set to 1, force the output to terminate when the shortest input
terminates. Default value is 0.
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.
stack
Display separate graph for the color components side by side in
"row" mode or one below the other in "column" mode.
parade
Display separate graph for the color components side by side in
"column" mode or one below the other in "row" 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 "stack".
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.
color
Displays actual color value on waveform.
acolor
Similar as above, but with luma showing frequency of chroma
values.
graticule, g
Set which graticule to display.
none
Do not display graticule.
green
Display green graticule showing legal broadcast ranges.
opacity, o
Set graticule opacity.
flags, fl
Set graticule flags.
numbers
Draw numbers above lines. By default enabled.
dots
Draw dots instead of lines.
scale, s
Set scale used for displaying graticule.
digital
millivolts
ire
Default is digital.
bgopacity, b
Set background opacity.
weave, doubleweave
The "weave" takes a field-based video input and join each two
sequential fields into single frame, producing a new double height clip
with half the frame rate and half the frame count.
The "doubleweave" works same as "weave" but without halving frame rate
and frame count.
It accepts the following option:
first_field
Set first field. Available values are:
top, t
Set the frame as top-field-first.
bottom, b
Set the frame as bottom-field-first.
Examples
路 Interlace video using select and separatefields filter:
separatefields,select=eq(mod(n,4),0)+eq(mod(n,4),3),weave
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'.
fps Set the output frame rate, default is '25'.
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
路 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
路 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)'
路 Same as above but without pausing:
zoompan=z='min(max(zoom,pzoom)+0.0015,1.5)':d=1:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'
zscale
Scale (resize) the input video, using the z.lib library:
https://github.com/sekrit-twc/zimg.
The zscale filter forces the output display aspect ratio to be the same
as 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 zscale filter will convert the input to the requested
format.
Options
The filter accepts the following options.
width, w
height, h
Set the output video dimension expression. Default value is the
input dimension.
If the width or w value is 0, the input width is used for the
output. If the height or h value is 0, the input height is used for
the output.
If one and only one of the values is -n with n >= 1, the zscale
filter will 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.
If both values are -n with n >= 1, the behavior will be identical
to both values being set to 0 as previously detailed.
See below for the list of accepted constants for use in the
dimension expression.
size, s
Set the video size. For the syntax of this option, check the "Video
size" section in the ffmpeg-utils manual.
dither, d
Set the dither type.
Possible values are:
none
ordered
random
error_diffusion
Default is none.
filter, f
Set the resize filter type.
Possible values are:
point
bilinear
bicubic
spline16
spline36
lanczos
Default is bilinear.
range, r
Set the color range.
Possible values are:
input
limited
full
Default is same as input.
primaries, p
Set the color primaries.
Possible values are:
input
709
unspecified
170m
240m
2020
Default is same as input.
transfer, t
Set the transfer characteristics.
Possible values are:
input
709
unspecified
601
linear
2020_10
2020_12
smpte2084
iec61966-2-1
arib-std-b67
Default is same as input.
matrix, m
Set the colorspace matrix.
Possible value are:
input
709
unspecified
470bg
170m
2020_ncl
2020_cl
Default is same as input.
rangein, rin
Set the input color range.
Possible values are:
input
limited
full
Default is same as input.
primariesin, pin
Set the input color primaries.
Possible values are:
input
709
unspecified
170m
240m
2020
Default is same as input.
transferin, tin
Set the input transfer characteristics.
Possible values are:
input
709
unspecified
601
linear
2020_10
2020_12
Default is same as input.
matrixin, min
Set the input colorspace matrix.
Possible value are:
input
709
unspecified
470bg
170m
2020_ncl
2020_cl
chromal, c
Set the output chroma location.
Possible values are:
input
left
center
topleft
top
bottomleft
bottom
chromalin, cin
Set the input chroma location.
Possible values are:
input
left
center
topleft
top
bottomleft
bottom
npl Set the nominal peak luminance.
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.
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.
hw_frames_ctx
When using a hardware pixel format, this should be a reference to
an AVHWFramesContext describing input frames.
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
路 Read the initial state from pattern, and specify an output of size
200x400.
cellauto=f=pattern:s=200x400
路 Generate a random initial row with a width of 200 cells, with a
fill ratio of 2/3:
cellauto=ratio=2/3:s=200x200
路 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
路 Specify a more elaborated initial pattern:
cellauto=p='@@ @ @@':s=100x400:full=0:rule=18
coreimagesrc
Video source generated on GPU using Apple's CoreImage API on OSX.
This video source is a specialized version of the coreimage video
filter. Use a core image generator at the beginning of the applied
filterchain to generate the content.
The coreimagesrc video source accepts the following options:
list_generators
List all available generators along with all their respective
options as well as possible minimum and maximum values along with
the default values.
list_generators=true
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".
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".
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.
Additionally, all options of the coreimage video filter are accepted.
A complete filterchain can be used for further processing of the
generated input without CPU-HOST transfer. See coreimage documentation
and examples for details.
Examples
路 Use CIQRCodeGenerator to create a QR code for the FFmpeg homepage,
given as complete and escaped command-line for Apple's standard
bash shell:
ffmpeg -f lavfi -i coreimagesrc=s=100x100:filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png
This example is equivalent to the QRCode example of coreimage
without the need for a nullsrc video source.
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
路 Read a grid from pattern, and center it on a grid of size 300x300
pixels:
life=f=pattern:s=300x300
路 Generate a random grid of size 200x200, with a fill ratio of 2/3:
life=ratio=2/3:s=200x200
路 Specify a custom rule for evolving a randomly generated grid:
life=rule=S14/B34
路 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, testsrc2, yuvtestsrc
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 "testsrc2" source is similar to testsrc, but supports more pixel
formats instead of just "rgb24". This allows using it as an input for
other tests without requiring a format conversion.
The "yuvtestsrc" source generates an YUV test pattern. You should see a
y, cb and cr stripe from top to bottom.
The sources accept the following parameters:
alpha
Specify the alpha (opacity) of the background, only available in
the "testsrc2" source. The value must be between 0 (fully
transparent) and 255 (fully opaque, the default).
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.
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".
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.
abitscope
Convert input audio to a video output, displaying the audio bit scope.
The filter accepts the following options:
rate, r
Set frame rate, expressed as number of frames per second. Default
value is "25".
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 "1024x256".
colors
Specify list of colors separated by space or by '|' which will be
used to draw channels. Unrecognized or missing colors will be
replaced by white color.
ahistogram
Convert input audio to a video output, displaying the volume histogram.
The filter accepts the following options:
dmode
Specify how histogram is calculated.
It accepts the following values:
single
Use single histogram for all channels.
separate
Use separate histogram for each channel.
Default is "single".
rate, r
Set frame rate, expressed as number of frames per second. Default
value is "25".
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 "hd720".
scale
Set display scale.
It accepts the following values:
log logarithmic
sqrt
square root
cbrt
cubic root
lin linear
rlog
reverse logarithmic
Default is "log".
ascale
Set amplitude scale.
It accepts the following values:
log logarithmic
lin linear
Default is "log".
acount
Set how much frames to accumulate in histogram. Defauls is 1.
Setting this to -1 accumulates all frames.
rheight
Set histogram ratio of window height.
slide
Set sonogram sliding.
It accepts the following values:
replace
replace old rows with new ones.
scroll
scroll from top to bottom.
Default is "replace".
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.
video
Enable video output. Default is enabled.
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 "[0,
10]". Values lower than 1 will auto adjust zoom factor to maximal
possible value.
draw
Set the vectorscope drawing mode.
Available values are:
dot Draw dot for each sample.
line
Draw line between previous and current sample.
Default value is dot.
scale
Specify amplitude scale of audio samples.
Available values are:
lin Linear.
sqrt
Square root.
cbrt
Cubic root.
log Logarithmic.
Examples
路 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]'
bench, abench
Benchmark part of a filtergraph.
The filter accepts the following options:
action
Start or stop a timer.
Available values are:
start
Get the current time, set it as frame metadata (using the key
"lavfi.bench.start_time"), and forward the frame to the next
filter.
stop
Get the current time and fetch the "lavfi.bench.start_time"
metadata from the input frame metadata to get the time
difference. Time difference, average, maximum and minimum time
(respectively "t", "avg", "max" and "min") are then printed.
The timestamps are expressed in seconds.
Examples
路 Benchmark selectivecolor filter:
bench=start,selectivecolor=reds=-.2 .12 -.49,bench=stop
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
路 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
路 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.
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.
picture
Draw single picture.
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
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".
dualmono
Treat mono input files as "dual mono". If a mono file is intended
for playback on a stereo system, its EBU R128 measurement will be
perceptually incorrect. If set to "true", this option will
compensate for this effect. Multi-channel input files are not
affected by this option.
panlaw
Set a specific pan law to be used for the measurement of dual mono
files. This parameter is optional, and has a default value of
-3.01dB.
Examples
路 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]"
路 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 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 drops 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 sends 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
路 Interleave frames belonging to different streams using ffmpeg:
ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi
路 Add flickering blur effect:
select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave
metadata, ametadata
Manipulate frame metadata.
This filter accepts the following options:
mode
Set mode of operation of the filter.
Can be one of the following:
select
If both "value" and "key" is set, select frames which have such
metadata. If only "key" is set, select every frame that has
such key in metadata.
add Add new metadata "key" and "value". If key is already available
do nothing.
modify
Modify value of already present key.
delete
If "value" is set, delete only keys that have such value.
Otherwise, delete key. If "key" is not set, delete all metadata
values in the frame.
print
Print key and its value if metadata was found. If "key" is not
set print all metadata values available in frame.
key Set key used with all modes. Must be set for all modes except
"print" and "delete".
value
Set metadata value which will be used. This option is mandatory for
"modify" and "add" mode.
function
Which function to use when comparing metadata value and "value".
Can be one of following:
same_str
Values are interpreted as strings, returns true if metadata
value is same as "value".
starts_with
Values are interpreted as strings, returns true if metadata
value starts with the "value" option string.
less
Values are interpreted as floats, returns true if metadata
value is less than "value".
equal
Values are interpreted as floats, returns true if "value" is
equal with metadata value.
greater
Values are interpreted as floats, returns true if metadata
value is greater than "value".
expr
Values are interpreted as floats, returns true if expression
from option "expr" evaluates to true.
expr
Set expression which is used when "function" is set to "expr". The
expression is evaluated through the eval API and can contain the
following constants:
VALUE1
Float representation of "value" from metadata key.
VALUE2
Float representation of "value" as supplied by user in "value"
option.
file
If specified in "print" mode, output is written to the named file.
Instead of plain filename any writable url can be specified.
Filename ``-'' is a shorthand for standard output. If "file" option
is not set, output is written to the log with AV_LOG_INFO loglevel.
Examples
路 Print all metadata values for frames with key
"lavfi.singnalstats.YDIF" with values between 0 and 1.
signalstats,metadata=print:key=lavfi.signalstats.YDIF:value=0:function=expr:expr='between(VALUE1,0,1)'
路 Print silencedetect output to file metadata.txt.
silencedetect,ametadata=mode=print:file=metadata.txt
路 Direct all metadata to a pipe with file descriptor 4.
metadata=mode=print:file='pipe\:4'
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.
realtime, arealtime
Slow down filtering to match real time approximately.
These filters will pause the filtering for a variable amount of time to
match the output rate with the input timestamps. They are similar to
the re option to "ffmpeg".
They accept the following options:
limit
Time limit for the pauses. Any pause longer than that will be
considered a timestamp discontinuity and reset the timer. Default
is 2 seconds.
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)
concatdec_select
The concat demuxer can select only part of a concat input file by
setting an inpoint and an outpoint, but the output packets may not
be entirely contained in the selected interval. By using this
variable, it is possible to skip frames generated by the concat
demuxer which are not exactly contained in the selected interval.
This works by comparing the frame pts against the
lavf.concat.start_time and the lavf.concat.duration packet metadata
values which are also present in the decoded frames.
The concatdec_select variable is -1 if the frame pts is at least
start_time and either the duration metadata is missing or the frame
pts is less than start_time + duration, 0 otherwise, and NaN if the
start_time metadata is missing.
That basically means that an input frame is selected if its pts is
within the interval set by the concat demuxer.
The default value of the select expression is "1".
Examples
路 Select all frames in input:
select
The example above is the same as:
select=1
路 Skip all frames:
select=0
路 Select only I-frames:
select='eq(pict_type\,I)'
路 Select one frame every 100:
select='not(mod(n\,100))'
路 Select only frames contained in the 10-20 time interval:
select=between(t\,10\,20)
路 Select only I-frames contained in the 10-20 time interval:
select=between(t\,10\,20)*eq(pict_type\,I)
路 Select frames with a minimum distance of 10 seconds:
select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)'
路 Use aselect to select only audio frames with samples number > 100:
aselect='gt(samples_n\,100)'
路 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.
路 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
路 Select useful frames from an ffconcat file which is using inpoints
and outpoints but where the source files are not intra frame only.
ffmpeg -copyts -vsync 0 -segment_time_metadata 1 -i input.ffconcat -vf select=concatdec_select -af aselect=concatdec_select output.avi
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
路 Specify audio tempo change at second 4:
asendcmd=c='4.0 atempo tempo 1.5',atempo
路 Target a specific filter instance:
asendcmd=c='4.0 atempo@my tempo 1.5',atempo@my
路 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
路 Start counting PTS from zero
setpts=PTS-STARTPTS
路 Apply fast motion effect:
setpts=0.5*PTS
路 Apply slow motion effect:
setpts=2.0*PTS
路 Set fixed rate of 25 frames per second:
setpts=N/(25*TB)
路 Set fixed rate 25 fps with some jitter:
setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'
路 Apply an offset of 10 seconds to the input PTS:
setpts=PTS+10/TB
路 Generate timestamps from a "live source" and rebase onto the
current timebase:
setpts='(RTCTIME - RTCSTART) / (TB * 1000000)'
路 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
路 Set the timebase to 1/25:
settb=expr=1/25
路 Set the timebase to 1/10:
settb=expr=0.1
路 Set the timebase to 1001/1000:
settb=1+0.001
路 Set the timebase to 2*intb:
settb=2*intb
路 Set the default timebase value:
settb=AVTB
showcqt
Convert input audio to a video output representing frequency spectrum
logarithmically using Brown-Puckette constant Q transform algorithm
with direct frequency domain coefficient calculation (but the transform
itself is not really constant Q, instead the Q factor is actually
variable/clamped), with musical tone scale, from E0 to D#10.
The filter accepts the following options:
size, s
Specify the video size for the output. It must be even. For the
syntax of this option, check the "Video size" section in the
ffmpeg-utils manual. Default value is "1920x1080".
fps, rate, r
Set the output frame rate. Default value is 25.
bar_h
Set the bargraph height. It must be even. Default value is "-1"
which computes the bargraph height automatically.
axis_h
Set the axis height. It must be even. Default value is "-1" which
computes the axis height automatically.
sono_h
Set the sonogram height. It must be even. Default value is "-1"
which computes the sonogram height automatically.
fullhd
Set the fullhd resolution. This option is deprecated, use size, s
instead. Default value is 1.
sono_v, volume
Specify the sonogram volume expression. It can contain variables:
bar_v
the bar_v evaluated expression
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the 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.
bar_v, volume2
Specify the bargraph volume expression. It can contain variables:
sono_v
the sono_v evaluated expression
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the 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 "sono_v".
sono_g, gamma
Specify the sonogram gamma. Lower gamma makes the spectrum more
contrast, higher gamma makes the spectrum having more range.
Default value is 3. Acceptable range is "[1, 7]".
bar_g, gamma2
Specify the bargraph gamma. Default value is 1. Acceptable range is
"[1, 7]".
bar_t
Specify the bargraph transparency level. Lower value makes the
bargraph sharper. Default value is 1. Acceptable range is "[0,
1]".
timeclamp, tc
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 range is "[0.002, 1]". Default value is 0.17.
attack
Set attack time in seconds. The default is 0 (disabled). Otherwise,
it limits future samples by applying asymmetric windowing in time
domain, useful when low latency is required. Accepted range is "[0,
1]".
basefreq
Specify the transform base frequency. Default value is
20.01523126408007475, which is frequency 50 cents below E0.
Acceptable range is "[10, 100000]".
endfreq
Specify the transform end frequency. Default value is
20495.59681441799654, which is frequency 50 cents above D#10.
Acceptable range is "[10, 100000]".
coeffclamp
This option is deprecated and ignored.
tlength
Specify the transform length in time domain. Use this option to
control accuracy trade-off between time domain and frequency domain
at every frequency sample. It can contain variables:
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the value of timeclamp option.
Default value is "384*tc/(384+tc*f)".
count
Specify the transform count for every video frame. Default value is
6. Acceptable range is "[1, 30]".
fcount
Specify the transform count for every single pixel. Default value
is 0, which makes it computed automatically. Acceptable range is
"[0, 10]".
fontfile
Specify font file for use with freetype to draw the axis. If not
specified, use embedded font. Note that drawing with font file or
embedded font is not implemented with custom basefreq and endfreq,
use axisfile option instead.
font
Specify fontconfig pattern. This has lower priority than fontfile.
The : in the pattern may be replaced by | to avoid unnecessary
escaping.
fontcolor
Specify font color expression. This is arithmetic expression that
should return integer value 0xRRGGBB. It can contain variables:
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the 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))".
axisfile
Specify image file to draw the axis. This option override fontfile
and fontcolor option.
axis, text
Enable/disable drawing text to the axis. If it is set to 0, drawing
to the axis is disabled, ignoring fontfile and axisfile option.
Default value is 1.
csp Set colorspace. The accepted values are:
unspecified
Unspecified (default)
bt709
BT.709
fcc FCC
bt470bg
BT.470BG or BT.601-6 625
smpte170m
SMPTE-170M or BT.601-6 525
smpte240m
SMPTE-240M
bt2020ncl
BT.2020 with non-constant luminance
cscheme
Set spectrogram color scheme. This is list of floating point values
with format "left_r|left_g|left_b|right_r|right_g|right_b". The
default is "1|0.5|0|0|0.5|1".
Examples
路 Playing audio while showing the spectrum:
ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]'
路 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]'
路 Playing at 1280x720:
ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=s=1280x720:count=4 [out0]'
路 Disable sonogram display:
sono_h=0
路 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]'
路 Same as above, but with more accuracy in frequency domain:
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]'
路 Custom volume:
bar_v=10:sono_v=bar_v*a_weighting(f)
路 Custom gamma, now spectrum is linear to the amplitude.
bar_g=2:sono_g=2
路 Custom tlength equation:
tc=0.33:tlength='st(0,0.17); 384*tc / (384 / ld(0) + tc*f /(1-ld(0))) + 384*tc / (tc*f / ld(0) + 384 /(1-ld(0)))'
路 Custom fontcolor and fontfile, C-note is colored green, others are
colored blue:
fontcolor='if(mod(floor(midi(f)+0.5),12), 0x0000FF, g(1))':fontfile=myfont.ttf
路 Custom font using fontconfig:
font='Courier New,Monospace,mono|bold'
路 Custom frequency range with custom axis using image file:
axisfile=myaxis.png:basefreq=40:endfreq=10000
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
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
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.
colors
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.
cmode
Set channel display mode.
It accepts the following values:
combined
separate
Default is "combined".
minamp
Set minimum amplitude used in "log" amplitude scaler.
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
rscroll
the samples scroll from left to 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 displayed using the same color scheme
rainbow
each channel is displayed using the rainbow color scheme
moreland
each channel is displayed using the moreland color scheme
nebulae
each channel is displayed using the nebulae color scheme
fire
each channel is displayed using the fire color scheme
fiery
each channel is displayed using the fiery color scheme
fruit
each channel is displayed using the fruit color scheme
cool
each channel is displayed using the cool 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
4thrt
4th root
5thrt
5th root
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:
rect
bartlett
hann
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
Default value is "hann".
orientation
Set orientation of time vs frequency axis. Can be "vertical" or
"horizontal". Default is "vertical".
overlap
Set ratio of overlap window. Default value is 0. When value is 1
overlap is set to recommended size for specific window function
currently used.
gain
Set scale gain for calculating intensity color values. Default
value is 1.
data
Set which data to display. Can be "magnitude", default or "phase".
rotation
Set color rotation, must be in [-1.0, 1.0] range. Default value is
0.
The usage is very similar to the showwaves filter; see the examples in
that section.
Examples
路 Large window with logarithmic color scaling:
showspectrum=s=1280x480:scale=log
路 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]'
showspectrumpic
Convert input audio to a single video frame, 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 "4096x2048".
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 displayed using the same color scheme
rainbow
each channel is displayed using the rainbow color scheme
moreland
each channel is displayed using the moreland color scheme
nebulae
each channel is displayed using the nebulae color scheme
fire
each channel is displayed using the fire color scheme
fiery
each channel is displayed using the fiery color scheme
fruit
each channel is displayed using the fruit color scheme
cool
each channel is displayed using the cool color scheme
Default value is intensity.
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
4thrt
4th root
5thrt
5th root
Default value is log.
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:
rect
bartlett
hann
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
Default value is "hann".
orientation
Set orientation of time vs frequency axis. Can be "vertical" or
"horizontal". Default is "vertical".
gain
Set scale gain for calculating intensity color values. Default
value is 1.
legend
Draw time and frequency axes and legends. Default is enabled.
rotation
Set color rotation, must be in [-1.0, 1.0] range. Default value is
0.
Examples
路 Extract an audio spectrogram of a whole audio track in a 1024x1024
picture using ffmpeg:
ffmpeg -i audio.flac -lavfi showspectrumpic=s=1024x1024 spectrogram.png
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 [80, 8192]. Default is 400.
h Set channel height, allowed range is [1, 900]. Default is 20.
f Set fade, allowed range is [0.001, 1]. Default is 0.95.
c Set volume color expression.
The expression can use the following variables:
VOLUME
Current max volume of channel in dB.
PEAK
Current peak.
CHANNEL
Current channel number, starting from 0.
t If set, displays channel names. Default is enabled.
v If set, displays volume values. Default is enabled.
o Set orientation, can be "horizontal" or "vertical", default is
"horizontal".
s Set step size, allowed range s [0, 5]. Default is 0, which means
step is disabled.
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.
colors
Set colors separated by '|' which are going to be used for drawing
of each channel.
scale
Set amplitude scale.
Available values are:
lin Linear.
log Logarithmic.
sqrt
Square root.
cbrt
Cubic root.
Default is linear.
Examples
路 Output the input file audio and the corresponding video
representation at the same time:
amovie=a.mp3,asplit[out0],showwaves[out1]
路 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.
colors
Set colors separated by '|' which are going to be used for drawing
of each channel.
scale
Set amplitude scale.
Available values are:
lin Linear.
log Logarithmic.
sqrt
Square root.
cbrt
Cubic root.
Default is linear.
Examples
路 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
sidedata, asidedata
Delete frame side data, or select frames based on it.
This filter accepts the following options:
mode
Set mode of operation of the filter.
Can be one of the following:
select
Select every frame with side data of "type".
delete
Delete side data of "type". If "type" is not set, delete all
side data in the frame.
type
Set side data type used with all modes. Must be set for "select"
mode. For the list of frame side data types, refer to the
"AVFrameSideDataType" enum in libavutil/frame.h. For example, to
choose "AV_FRAME_DATA_PANSCAN" side data, you must specify
"PANSCAN".
spectrumsynth
Sythesize audio from 2 input video spectrums, first input stream
represents magnitude across time and second represents phase across
time. The filter will transform from frequency domain as displayed in
videos back to time domain as presented in audio output.
This filter is primarily created for reversing processed showspectrum
filter outputs, but can synthesize sound from other spectrograms too.
But in such case results are going to be poor if the phase data is not
available, because in such cases phase data need to be recreated,
usually its just recreated from random noise. For best results use
gray only output ("channel" color mode in showspectrum filter) and
"log" scale for magnitude video and "lin" scale for phase video. To
produce phase, for 2nd video, use "data" option. Inputs videos should
generally use "fullframe" slide mode as that saves resources needed for
decoding video.
The filter accepts the following options:
sample_rate
Specify sample rate of output audio, the sample rate of audio from
which spectrum was generated may differ.
channels
Set number of channels represented in input video spectrums.
scale
Set scale which was used when generating magnitude input spectrum.
Can be "lin" or "log". Default is "log".
slide
Set slide which was used when generating inputs spectrums. Can be
"replace", "scroll", "fullframe" or "rscroll". Default is
"fullframe".
win_func
Set window function used for resynthesis.
overlap
Set window overlap. In range "[0, 1]". Default is 1, which means
optimal overlap for selected window function will be picked.
orientation
Set orientation of input videos. Can be "vertical" or "horizontal".
Default is "vertical".
Examples
路 First create magnitude and phase videos from audio, assuming audio
is stereo with 44100 sample rate, then resynthesize videos back to
audio with spectrumsynth:
ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=log:overlap=0.875:color=channel:slide=fullframe:data=magnitude -an -c:v rawvideo magnitude.nut
ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=lin:overlap=0.875:color=channel:slide=fullframe:data=phase -an -c:v rawvideo phase.nut
ffmpeg -i magnitude.nut -i phase.nut -lavfi spectrumsynth=channels=2:sample_rate=44100:win_func=hann:overlap=0.875:slide=fullframe output.flac
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
路 Create two separate outputs from the same input:
[in] split [out0][out1]
路 To create 3 or more outputs, you need to specify the number of
outputs, like in:
[in] asplit=3 [out0][out1][out2]
路 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];
路 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 0, the stream will be looped infinitely. 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.
discontinuity
Specifies the time difference between frames above which the point
is considered a timestamp discontinuity which is removed by
adjusting the later 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
路 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]
路 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]
路 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]
Commands
Both movie and amovie support the following commands:
seek
Perform seek using "av_seek_frame". The syntax is: seek
stream_index|timestamp|flags
路 stream_index: If stream_index is -1, a default stream is
selected, and timestamp is automatically converted from
AV_TIME_BASE units to the stream specific time_base.
路 timestamp: Timestamp in AVStream.time_base units or, if no
stream is specified, in AV_TIME_BASE units.
路 flags: Flags which select direction and seeking mode.
get_duration
Get movie duration in AV_TIME_BASE units.
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)