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

NAME
ffmpeg - ffmpeg media converter

SYNOPSIS
ffmpeg [global_options] {[input_file_options] -i input_url} ...
{[output_file_options] output_url} ...

DESCRIPTION
ffmpeg is a universal media converter. It can read a wide variety of
inputs - including live grabbing/recording devices - filter, and
transcode them into a plethora of output formats.

ffmpeg reads from an arbitrary number of inputs (which can be regular
files, pipes, network streams, grabbing devices, etc.), specified by
the "-i" option, and writes to an arbitrary number of outputs, 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 can, in principle, contain any number of
elementary streams of different types
(video/audio/subtitle/attachment/data), though the allowed stream
counts and/or types 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 inputs/outputs in options, you must use their indices
(0-based). E.g. the first input is 0, the second is 1, etc. Similarly,
streams within an input/output are referred to by their indices. E.g.
"2:3" refers to the fourth stream in the third input or output. 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.

Some simple examples follow.

• Convert an input media file to a different format, by re-encoding
media streams:

ffmpeg -i input.avi output.mp4

• Set the video bitrate of the output file to 64 kbit/s:

ffmpeg -i input.avi -b:v 64k -bufsize 64k output.mp4

• Force the frame rate of the output file to 24 fps:

ffmpeg -i input.avi -r 24 output.mp4

• 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.mp4

The format option may be needed for raw input files.

DETAILED DESCRIPTION
ffmpeg builds a transcoding pipeline out of the components listed
below. The program's operation then consists of input data chunks
flowing from the sources down the pipes towards the sinks, while being
transformed by the components they encounter along the way.

The following kinds of components are available:

• Demuxers (short for "demultiplexers") read an input source in order
to extract

• global properties such as metadata or chapters;

• list of input elementary streams and their properties

One demuxer instance is created for each -i option, and sends
encoded packets to decoders or muxers.

In other literature, demuxers are sometimes called splitters,
because their main function is splitting a file into elementary
streams (though some files only contain one elementary stream).

A schematic representation of a demuxer looks like this:

demuxer packets for stream 0
elementary stream 0

global
properties packets for stream 1
and elementary stream 1
metadata

...........

packets for stream N
elementary stream N

read from file, network stream,
grabbing device, etc.

• Decoders receive encoded (compressed) packets for an audio, video,
or subtitle elementary stream, and decode them into raw frames
(arrays of pixels for video, PCM for audio). A decoder is typically
associated with (and receives its input from) an elementary stream
in a demuxer, but sometimes may also exist on its own (see Loopback
decoders).

A schematic representation of a decoder looks like this:

packets raw frames
decoder

• Filtergraphs process and transform raw audio or video frames. A
filtergraph consists of one or more individual filters linked into
a graph. Filtergraphs come in two flavors - simple and complex,
configured with the -filter and -filter_complex options,
respectively.

A simple filtergraph is associated with an output elementary
stream; it receives the input to be filtered from a decoder and
sends filtered output to that output stream's encoder.

A simple video filtergraph that performs deinterlacing (using the
"yadif" deinterlacer) followed by resizing (using the "scale"
filter) can look like this:

simple filtergraph
frames from frames for
a decoder an encoder
yadif scale

A complex filtergraph is standalone and not associated with any
specific stream. It may have multiple (or zero) inputs,
potentially of different types (audio or video), each of which
receiving data either from a decoder or another complex
filtergraph's output. It also has one or more outputs that feed
either an encoder or another complex filtergraph's input.

The following example diagram represents a complex filtergraph with
3 inputs and 2 outputs (all video):

complex filtergraph

frames frames
input 0 overlay overlay output 0

frames
input 1

frames frames
input 2scalesplitoutput 1

Frames from second input are overlaid over those from the first.
Frames from the third input are rescaled, then the duplicated into
two identical streams. One of them is overlaid over the combined
first two inputs, with the result exposed as the filtergraph's
first output. The other duplicate ends up being the filtergraph's
second output.

• Encoders receive raw audio, video, or subtitle frames and encode
them into encoded packets. The encoding (compression) process is
typically lossy - it degrades stream quality to make the output
smaller; some encoders are lossless, but at the cost of much higher
output size. A video or audio encoder receives its input from some
filtergraph's output, subtitle encoders receive input from a
decoder (since subtitle filtering is not supported yet). Every
encoder is associated with some muxer's output elementary stream
and sends its output to that muxer.

A schematic representation of an encoder looks like this:

raw frames packets
encoder

• Muxers (short for "multiplexers") receive encoded packets for their
elementary streams from encoders (the transcoding path) or directly
from demuxers (the streamcopy path), interleave them (when there is
more than one elementary stream), and write the resulting bytes
into the output file (or pipe, network stream, etc.).

A schematic representation of a muxer looks like this:

packets for stream 0 muxer
elementary stream 0

global
packets for stream 1 properties
elementary stream 1 and
metadata

...........

packets for stream N
elementary stream N

write to file, network stream,
grabbing device, etc.

Streamcopy
The simplest pipeline in ffmpeg is single-stream streamcopy, that is
copying one input elementary stream's packets without decoding,
filtering, or encoding them. As an example, consider an input file
called INPUT.mkv with 3 elementary streams, from which we take the
second and write it to file OUTPUT.mp4. A schematic representation of
such a pipeline looks like this:

demuxer unused
elementary stream 0

INPUT.mkv
packets muxer
elementary stream 1 elementary stream 0
OUTPUT.mp4

unused
elementary stream 2

The above pipeline can be constructed with the following commandline:

ffmpeg -i INPUT.mkv -map 0:1 -c copy OUTPUT.mp4

In this commandline

• there is a single input INPUT.mkv;

• there are no input options for this input;

• there is a single output OUTPUT.mp4;

• there are two output options for this output:

• "-map 0:1" selects the input stream to be used - from input
with index 0 (i.e. the first one) the stream with index 1 (i.e.
the second one);

• "-c copy" selects the "copy" encoder, i.e. streamcopy with no
decoding or encoding.

Streamcopy is useful for changing the elementary stream count,
container format, or modifying container-level metadata. 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 a variety of
factors (e.g. certain information required by the target container is
not available in the source). Applying filters is obviously also
impossible, since filters work on decoded frames.

More complex streamcopy scenarios can be constructed - e.g. combining
streams from two input files into a single output:

demuxer 0 packets muxer
elementary stream 0 elementary stream 0
INPUT0.mkv OUTPUT.mp4

demuxer 1 packets elementary stream 1
elementary stream 0
INPUT1.aac

that can be built by the commandline

ffmpeg -i INPUT0.mkv -i INPUT1.aac -map 0:0 -map 1:0 -c copy OUTPUT.mp4

The output -map option is used twice here, creating two streams in the
output file - one fed by the first input and one by the second. The
single instance of the -c option selects streamcopy for both of those
streams. You could also use multiple instances of this option together
with Stream specifiers to apply different values to each stream, as
will be demonstrated in following sections.

A converse scenario is splitting multiple streams from a single input
into multiple outputs:

demuxer packets muxer 0
elementary stream 0 elementary stream 0
OUTPUT0.mp4
INPUT.mkv
packets
elementary stream 1 muxer 1
elementary stream 0
OUTPUT1.mp4

built with

ffmpeg -i INPUT.mkv -map 0:0 -c copy OUTPUT0.mp4 -map 0:1 -c copy OUTPUT1.mp4

Note how a separate instance of the -c option is needed for every
output file even though their values are the same. This is because
non-global options (which is most of them) only apply in the context of
the file before which they are placed.

These examples can of course be further generalized into arbitrary
remappings of any number of inputs into any number of outputs.

Transcoding
Transcoding is the process of decoding a stream and then encoding it
again. Since encoding tends to be computationally expensive and in most
cases degrades the stream quality (i.e. it is lossy), you should only
transcode when you need to and perform streamcopy otherwise. Typical
reasons to transcode are:

• applying filters - e.g. resizing, deinterlacing, or overlaying
video; resampling or mixing audio;

• you want to feed the stream to something that cannot decode the
original codec.

Note that ffmpeg will transcode all audio, video, and subtitle streams
unless you specify -c copy for them.

Consider an example pipeline that reads an input file with one audio
and one video stream, transcodes the video and copies the audio into a
single output file. This can be schematically represented as follows

demuxer audio packets
stream 0 (audio)

INPUT.mkv video raw
packets video video frames
stream 1 (video) decoder

video
muxer packets video
stream 0 (video) encoder
(libx264)
OUTPUT.mp4

stream 1 (audio)

and implemented with the following commandline:

ffmpeg -i INPUT.mkv -map 0:v -map 0:a -c:v libx264 -c:a copy OUTPUT.mp4

Note how it uses stream specifiers ":v" and ":a" to select input
streams and apply different values of the -c option to them; see the
Stream specifiers section for more details.

Filtering
When transcoding, audio and video streams can be filtered before
encoding, with either a simple or complex filtergraph.

Simple filtergraphs

Simple filtergraphs are those that have exactly one input and output,
both of the same type (audio or video). They are configured with the
per-stream -filter option (with -vf and -af aliases for -filter:v
(video) and -filter:a (audio) respectively). Note that simple
filtergraphs are tied to their output stream, so e.g. if you have
multiple audio streams, -af will create a separate filtergraph for each
one.

Taking the transcoding example from above, adding filtering (and
omitting audio, for clarity) makes it look like this:

demuxer
video stream packets video frames
INPUT.mkv decoder

simple filtergraph

yadif scale

video
muxer packets video
video stream encoder
OUTPUT.mp4

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. 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. Each instance of
-filter_complex creates a new complex filtergraph, and there can be any
number of them.

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.

Loopback decoders
While decoders are normally associated with demuxer streams, it is also
possible to create "loopback" decoders that decode the output from some
encoder and allow it to be fed back to complex filtergraphs. This is
done with the "-dec" directive, which takes as a parameter the index of
the output stream that should be decoded. Every such directive creates
a new loopback decoder, indexed with successive integers starting at
zero. These indices should then be used to refer to loopback decoders
in complex filtergraph link labels, as described in the documentation
for -filter_complex.

Decoding AVOptions can be passed to loopback decoders by placing them
before "-dec", analogously to input/output options.

E.g. the following example:

ffmpeg -i INPUT \
-map 0:v:0 -c:v libx264 -crf 45 -f null - \
-threads 3 -dec 0:0 \
-filter_complex '[0:v][dec:0]hstack[stack]' \
-map '[stack]' -c:v ffv1 OUTPUT

reads an input video and

• (line 2) encodes it with "libx264" at low quality;

• (line 3) decodes this encoded stream using 3 threads;

• (line 4) places decoded video side by side with the original input
video;

• (line 5) combined video is then losslessly encoded and written into
OUTPUT.

Such a transcoding pipeline can be represented with the following
diagram:

demuxer
video stream video video null muxer
INPUT decoder encoder (discards its input)
(libx264)

loopback
decoder

complex filtergraph

hstack

muxer video
video stream encoder
OUTPUT (ffv1)

STREAM SELECTION
ffmpeg provides the "-map" option for manual control of stream
selection in each output file. Users can skip "-map" and let ffmpeg
perform automatic stream selection as described below. The "-vn / -an /
-sn / -dn" options can be used to skip inclusion of video, audio,
subtitle and data streams respectively, whether manually mapped or
automatically selected, except for those streams which are outputs of
complex filtergraphs.

Description
The sub-sections that follow describe the various rules that are
involved in stream selection. The examples that follow next show how
these rules are applied in practice.

While every effort is made to accurately reflect the behavior of the
program, FFmpeg is under continuous development and the code may have
changed since the time of this writing.

Automatic stream selection

In the absence of any map options for a particular output file, ffmpeg
inspects the output format to check which type of streams can be
included in it, viz. video, audio and/or subtitles. For each acceptable
stream type, ffmpeg will pick one stream, when available, from among
all the inputs.

It will select that stream 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 found but there's a
caveat. The output format's default subtitle encoder can be either
text-based or image-based, and only a subtitle stream of the same
type will be chosen.

In the case where several streams of the same type rate equally, the
stream with the lowest index is chosen.

Data or attachment streams are not automatically selected and can only
be included using "-map".

Manual stream selection

When "-map" is used, only user-mapped streams are included in that
output file, with one possible exception for filtergraph outputs
described below.

Complex filtergraphs

If there are any complex filtergraph output streams with unlabeled
pads, they will be added to the first output file. This will lead to a
fatal error if the stream type is not supported by the output format.
In the absence of the map option, the inclusion of these streams leads
to the automatic stream selection of their types being skipped. If map
options are present, these filtergraph streams are included in addition
to the mapped streams.

Complex filtergraph output streams with labeled pads must be mapped
once and exactly once.

Stream handling

Stream handling is independent of stream selection, with an exception
for subtitles described below. Stream handling is set via the "-codec"
option addressed to streams within a specific output file. In
particular, codec options are applied by ffmpeg after the stream
selection process and thus do not influence the latter. If no "-codec"
option is specified for a stream type, ffmpeg will select the default
encoder registered by the output file muxer.

An exception exists for subtitles. If a subtitle encoder is specified
for an output file, the first subtitle stream found of any type, text
or image, will be included. ffmpeg does not validate if the specified
encoder can convert the selected stream or if the converted stream is
acceptable within the output format. This applies generally as well:
when the user sets an encoder manually, the stream selection process
cannot check if the encoded stream can be muxed into the output file.
If it cannot, ffmpeg will abort and all output files will fail to be
processed.

Examples
The following examples illustrate the behavior, quirks and limitations
of ffmpeg's stream selection methods.

They assume the following three input files.

input file 'A.avi'
stream 0: video 640x360
stream 1: audio 2 channels

input file 'B.mp4'
stream 0: video 1920x1080
stream 1: audio 2 channels
stream 2: subtitles (text)
stream 3: audio 5.1 channels
stream 4: subtitles (text)

input file 'C.mkv'
stream 0: video 1280x720
stream 1: audio 2 channels
stream 2: subtitles (image)

Example: automatic stream selection

ffmpeg -i A.avi -i B.mp4 out1.mkv out2.wav -map 1:a -c:a copy out3.mov

There are three output files specified, and for the first two, no
"-map" options are set, so ffmpeg will select streams for these two
files automatically.

out1.mkv is a Matroska container file and accepts video, audio and
subtitle streams, so ffmpeg will try to select one of each type.For
video, it will select "stream 0" from B.mp4, which has the highest
resolution among all the input video streams.For audio, it will select
"stream 3" from B.mp4, since it has the greatest number of channels.For
subtitles, it will select "stream 2" from B.mp4, which is the first
subtitle stream from among A.avi and B.mp4.

out2.wav accepts only audio streams, so only "stream 3" from B.mp4 is
selected.

For out3.mov, since a "-map" option is set, no automatic stream
selection will occur. The "-map 1:a" option will select all audio
streams from the second input B.mp4. No other streams will be included
in this output file.

For the first two outputs, all included streams will be transcoded. The
encoders chosen will be the default ones registered by each output
format, which may not match the codec of the selected input streams.

For the third output, codec option for audio streams has been set to
"copy", so no decoding-filtering-encoding operations will occur, or can
occur. Packets of selected streams shall be conveyed from the input
file and muxed within the output file.

Example: automatic subtitles selection

ffmpeg -i C.mkv out1.mkv -c:s dvdsub -an out2.mkv

Although out1.mkv is a Matroska container file which accepts subtitle
streams, only a video and audio stream shall be selected. The subtitle
stream of C.mkv is image-based and the default subtitle encoder of the
Matroska muxer is text-based, so a transcode operation for the
subtitles is expected to fail and hence the stream isn't selected.
However, in out2.mkv, a subtitle encoder is specified in the command
and so, the subtitle stream is selected, in addition to the video
stream. The presence of "-an" disables audio stream selection for
out2.mkv.

Example: unlabeled filtergraph outputs

ffmpeg -i A.avi -i C.mkv -i B.mp4 -filter_complex "overlay" out1.mp4 out2.srt

A filtergraph is setup here using the "-filter_complex" option and
consists of a single video filter. The "overlay" filter requires
exactly two video inputs, but none are specified, so the first two
available video streams are used, those of A.avi and C.mkv. The output
pad of the filter has no label and so is sent to the first output file
out1.mp4. Due to this, automatic selection of the video stream is
skipped, which would have selected the stream in B.mp4. The audio
stream with most channels viz. "stream 3" in B.mp4, is chosen
automatically. No subtitle stream is chosen however, since the MP4
format has no default subtitle encoder registered, and the user hasn't
specified a subtitle encoder.

The 2nd output file, out2.srt, only accepts text-based subtitle
streams. So, even though the first subtitle stream available belongs to
C.mkv, it is image-based and hence skipped. The selected stream,
"stream 2" in B.mp4, is the first text-based subtitle stream.

Example: labeled filtergraph outputs

ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-map '[outv]' -an out1.mp4 \
out2.mkv \
-map '[outv]' -map 1:a:0 out3.mkv

The above command will fail, as the output pad labelled "[outv]" has
been mapped twice. None of the output files shall be processed.

ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-an out1.mp4 \
out2.mkv \
-map 1:a:0 out3.mkv

This command above will also fail as the hue filter output has a label,
"[outv]", and hasn't been mapped anywhere.

The command should be modified as follows,

ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0,split=2[outv1][outv2];overlay;aresample" \
-map '[outv1]' -an out1.mp4 \
out2.mkv \
-map '[outv2]' -map 1:a:0 out3.mkv

The video stream from B.mp4 is sent to the hue filter, whose output is
cloned once using the split filter, and both outputs labelled. Then a
copy each is mapped to the first and third output files.

The overlay filter, requiring two video inputs, uses the first two
unused video streams. Those are the streams from A.avi and C.mkv. The
overlay output isn't labelled, so it is sent to the first output file
out1.mp4, regardless of the presence of the "-map" option.

The aresample filter is sent the first unused audio stream, that of
A.avi. Since this filter output is also unlabelled, it too is mapped to
the first output file. The presence of "-an" only suppresses automatic
or manual stream selection of audio streams, not outputs sent from
filtergraphs. Both these mapped streams shall be ordered before the
mapped stream in out1.mp4.

The video, audio and subtitle streams mapped to "out2.mkv" are entirely
determined by automatic stream selection.

out3.mkv consists of the cloned video output from the hue filter and
the first audio stream from B.mp4.

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.

Options that take arguments support a special syntax where the argument
given on the command line is interpreted as a path to the file from
which the actual argument value is loaded. To use this feature, add a
forward slash '/' immediately before the option name (after the leading
dash). E.g.

ffmpeg -i INPUT -/filter:v filter.script OUTPUT

will load a filtergraph description from the file named filter.script.

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. If stream_index is
used as an additional stream specifier (see below), then it selects
stream number stream_index from the matching streams. Stream
numbering is based on the order of the streams as detected by
libavformat except when a stream group specifier or program ID is
also specified. In this case it is based on the ordering of the
streams in the group or program.

stream_type[:additional_stream_specifier]
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
additional_stream_specifier is used, then it matches streams which
both have this type and match the additional_stream_specifier.
Otherwise, it matches all streams of the specified type.

g:group_specifier[:additional_stream_specifier]
Matches streams which are in the group with the specifier
group_specifier. if additional_stream_specifier is used, then it
matches streams which both are part of the group and match the
additional_stream_specifier. group_specifier may be one of the
following:

group_index
Match the stream with this group index.

#group_id or i:group_id
Match the stream with this group id.

p:program_id[:additional_stream_specifier]
Matches streams which are in the program with the id program_id. If
additional_stream_specifier is used, then it matches streams which
both are part of the program and match the
additional_stream_specifier.

#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. The colon character ':' in key or value
needs to be backslash-escaped.

disp:dispositions[:additional_stream_specifier]
Matches streams with the given disposition(s). dispositions is a
list of one or more dispositions (as printed by the -dispositions
option) joined with '+'.

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 named filter_name.
Use the -filters option to get a list of all filters.

bsf=bitstream_filter_name
Print detailed information about the bitstream filter named
bitstream_filter_name. Use the -bsfs option to get a list of
all bitstream filters.

protocol=protocol_name
Print detailed information about the protocol named
protocol_name. Use the -protocols option to get a list of all
protocols.

-version
Show version.

-buildconf
Show the build configuration, one option per line.

-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.

-dispositions
Show stream dispositions.

-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 [flags+]loglevel | -v [flags+]loglevel
Set logging level and flags used by the library.

The optional flags prefix can consist of the following values:

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.

level
Indicates that log output should add a "[level]" prefix to each
message line. This can be used as an alternative to log
coloring, e.g. when dumping the log to file.

time
Indicates that log lines should be prefixed with time
information.

datetime
Indicates that log lines should be prefixed with date and time
information.

Flags can also be used alone by adding a '+'/'-' prefix to
set/reset a single flag without affecting other flags or changing
loglevel. When setting both flags and loglevel, a '+' separator is
expected between the last flags value and before 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

For example to enable repeated log output, add the "level" prefix,
and set loglevel to "verbose":

ffmpeg -loglevel repeat+level+verbose -i input output

Another example that enables repeated log output without affecting
current state of "level" prefix flag or loglevel:

ffmpeg [...] -loglevel +repeat

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 can be forced setting the environment
variable AV_LOG_FORCE_COLOR.

-report
Dump full command line and log 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 debug".

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

-cpucount count (global)
Override detection of CPU count. This option is intended for
testing. Do not use it unless you know what you're doing.

ffmpeg -cpucount 2

-max_alloc bytes
Set the maximum size limit for allocating a block on the heap by
ffmpeg's family of malloc functions. Exercise extreme caution when
using this option. Don't use if you do not understand the full
consequence of doing so. Default is INT_MAX.

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:

ffmpeg -i multichannel.mxf -map 0:v:0 -map 0:a:0 -map 0:a:0 -c:a:0 ac3 -b:a:0 640k -ac:a:1 2 -c:a:1 aac -b:2 128k out.mp4

In the above example, a multichannel audio stream is mapped twice for
output. The first instance is encoded with codec ac3 and bitrate 640k.
The second instance is downmixed to 2 channels and encoded with codec
aac. A bitrate of 128k is specified for it using absolute index of the
output stream.

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.

-recast_media (global)
Allow forcing a decoder of a different media type than the one
detected or designated by the demuxer. Useful for decoding media
data muxed as data streams.

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

-isync input_index (input)
Assign an input as a sync source.

This will take the difference between the start times of the target
and reference inputs and offset the timestamps of the target file
by that difference. The source timestamps of the two inputs should
derive from the same clock source for expected results. If "copyts"
is set then "start_at_zero" must also be set. If either of the
inputs has no starting timestamp then no sync adjustment is made.

Acceptable values are those that refer to a valid ffmpeg input
index. If the sync reference is the target index itself or -1, then
no adjustment is made to target timestamps. A sync reference may
not itself be synced to any other input.

Default value is -1.

-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.

-itsscale scale (input,per-stream)
Rescale input timestamps. scale should be a floating point number.

-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 flags for a stream.

Default value: by default, all disposition flags are copied from
the input stream, unless the output stream this option applies to
is fed by a complex filtergraph - in that case no disposition flags
are set by default.

value is a sequence of disposition flags separated by '+' or '-'. A
'+' prefix adds the given disposition, '-' removes it. If the first
flag is also prefixed with '+' or '-', the resulting disposition is
the default value updated by value. If the first flag is not
prefixed, the resulting disposition is value. It is also possible
to clear the disposition by setting it to 0.

If no "-disposition" options were specified for an output file,
ffmpeg will automatically set the 'default' disposition flag on the
first stream of each type, when there are multiple streams of this
type in the output file and no stream of that type is already
marked as default.

The "-dispositions" option lists the known disposition flags.

For example, to make the second audio stream the default stream:

ffmpeg -i in.mkv -c copy -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 in.mkv -c copy -disposition:s:0 0 -disposition:s:1 default out.mkv

To add an embedded cover/thumbnail:

ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4

To add the 'original' and remove the 'comment' disposition flag
from the first audio stream without removing its other disposition
flags:

ffmpeg -i in.mkv -c copy -disposition:a:0 +original-comment out.mkv

To remove the 'original' and add the 'comment' disposition flag to
the first audio stream without removing its other disposition
flags:

ffmpeg -i in.mkv -c copy -disposition:a:0 -original+comment out.mkv

To set only the 'original' and 'comment' disposition flags on the
first audio stream (and remove its other disposition flags):

ffmpeg -i in.mkv -c copy -disposition:a:0 original+comment out.mkv

To remove all disposition flags from the first audio stream:

ffmpeg -i in.mkv -c copy -disposition:a:0 0 out.mkv

Not all muxers support embedded thumbnails, and those who do, only
support a few formats, like JPEG or PNG.

-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.

-stream_group
[map=input_file_id=stream_group][type=type:]st=stream[:st=stream][:stg=stream_group][:id=stream_group_id...]
(output)
Creates a stream group of the specified type and stream_group_id,
or by mapping an input group, adding the specified stream(s) and/or
previously defined stream_group(s) to it.

type can be one of the following:

iamf_audio_element
Groups streams that belong to the same IAMF Audio Element

For this group type, the following options are available

audio_element_type
The Audio Element type. The following values are supported:

channel
Scalable channel audio representation

scene
Ambisonics representation

demixing
Demixing information used to reconstruct a scalable channel
audio representation. This option must be separated from
the rest with a ',', and takes the following key=value
options

parameter_id
An identifier parameters blocks in frames may refer to

dmixp_mode
A pre-defined combination of demixing parameters

recon_gain
Recon gain information used to reconstruct a scalable
channel audio representation. This option must be
separated from the rest with a ',', and takes the following
key=value options

parameter_id
An identifier parameters blocks in frames may refer to

layer
A layer defining a Channel Layout in the Audio Element.
This option must be separated from the rest with a ','.
Several ',' separated entries can be defined, and at least
one must be set.

It takes the following ":"-separated key=value options

ch_layout
The layer's channel layout

flags
The following flags are available:

recon_gain
Whether to signal if recon_gain is present as
metadata in parameter blocks within frames

output_gain
output_gain_flags
Which channels output_gain applies to. The following
flags are available:

FL
FR
BL
BR
TFL
TFR

ambisonics_mode
The ambisonics mode. This has no effect if
audio_element_type is set to channel.

The following values are supported:

mono
Each ambisonics channel is coded as an individual
mono stream in the group

default_w
Default weight value

iamf_mix_presentation
Groups streams that belong to all IAMF Audio Element the same
IAMF Mix Presentation references

For this group type, the following options are available

submix
A sub-mix within the Mix Presentation. This option must be
separated from the rest with a ','. Several ',' separated
entries can be defined, and at least one must be set.

It takes the following ":"-separated key=value options

parameter_id
An identifier parameters blocks in frames may refer to,
for post-processing the mixed audio signal to generate
the audio signal for playback

parameter_rate
The sample rate duration fields in parameters blocks in
frames that refer to this parameter_id are expressed as

default_mix_gain
Default mix gain value to apply when there are no
parameter blocks sharing the same parameter_id for a
given frame

element
References an Audio Element used in this Mix
Presentation to generate the final output audio signal
for playback. This option must be separated from the
rest with a '|'. Several '|' separated entries can be
defined, and at least one must be set.

It takes the following ":"-separated key=value options:

stg The stream_group_id for an Audio Element which this
sub-mix refers to

parameter_id
An identifier parameters blocks in frames may refer
to, for applying any processing to the referenced
and rendered Audio Element before being summed with
other processed Audio Elements

parameter_rate
The sample rate duration fields in parameters
blocks in frames that refer to this parameter_id
are expressed as

default_mix_gain
Default mix gain value to apply when there are no
parameter blocks sharing the same parameter_id for
a given frame

annotations
A key=value string describing the sub-mix element
where "key" is a string conforming to BCP-47 that
specifies the language for the "value" string.
"key" must be the same as the one in the mix's
annotations

headphones_rendering_mode
Indicates whether the input channel-based Audio
Element is rendered to stereo loudspeakers or
spatialized with a binaural renderer when played
back on headphones. This has no effect if the
referenced Audio Element's audio_element_type is
set to channel.

The following values are supported:

stereo
binaural

layout
Specifies the layouts for this sub-mix on which the
loudness information was measured. This option must be
separated from the rest with a '|'. Several '|'
separated entries can be defined, and at least one must
be set.

It takes the following ":"-separated key=value options:

layout_type
loudspeakers
The layout follows the loudspeaker sound system
convention of ITU-2051-3.

binaural
The layout is binaural.

sound_system
Channel layout matching one of Sound Systems A to J
of ITU-2051-3, plus 7.1.2 and 3.1.2 This has no
effect if layout_type is set to binaural.

integrated_loudness
The program integrated loudness information, as
defined in ITU-1770-4.

digital_peak
The digital (sampled) peak value of the audio
signal, as defined in ITU-1770-4.

true_peak
The true peak of the audio signal, as defined in
ITU-1770-4.

dialog_anchored_loudness
The Dialogue loudness information, as defined in
ITU-1770-4.

album_anchored_loudness
The Album loudness information, as defined in
ITU-1770-4.

annotations
A key=value string string describing the mix where "key" is
a string conforming to BCP-47 that specifies the language
for the "value" string. "key" must be the same as the ones
in all sub-mix element's annotationss

E.g. to create an scalable 5.1 IAMF file from several WAV input
files

ffmpeg -i front.wav -i back.wav -i center.wav -i lfe.wav
-map 0:0 -map 1:0 -map 2:0 -map 3:0 -c:a opus
-stream_group type=iamf_audio_element:id=1:st=0:st=1:st=2:st=3,
demixing=parameter_id=998,
recon_gain=parameter_id=101,
layer=ch_layout=stereo,
layer=ch_layout=5.1(side),
-stream_group type=iamf_mix_presentation:id=2:stg=0:annotations=en-us=Mix_Presentation,
submix=parameter_id=100:parameter_rate=48000|element=stg=0:parameter_id=100:annotations=en-us=Scalable_Submix|layout=sound_system=stereo|layout=sound_system=5.1(side)
-streamid 0:0 -streamid 1:1 -streamid 2:2 -streamid 3:3 output.iamf

To copy the two stream groups (Audio Element and Mix Presentation)
from an input IAMF file with four streams into an mp4 output

ffmpeg -i input.iamf -c:a copy -stream_group map=0=0:st=0:st=1:st=2:st=3 -stream_group map=0=1:stg=0
-streamid 0:0 -streamid 1:1 -streamid 2:2 -streamid 3:3 output.mp4

-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

The parameters set for each target are as follows.

VCD

<pal>:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x288 -r 25
-codec:v mpeg1video -g 15 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k

<ntsc>:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x240 -r 30000/1001
-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k

<film>:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x240 -r 24000/1001
-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k

SVCD

<pal>:
-f svcd -packetsize 2324
-s 480x576 -pix_fmt yuv420p -r 25
-codec:v mpeg2video -g 15 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k

<ntsc>:
-f svcd -packetsize 2324
-s 480x480 -pix_fmt yuv420p -r 30000/1001
-codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k

<film>:
-f svcd -packetsize 2324
-s 480x480 -pix_fmt yuv420p -r 24000/1001
-codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k

DVD

<pal>:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x576 -pix_fmt yuv420p -r 25
-codec:v mpeg2video -g 15 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k

<ntsc>:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x480 -pix_fmt yuv420p -r 30000/1001
-codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k

<film>:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x480 -pix_fmt yuv420p -r 24000/1001
-codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k

<pal>:
-f dv
-s 720x576 -pix_fmt yuv420p -r 25
-ar 48000 -ac 2

<ntsc>:
-f dv
-s 720x480 -pix_fmt yuv411p -r 30000/1001
-ar 48000 -ac 2

<film>:
-f dv
-s 720x480 -pix_fmt yuv411p -r 24000/1001
-ar 48000 -ac 2

The "dv50" target is identical to the "dv" target except that the
pixel format set is "yuv422p" for all three standards.

Any user-set value for a parameter above will override the target
preset value. In that case, the output may not comply with the
target standard.

-dn (input/output)
As an input option, blocks all data streams of a file from being
filtered or being automatically selected or mapped for any output.
See "-discard" option to disable streams individually.

As an output option, disables data recording i.e. automatic
selection or mapping of any data stream. For full manual control
see the "-map" option.

-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.

-reinit_filter[:stream_specifier] integer (input,per-stream)
This boolean option determines if the filtergraph(s) to which this
stream is fed gets reinitialized when input frame parameters change
mid-stream. This option is enabled by default as most video and all
audio filters cannot handle deviation in input frame properties.
Upon reinitialization, existing filter state is lost, like e.g. the
frame count "n" reference available in some filters. Any frames
buffered at time of reinitialization are lost. The properties
where a change triggers reinitialization are, for video, frame
resolution or pixel format; for audio, sample format, sample rate,
channel count or channel layout.

-drop_changed[:stream_specifier] integer (input,per-stream)
This boolean option determines whether a frame with differing frame
parameters mid-stream gets dropped instead of leading to
filtergraph reinitialization, as that would lead to loss of filter
state. Generally useful to avoid corrupted yet decodable packets in
live streaming inputs. Default is false.

-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.

-filter_buffered_frames nb_frames (global)
Defines the maximum number of buffered frames allowed in a
filtergraph. Under normal circumstances, a filtergraph should not
buffer more than a few frames, especially if frames are being fed
to it and read from it in a balanced way (which is the intended
behavior in ffmpeg). That said, this option allows you to limit the
total number of frames buffered across all links in a filtergraph.
If more frames are generated, filtering is aborted and an error is
returned. The default value is 0, which means no limit.

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

-stats (global)
Log encoding progress/statistics as "info"-level log (see
"-loglevel"). It is on by default, to explicitly disable it you
need to specify "-nostats".

-stats_period time (global)
Set period at which encoding progress/statistics are updated.
Default is 0.5 seconds.

-print_graphs (global)
Prints execution graph details to stderr in the format set via
-print_graphs_format.

-print_graphs_file filename (global)
Writes execution graph details to the specified file in the format
set via -print_graphs_format.

-print_graphs_format format (global)
Sets the output format (available formats are: default, compact,
csv, flat, ini, json, xml, mermaid, mermaidhtml) The default format
is json.

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

Progress information is written periodically 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" with the value "continue"
or "end".

The update period is set using "-stats_period".

For example, log progress information to stdout:

ffmpeg -progress pipe:1 -i in.mkv out.mkv

-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/latency 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 hash and framehash muxers.

tee
The tee muxer can be used to write the same data to several outputs,
such as files or streams. It can be used, for example, to stream a
video over a network and save it to disk at the same time.

It is different from specifying several outputs to the ffmpeg
command-line tool. With the tee muxer, the audio and video data will be
encoded only once. With conventional multiple outputs, multiple
encoding operations in parallel are initiated, which can be a very
expensive process. The tee muxer is not useful when using the
libavformat API directly because it is then possible to feed the same
packets to several muxers directly.

Since the tee muxer does not represent any particular output format,
ffmpeg cannot auto-select output streams. So all streams intended for
output must be specified using "-map". See the examples below.

Some encoders may need different options depending on the output
format; the auto-detection of this can not work with the tee muxer, so
they need to be explicitly specified. The main example is the
global_header flag.

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, those must be
escaped (see the "Quoting and escaping" section in the ffmpeg-utils(1)
manual).

Options

use_fifo bool
If set to 1, slave outputs will be processed in separate threads
using the 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.

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. Required if it cannot be guessed from the
output URL.

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

It is possible to specify to which streams a given bitstream filter
applies, by appending a stream specifier to the option separated by
"/". spec must be a stream specifier (see Format stream
specifiers).

If the stream specifier is not specified, the bitstream filters
will be applied to all streams in the output. This will cause that
output operation to fail if the output contains streams to which
the bitstream filter cannot be applied e.g. "h264_mp4toannexb"
being applied to an output containing an audio stream.

Options for a bitstream filter must be specified in the form of
"opt=value".

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

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.

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 mapped streams. This will cause that output operation to
fail if the output format does not accept all mapped 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

• Encode something and both archive it in a WebM file and stream it
as MPEG-TS over UDP:

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/"

• 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/"

• 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=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"

• As above, 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=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"

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

webm_dash_manifest
WebM DASH Manifest muxer.

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

For more information see:

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

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

Options

This muxer supports the following options:

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

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

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

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

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

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

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

Example

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

whip
WebRTC (Real-Time Communication) muxer that supports sub-second latency
streaming according to the WHIP (WebRTC-HTTP ingestion protocol)
specification.

This is an experimental feature.

It uses HTTP as a signaling protocol to exchange SDP capabilities and
ICE lite candidates. Then, it uses STUN binding requests and responses
to establish a session over UDP. Subsequently, it initiates a DTLS
handshake to exchange the SRTP encryption keys. Lastly, it splits video
and audio frames into RTP packets and encrypts them using SRTP.

Ensure that you use H.264 without B frames and Opus for the audio
codec. For example, to convert an input file with ffmpeg to WebRTC:

ffmpeg -re -i input.mp4 -acodec libopus -ar 48000 -ac 2 \
-vcodec libx264 -profile:v baseline -tune zerolatency -threads 1 -bf 0 \
-f whip "http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream"

For this example, we have employed low latency options, resulting in an
end-to-end latency of approximately 150ms.

Options

This muxer supports the following options:

handshake_timeout integer
Set the timeout in milliseconds for ICE and DTLS handshake.
Default value is 5000.

pkt_size integer
Set the maximum size, in bytes, of RTP packets that send out.
Default value is 1500.

authorization string
The optional Bearer token for WHIP Authorization.

cert_file string
The optional certificate file path for DTLS.

key_file string
The optional private key file path for DTLS.

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 nanoseconds.

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.

amqp
Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker
based publish-subscribe communication protocol.

FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A
separate AMQP broker must also be run. An example open-source AMQP
broker is RabbitMQ.

After starting the broker, an FFmpeg client may stream data to the
broker using the command:

ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost]

Where hostname and port (default is 5672) is the address of the broker.
The client may also set a user/password for authentication. The default
for both fields is "guest". Name of virtual host on broker can be set
with vhost. The default value is "/".

Multiple subscribers may stream from the broker using the command:

ffplay amqp://[[user]:[password]@]hostname[:port][/vhost]

In RabbitMQ all data published to the broker flows through a specific
exchange, and each subscribing client has an assigned queue/buffer.
When a packet arrives at an exchange, it may be copied to a client's
queue depending on the exchange and routing_key fields.

The following options are supported:

exchange
Sets the exchange to use on the broker. RabbitMQ has several
predefined exchanges: "amq.direct" is the default exchange, where
the publisher and subscriber must have a matching routing_key;
"amq.fanout" is the same as a broadcast operation (i.e. the data is
forwarded to all queues on the fanout exchange independent of the
routing_key); and "amq.topic" is similar to "amq.direct", but
allows for more complex pattern matching (refer to the RabbitMQ
documentation).

routing_key
Sets the routing key. The default value is "amqp". The routing key
is used on the "amq.direct" and "amq.topic" exchanges to decide
whether packets are written to the queue of a subscriber.

pkt_size
Maximum size of each packet sent/received to the broker. Default is
131072. Minimum is 4096 and max is any large value (representable
by an int). When receiving packets, this sets an internal buffer
size in FFmpeg. It should be equal to or greater than the size of
the published packets to the broker. Otherwise the received message
may be truncated causing decoding errors.

connection_timeout
The timeout in seconds during the initial connection to the broker.
The default value is rw_timeout, or 5 seconds if rw_timeout is not
set.

delivery_mode mode
Sets the delivery mode of each message sent to broker. The
following values are accepted:

persistent
Delivery mode set to "persistent" (2). This is the default
value. Messages may be written to the broker's disk depending
on its setup.

non-persistent
Delivery mode set to "non-persistent" (1). Messages will stay
in broker's memory unless the broker is under memory pressure.

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.

The accepted options are:

read_ahead_limit
Amount in bytes that may be read ahead when seeking isn't
supported. Range is -1 to INT_MAX. -1 for unlimited. Default is
65536.

URL Syntax is

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.

concatf
Physical concatenation protocol using a line break delimited list of
resources.

Read and seek from many resources in sequence as if they were a unique
resource.

A URL accepted by this protocol has the syntax:

concatf:<URL>

where URL is the url containing a line break delimited list of
resources to be concatenated, each one possibly specifying a distinct
protocol. Special characters must be escaped with backslash or single
quotes. See the "Quoting and escaping" section in the ffmpeg-utils(1)
manual.

For example to read a sequence of files split1.mpeg, split2.mpeg,
split3.mpeg listed in separate lines within a file split.txt with
ffplay use the command:

ffplay concatf:split.txt

Where split.txt contains the lines:

split1.mpeg
split2.mpeg
split3.mpeg

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

fd
File descriptor access protocol.

The accepted syntax is:

fd: -fd <file_descriptor>

If fd is not specified, by default the stdout file descriptor will be
used for writing, stdin for reading. Unlike the pipe protocol, fd
protocol has seek support if it corresponding to a regular file. fd
protocol doesn't support pass file descriptor via URL for security.

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.

fd Set file descriptor.

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.

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).

seekable
Controls if seekability is advertised on the file. 0 means
non-seekable, -1 means auto (seekable for normal files,
non-seekable for named pipes).

Many demuxers handle seekable and non-seekable resources
differently, overriding this might speed up opening certain files
at the cost of losing some features (e.g. accurate seeking).

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-user
Set a user to be used for authenticating to the FTP server. This is
overridden by the user in the FTP URL.

ftp-password
Set a password to be used for authenticating to the FTP server.
This is overridden by the password in the FTP URL, or by
ftp-anonymous-password if no user is set.

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

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

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

gopher
Gopher protocol.

gophers
Gophers protocol.

The Gopher protocol with TLS encapsulation.

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.

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.

content_type
Set a specific content type for the POST messages or for listen
mode.

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

referer
Set the Referer header. Include 'Referer: URL' header in HTTP
request.

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

post_data
Set custom HTTP post data.

mime_type
Export the MIME type.

http_version
Exports the HTTP response version number. Usually "1.0" or "1.1".

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.

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.

metadata
Set an exported dictionary containing Icecast metadata from the
bitstream, if present. Only useful with the C API.

auth_type
Set HTTP authentication type. No option for Digest, since this
method requires getting nonce parameters from the server first and
can't be used straight away like Basic.

none
Choose the HTTP authentication type automatically. This is the
default.

basic
Choose the HTTP basic authentication.

Basic authentication sends a Base64-encoded string that
contains a user name and password for the client. Base64 is not
a form of encryption and should be considered the same as
sending the user name and password in clear text (Base64 is a
reversible encoding). If a resource needs to be protected,
strongly consider using an authentication scheme other than
basic authentication. HTTPS/TLS should be used with basic
authentication. Without these additional security
enhancements, basic authentication should not be used to
protect sensitive or valuable information.

send_expect_100
Send an Expect: 100-continue header for POST. If set to 1 it will
send, if set to 0 it won't, if set to -1 it will try to send if it
is applicable. Default value is -1.

location
An exported dictionary containing the content location. Only useful
with the C API.

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.

reconnect
Reconnect automatically when disconnected before EOF is hit.

reconnect_at_eof
If set then eof is treated like an error and causes reconnection,
this is useful for live / endless streams.

reconnect_on_network_error
Reconnect automatically in case of TCP/TLS errors during connect.

reconnect_on_http_error
A comma separated list of HTTP status codes to reconnect on. The
list can include specific status codes (e.g. '503') or the strings
'4xx' / '5xx'.

reconnect_streamed
If set then even streamed/non seekable streams will be reconnected
on errors.

reconnect_delay_max
Set the maximum delay in seconds after which to give up
reconnecting.

reconnect_max_retries
Set the maximum number of times to retry a connection. Default
unset.

reconnect_delay_total_max
Set the maximum total delay in seconds after which to give up
reconnecting.

respect_retry_after
If enabled, and a Retry-After header is encountered, its requested
reconnection delay will be honored, rather than using exponential
backoff. Useful for 429 and 503 errors. Default enabled.

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 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>

resource
The resource requested by a client, when the experimental HTTP
server is in use.

reply_code
The HTTP code returned to the client, when the experimental HTTP
server is in use.

short_seek_size
Set the threshold, in bytes, for when a readahead should be
preferred over a seek and new HTTP request. This is useful, for
example, to make sure the same connection is used for reading large
video packets with small audio packets in between.

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.

tls Establish a TLS (HTTPS) connection to Icecast.

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

ipfs
InterPlanetary File System (IPFS) protocol support. One can access
files stored on the IPFS network through so-called gateways. These are
http(s) endpoints. This protocol wraps the IPFS native protocols
(ipfs:// and ipns://) to be sent to such a gateway. Users can (and
should) host their own node which means this protocol will use one's
local gateway to access files on the IPFS network.

This protocol accepts the following options:

gateway
Defines the gateway to use. When not set, the protocol will first
try locating the local gateway by looking at $IPFS_GATEWAY,
$IPFS_PATH and "$HOME/.ipfs/", in that order.

One can use this protocol in 2 ways. Using IPFS:

ffplay ipfs://<hash>

Or the IPNS protocol (IPNS is mutable IPFS):

ffplay ipns://<hash>

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>]

If fd isn't specified, 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:

fd Set file descriptor.

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>

rist
Reliable Internet Streaming Transport protocol

The accepted options are:

rist_profile
Supported values:

simple
main
This one is default.

advanced

buffer_size
Set internal RIST buffer size in milliseconds for retransmission of
data. Default value is 0 which means the librist default (1 sec).
Maximum value is 30 seconds.

fifo_size
Size of the librist receiver output fifo in number of packets. This
must be a power of 2. Defaults to 8192 (vs the librist default of
1024).

overrun_nonfatal=1|0
Survive in case of librist fifo buffer overrun. Default value is 0.

pkt_size
Set maximum packet size for sending data. 1316 by default.

log_level
Set loglevel for RIST logging messages. You only need to set this
if you explicitly want to enable debug level messages or packet
loss simulation, otherwise the regular loglevel is respected.

secret
Set override of encryption secret, by default is unset.

encryption
Set encryption type, by default is disabled. Acceptable values are
128 and 256.

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_enhanced_codecs
Specify the list of codecs the client advertises to support in an
enhanced RTMP stream. This option should be set to a comma
separated list of fourcc values, like "hvc1,av01,vp09" for multiple
codecs or "hvc1" for only one codec. The specified list will be
presented in the "fourCcLive" property of the Connect Command
Message.

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.

tcp_nodelay=1|0
Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

Remark: Writing to the socket is currently not optimized to
minimize system calls and reduces the efficiency / effect of
TCP_NODELAY.

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>][?<options>]

port specifies the RTP port to use.

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

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.

buffer_size=size
Set the maximum UDP socket buffer size in bytes.

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.

localaddr=addr
Local IP address of a network interface used for sending packets or
joining multicast groups.

timeout=n
Set timeout (in microseconds) of socket I/O operations to n.

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".

Muxer

The following options are supported.

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.

Default value is 0.

rtsp_flags
Set RTSP flags.

The following values are accepted:

latm
Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.

rfc2190
Use RFC 2190 packetization instead of RFC 4629 for H.263.

skip_rtcp
Don't send RTCP sender reports.

h264_mode0
Use mode 0 for H.264 in RTP.

send_bye
Send RTCP BYE packets when finishing.

Default value is 0.

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

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

buffer_size
Set the maximum socket buffer size in bytes.

pkt_size
Set max send packet size (in bytes). Default value is 1472.

Demuxer

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.

https
Use HTTPs tunneling as lower transport protocol, which is
useful for passing proxies and widely used for security
consideration.

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.

satip_raw
Export raw MPEG-TS stream instead of demuxing. The flag will
simply write out the raw stream, with the original PAT/PMT/PIDs
intact.

Default value is none.

allowed_media_types
Set media types to accept from the server.

The following flags are accepted:

video
audio
data
subtitle

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.

listen_timeout
Set maximum timeout (in seconds) to establish an initial
connection. Setting listen_timeout > 0 sets rtsp_flags to listen.
Default is -1 which means an infinite timeout when listen mode is
set.

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

timeout
Set socket TCP I/O timeout in microseconds.

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

buffer_size
Set the maximum socket buffer size in bytes.

When receiving data over UDP, the demuxer tries to reorder received
packets (since they may arrive out of order, or packets may get lost
totally). This can be disabled by setting the maximum demuxing delay to
zero (via the "max_delay" field of AVFormatContext).

When watching multi-bitrate Real-RTSP streams with ffplay, the streams
to display can be chosen with "-vst" n and "-ast" n for video and audio
respectively, and can be switched on the fly by pressing "v" and "a".

Examples

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

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

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

• Watch a stream tunneled over HTTP:

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

• Send a stream in realtime to a RTSP server, for others to watch:

ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp

• Receive a stream in realtime:

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

sap
Session Announcement Protocol (RFC 2974). This is not technically a
protocol handler in libavformat, it is a muxer and demuxer. It is used
for signalling of RTP streams, by announcing the SDP for the streams
regularly on a separate port.

Muxer

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

sap://<destination>[:<port>][?<options>]

The RTP packets are sent to destination on port port, or to port 5004
if no port is specified. options is a "&"-separated list. The
following options are supported:

announce_addr=address
Specify the destination IP address for sending the announcements
to. If omitted, the announcements are sent to the commonly used
SAP announcement multicast address 224.2.127.254 (sap.mcast.net),
or ff0e::2:7ffe if destination is an IPv6 address.

announce_port=port
Specify the port to send the announcements on, defaults to 9875 if
not specified.

ttl=ttl
Specify the time to live value for the announcements and RTP
packets, defaults to 255.

same_port=0|1
If set to 1, send all RTP streams on the same port pair. If zero
(the default), all streams are sent on unique ports, with each
stream on a port 2 numbers higher than the previous. VLC/Live555
requires this to be set to 1, to be able to receive the stream.
The RTP stack in libavformat for receiving requires all streams to
be sent on unique ports.

Example command lines follow.

To broadcast a stream on the local subnet, for watching in VLC:

ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1

Similarly, for watching in ffplay:

ffmpeg -re -i <input> -f sap sap://224.0.0.255

And for watching in ffplay, over IPv6:

ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]

Demuxer

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

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

address is the multicast address to listen for announcements on, if
omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the
port that is listened on, 9875 if omitted.

The demuxers listens for announcements on the given address and port.
Once an announcement is received, it tries to receive that particular
stream.

Example command lines follow.

To play back the first stream announced on the normal SAP multicast
address:

ffplay sap://

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

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

sctp
Stream Control Transmission Protocol.

The accepted URL syntax is:

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

The protocol accepts the following options:

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

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

srt
Haivision Secure Reliable Transport Protocol via libsrt.

The supported syntax for a SRT URL is:

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

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

options contains a list of '-key val' options.

This protocol accepts the following options.

connect_timeout=milliseconds
Connection timeout; SRT cannot connect for RTT > 1500 msec (2
handshake exchanges) with the default connect timeout of 3 seconds.
This option applies to the caller and rendezvous connection modes.
The connect timeout is 10 times the value set for the rendezvous
mode (which can be used as a workaround for this connection problem
with earlier versions).

ffs=bytes
Flight Flag Size (Window Size), in bytes. FFS is actually an
internal parameter and you should set it to not less than
recv_buffer_size and mss. The default value is relatively large,
therefore unless you set a very large receiver buffer, you do not
need to change this option. Default value is 25600.

inputbw=bytes/seconds
Sender nominal input rate, in bytes per seconds. Used along with
oheadbw, when maxbw is set to relative (0), to calculate maximum
sending rate when recovery packets are sent along with the main
media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set
while maxbw is set to relative (0), the actual input rate is
evaluated inside the library. Default value is 0.

iptos=tos
IP Type of Service. Applies to sender only. Default value is 0xB8.

ipttl=ttl
IP Time To Live. Applies to sender only. Default value is 64.

latency=microseconds
Timestamp-based Packet Delivery Delay. Used to absorb bursts of
missed packet retransmissions. This flag sets both rcvlatency and
peerlatency to the same value. Note that prior to version 1.3.0
this is the only flag to set the latency, however this is
effectively equivalent to setting peerlatency, when side is sender
and rcvlatency when side is receiver, and the bidirectional stream
sending is not supported.

listen_timeout=microseconds
Set socket listen timeout.

maxbw=bytes/seconds
Maximum sending bandwidth, in bytes per seconds. -1 infinite
(CSRTCC limit is 30mbps) 0 relative to input rate (see inputbw) >0
absolute limit value Default value is 0 (relative)

mode=caller|listener|rendezvous
Connection mode. caller opens client connection. listener starts
server to listen for incoming connections. rendezvous use
Rendez-Vous connection mode. Default value is caller.

mss=bytes
Maximum Segment Size, in bytes. Used for buffer allocation and rate
calculation using a packet counter assuming fully filled packets.
The smallest MSS between the peers is used. This is 1500 by default
in the overall internet. This is the maximum size of the UDP
packet and can be only decreased, unless you have some unusual
dedicated network settings. Default value is 1500.

nakreport=1|0
If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
periodically until a lost packet is retransmitted or intentionally
dropped. Default value is 1.

oheadbw=percents
Recovery bandwidth overhead above input rate, in percents. See
inputbw. Default value is 25%.

passphrase=string
HaiCrypt Encryption/Decryption Passphrase string, length from 10 to
79 characters. The passphrase is the shared secret between the
sender and the receiver. It is used to generate the Key Encrypting
Key using PBKDF2 (Password-Based Key Derivation Function). It is
used only if pbkeylen is non-zero. It is used on the receiver only
if the received data is encrypted. The configured passphrase
cannot be recovered (write-only).

enforced_encryption=1|0
If true, both connection parties must have the same password set
(including empty, that is, with no encryption). If the password
doesn't match or only one side is unencrypted, the connection is
rejected. Default is true.

kmrefreshrate=packets
The number of packets to be transmitted after which the encryption
key is switched to a new key. Default is -1. -1 means auto
(0x1000000 in srt library). The range for this option is integers
in the 0 - "INT_MAX".

kmpreannounce=packets
The interval between when a new encryption key is sent and when
switchover occurs. This value also applies to the subsequent
interval between when switchover occurs and when the old encryption
key is decommissioned. Default is -1. -1 means auto (0x1000 in srt
library). The range for this option is integers in the 0 -
"INT_MAX".

snddropdelay=microseconds
The sender's extra delay before dropping packets. This delay is
added to the default drop delay time interval value.

Special value -1: Do not drop packets on the sender at all.

payload_size=bytes
Sets the maximum declared size of a packet transferred during the
single call to the sending function in Live mode. Use 0 if this
value isn't used (which is default in file mode). Default is -1
(automatic), which typically means MPEG-TS; if you are going to use
SRT to send any different kind of payload, such as, for example,
wrapping a live stream in very small frames, then you can use a
bigger maximum frame size, though not greater than 1456 bytes.

pkt_size=bytes
Alias for payload_size.

peerlatency=microseconds
The latency value (as described in rcvlatency) that is set by the
sender side as a minimum value for the receiver.

pbkeylen=bytes
Sender encryption key length, in bytes. Only can be set to 0, 16,
24 and 32. Enable sender encryption if not 0. Not required on
receiver (set to 0), key size obtained from sender in HaiCrypt
handshake. Default value is 0.

rcvlatency=microseconds
The time that should elapse since the moment when the packet was
sent and the moment when it's delivered to the receiver application
in the receiving function. This time should be a buffer time large
enough to cover the time spent for sending, unexpectedly extended
RTT time, and the time needed to retransmit the lost UDP packet.
The effective latency value will be the maximum of this options'
value and the value of peerlatency set by the peer side. Before
version 1.3.0 this option is only available as latency.

recv_buffer_size=bytes
Set UDP receive buffer size, expressed in bytes.

send_buffer_size=bytes
Set UDP send buffer size, expressed in bytes.

timeout=microseconds
Set raise error timeouts for read, write and connect operations.
Note that the SRT library has internal timeouts which can be
controlled separately, the value set here is only a cap on those.

tlpktdrop=1|0
Too-late Packet Drop. When enabled on receiver, it skips missing
packets that have not been delivered in time and delivers the
following packets to the application when their time-to-play has
come. It also sends a fake ACK to the sender. When enabled on
sender and enabled on the receiving peer, the sender drops the
older packets that have no chance of being delivered in time. It
was automatically enabled in the sender if the receiver supports
it.

sndbuf=bytes
Set send buffer size, expressed in bytes.

rcvbuf=bytes
Set receive buffer size, expressed in bytes.

Receive buffer must not be greater than ffs.

lossmaxttl=packets
The value up to which the Reorder Tolerance may grow. When Reorder
Tolerance is > 0, then packet loss report is delayed until that
number of packets come in. Reorder Tolerance increases every time a
"belated" packet has come, but it wasn't due to retransmission
(that is, when UDP packets tend to come out of order), with the
difference between the latest sequence and this packet's sequence,
and not more than the value of this option. By default it's 0,
which means that this mechanism is turned off, and the loss report
is always sent immediately upon experiencing a "gap" in sequences.

minversion
The minimum SRT version that is required from the peer. A
connection to a peer that does not satisfy the minimum version
requirement will be rejected.

The version format in hex is 0xXXYYZZ for x.y.z in human readable
form.

streamid=string
A string limited to 512 characters that can be set on the socket
prior to connecting. This stream ID will be able to be retrieved by
the listener side from the socket that is returned from srt_accept
and was connected by a socket with that set stream ID. SRT does not
enforce any special interpretation of the contents of this string.
This option doesnt make sense in Rendezvous connection; the result
might be that simply one side will override the value from the
other side and its the matter of luck which one would win

srt_streamid=string
Alias for streamid to avoid conflict with ffmpeg command line
option.

smoother=live|file
The type of Smoother used for the transmission for that socket,
which is responsible for the transmission and congestion control.
The Smoother type must be exactly the same on both connecting
parties, otherwise the connection is rejected.

messageapi=1|0
When set, this socket uses the Message API, otherwise it uses
Buffer API. Note that in live mode (see transtype) theres only
message API available. In File mode you can chose to use one of two
modes:

Stream API (default, when this option is false). In this mode you
may send as many data as you wish with one sending instruction, or
even use dedicated functions that read directly from a file. The
internal facility will take care of any speed and congestion
control. When receiving, you can also receive as many data as
desired, the data not extracted will be waiting for the next call.
There is no boundary between data portions in the Stream mode.

Message API. In this mode your single sending instruction passes
exactly one piece of data that has boundaries (a message). Contrary
to Live mode, this message may span across multiple UDP packets and
the only size limitation is that it shall fit as a whole in the
sending buffer. The receiver shall use as large buffer as necessary
to receive the message, otherwise the message will not be given up.
When the message is not complete (not all packets received or there
was a packet loss) it will not be given up.

transtype=live|file
Sets the transmission type for the socket, in particular, setting
this option sets multiple other parameters to their default values
as required for a particular transmission type.

live: Set options as for live transmission. In this mode, you
should send by one sending instruction only so many data that fit
in one UDP packet, and limited to the value defined first in
payload_size (1316 is default in this mode). There is no speed
control in this mode, only the bandwidth control, if configured, in
order to not exceed the bandwidth with the overhead transmission
(retransmitted and control packets).

file: Set options as for non-live transmission. See messageapi for
further explanations

linger=seconds
The number of seconds that the socket waits for unsent data when
closing. Default is -1. -1 means auto (off with 0 seconds in live
mode, on with 180 seconds in file mode). The range for this option
is integers in the 0 - "INT_MAX".

tsbpd=1|0
When true, use Timestamp-based Packet Delivery mode. The default
behavior depends on the transmission type: enabled in live mode,
disabled in file mode.

For more information see: <https://github.com/Haivision/srt>.

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. If set to 0,
extract till end of file.

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

Play a MPEG-TS file from start offset till end:

subfile,,start,32815239,end,0,,:video.ts

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=2|1|0
Listen for an incoming connection. 0 disables listen, 1 enables
listen in single client mode, 2 enables listen in multi-client
mode. Default value is 0.

local_addr=addr
Local IP address of a network interface used for tcp socket
connect.

local_port=port
Local port used for tcp socket connect.

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.

tcp_nodelay=1|0
Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

Remark: Writing to the socket is currently not optimized to
minimize system calls and reduces the efficiency / effect of
TCP_NODELAY.

tcp_mss=bytes
Set maximum segment size for outgoing TCP packets, expressed in
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 other
backends, 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.

http_proxy
The HTTP proxy to tunnel through, e.g. "http://example.com:1234".
The proxy must support the CONNECT method.

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 32 KB for output, 384 KB for input.
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
Local IP address of a network interface used for sending packets or
joining multicast groups.

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 from the specified addresses. In case of
multicast, also subscribe to multicast traffic coming from these
addresses only.

block=address[,address]
Ignore packets sent from the specified addresses. In case of
multicast, also exclude the source addresses in the multicast
subscription.

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

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

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

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

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

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

Examples

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

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

• Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP
packets, using a large input buffer:

ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535

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

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

unix
Unix local socket

The required syntax for a Unix socket URL is:

unix://<filepath>

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

timeout
Timeout in ms.

listen
Create the Unix socket in listening mode.

zmq
ZeroMQ asynchronous messaging using the libzmq library.

This library supports unicast streaming to multiple clients without
relying on an external server.

The required syntax for streaming or connecting to a stream is:

zmq:tcp://ip-address:port

Example: Create a localhost stream on port 5555:

ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555

Multiple clients may connect to the stream using:

ffplay zmq:tcp://127.0.0.1:5555

Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub
pattern. The server side binds to a port and publishes data. Clients
connect to the server (via IP address/port) and subscribe to the
stream. The order in which the server and client start generally does
not matter.

ffmpeg must be compiled with the --enable-libzmq option to support this
protocol.

Options can be set on the ffmpeg/ffplay command line. The following
options are supported:

pkt_size
Forces the maximum packet size for sending/receiving data. The
default value is 131,072 bytes. On the server side, this sets the
maximum size of sent packets via ZeroMQ. On the clients, it sets an
internal buffer size for receiving packets. Note that pkt_size on
the clients should be equal to or greater than pkt_size on the
server. Otherwise the received message may be truncated causing
decoding errors.

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.

android_camera
Android camera input device.

This input devices uses the Android Camera2 NDK API which is available
on devices with API level 24+. The availability of android_camera is
autodetected during configuration.

This device allows capturing from all cameras on an Android device,
which are integrated into the Camera2 NDK API.

The available cameras are enumerated internally and can be selected
with the camera_index parameter. The input file string is discarded.

Generally the back facing camera has index 0 while the front facing
camera has index 1.

Options

video_size
Set the video size given as a string such as 640x480 or hd720.
Falls back to the first available configuration reported by Android
if requested video size is not available or by default.

framerate
Set the video framerate. Falls back to the first available
configuration reported by Android if requested framerate is not
available or by default (-1).

camera_index
Set the index of the camera to use. Default is 0.

input_queue_size
Set the maximum number of frames to buffer. Default is 5.

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.

-capture_raw_data
Capture the raw device data. Default is 0. Using this option may
result in receiving the underlying data delivered to the
AVFoundation framework. E.g. for muxed devices that sends raw DV
data to the framework (like tape-based camcorders), setting this
option to false results in extracted video frames captured in the
designated pixel format only. Setting this option to true results
in receiving the raw DV stream untouched.

Examples

• Print the list of AVFoundation supported devices and exit:

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

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

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

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

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

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

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

• Record raw DV data from a suitable input device and write the
output into out.dv:

$ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv

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. This option is deprecated, please use the "-sources" option
of ffmpeg to list the available input devices.

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). Default behavior is autodetection of the input video
format, if the hardware supports it.

raw_format
Set the pixel format of the captured video. Available values are:

auto
This is the default which means 8-bit YUV 422 or 8-bit ARGB if
format autodetection is used, 8-bit YUV 422 otherwise.

uyvy422
8-bit YUV 422.

yuv422p10
10-bit YUV 422.

argb
8-bit RGB.

bgra
8-bit RGB.

rgb10
10-bit RGB.

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/profile mode. Must be unset, half,
full, one_sub_device_full, one_sub_device_half,
two_sub_device_full, four_sub_device_half Defaults to unset.

Note: DeckLink SDK 11.0 have replaced the duplex property by a
profile property. For the DeckLink Duo 2 and DeckLink Quad 2, a
profile is shared between any 2 sub-devices that utilize the same
connectors. For the DeckLink 8K Pro, a profile is shared between
all 4 sub-devices. So DeckLink 8K Pro support four profiles.

Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
one_sub_device_full, one_sub_device_half, two_sub_device_full,
four_sub_device_half

Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2: half,
full

timecode_format
Timecode type to include in the frame and video stream metadata.
Must be none, rp188vitc, rp188vitc2, rp188ltc, rp188hfr, rp188any,
vitc, vitc2, or serial. Defaults to none (not included).

In order to properly support 50/60 fps timecodes, the ordering of
the queried timecode types for rp188any is HFR, VITC1, VITC2 and
LTC for >30 fps content. Note that this is slightly different to
the ordering used by the DeckLink API, which is HFR, VITC1, LTC,
VITC2.

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, wallclock or abs_wallclock. Defaults to video.

audio_pts
Sets the audio packet timestamp source. Must be video, audio,
reference, wallclock or abs_wallclock. Defaults to audio.

draw_bars
If set to true, color bars are drawn in the event of a signal loss.
Defaults to true. This option is deprecated, please use the
"signal_loss_action" option.

signal_loss_action
Sets the action to take in the event of a signal loss. Accepts one
of the following values:

1, none
Do nothing on signal loss. This usually results in black
frames.

2, bars
Draw color bars on signal loss. Only supported for 8-bit input
signals.

3, repeat
Repeat the last video frame on signal loss.

Defaults to bars.

queue_size
Sets maximum input buffer size in bytes. If the buffering reaches
this value, incoming frames will be dropped. Defaults to
1073741824.

audio_depth
Sets the audio sample bit depth. Must be 16 or 32. Defaults to 16.

decklink_copyts
If set to true, timestamps are forwarded as they are without
removing the initial offset. Defaults to false.

timestamp_align
Capture start time alignment in seconds. If set to nonzero, input
frames are dropped till the system timestamp aligns with configured
value. Alignment difference of up to one frame duration is
tolerated. This is useful for maintaining input synchronization
across N different hardware devices deployed for 'N-way'
redundancy. The system time of different hardware devices should be
synchronized with protocols such as NTP or PTP, before using this
option. Note that this method is not foolproof. In some border
cases input synchronization may not happen due to thread scheduling
jitters in the OS. Either sync could go wrong by 1 frame or in a
rarer case timestamp_align seconds. Defaults to 0.

wait_for_tc (bool)
Drop frames till a frame with timecode is received. Sometimes
serial timecode isn't received with the first input frame. If that
happens, the stored stream timecode will be inaccurate. If this
option is set to true, input frames are dropped till a frame with
timecode is received. Option timecode_format must be specified.
Defaults to false.

enable_klv(bool)
If set to true, extracts KLV data from VANC and outputs KLV
packets. KLV VANC packets are joined based on MID and PSC fields
and aggregated into one KLV packet. Defaults to false.

Examples

• List input devices:

ffmpeg -sources decklink

• List supported formats:

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

• Capture video clip at 1080i50:

ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi

• Capture video clip at 1080i50 10 bit:

ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi

• 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

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.

use_video_device_timestamps
If set to false, the timestamp for video frames will be derived
from the wallclock instead of the timestamp provided by the capture
device. This allows working around devices that provide unreliable
timestamps.

Examples

• Print the list of DirectShow supported devices and exit:

$ ffmpeg -list_devices true -f dshow -i dummy

• Open video device Camera:

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

• Open second video device with name Camera:

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

• Open video device Camera and audio device Microphone:

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

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

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

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

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

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

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

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.

Amongst options for the input filenames are such elements as:

desktop

title=<window_title>

hwnd=<window_hwnd>

The first option will capture the entire desktop, or a fixed region of
the desktop. The second and third options 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

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

ffplay -f iec61883 -i auto

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

ffmpeg -f iec61883 -i auto -dvbuffer 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.

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. This can be autodetected if you
are running Linux 5.7 or later, but needs to be provided for
earlier versions. Defaults to bgr0, which is the most common
format used by the Linux console and Xorg X server.

format_modifier
Format modifier to signal on output frames. This is necessary to
import correctly into some APIs. It can be autodetected if you are
running Linux 5.7 or later, but will need to be provided explicitly
when needed in earlier versions. 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

• 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

• 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

• To capture only part of a plane the output can be cropped - this
can be used to capture a single window, as long as it has a known
absolute position and size. For example, to capture and encode the
middle quarter of a 1920x1080 plane:

ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4

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. The
filtergraph is specified through the option graph.

Options

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

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

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

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

dumpgraph
Dump graph to stderr.

Examples

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

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

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

ffplay -f lavfi color=c=pink

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

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

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

ffplay -f lavfi "amovie=test.wav"

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

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

• Dump decoded frames to images and Closed Captions to an RCWT
backup:

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

Options

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

pixel_format
Select the pixel format. Default is "uyvy422".

video_size
Set the video size given as a string such as "640x480" or "hd720".
Default is "qvga".

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
This option does nothing and is deprecated.

fragment_size
Specify the size in bytes of the minimal buffering fragment in
PulseAudio, it will affect the audio latency. By default it is set
to 50 ms amount of data.

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:

• List supported formats for a video4linux2 device:

ffplay -f video4linux2 -list_formats all /dev/video0

• Grab and show the input of a video4linux2 device:

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

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

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

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

Options

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

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

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

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

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

framerate
Set the preferred video frame rate.

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

Available values are:

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

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

compressed
Show only compressed formats.

list_standards
List supported standards and exit.

Available values are:

all Show all supported standards.

timestamps, ts
Set type of timestamps for grabbed frames.

Available values are:

default
Use timestamps from the kernel.

abs Use absolute timestamps (wall clock).

mono2abs
Force conversion from monotonic to absolute timestamps.

Default value is "default".

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

vfwcap
VfW (Video for Windows) capture input device.

The filename passed as input is the capture driver number, ranging from
0 to 9. You may use "list" as filename to print a list of drivers. Any
other filename will be interpreted as device number 0.

Options

video_size
Set the video frame size.

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

x11grab
X11 video input device.

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

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

select_region
Specify whether to select the grabbing area graphically using the
pointer. A value of 1 prompts the user to select the grabbing area
graphically by clicking and dragging. A single click with no
dragging will select the whole screen. A region with zero width or
height will also select the whole screen. This option overwrites
the video_size, grab_x, and grab_y options. Default value is 0.

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

window_id
Grab this window, instead of the whole screen. Default value is 0,
which maps to the whole screen (root window).

The id of a window can be found using the xwininfo program,
possibly with options -tree and -root.

If the window is later enlarged, the new area is not recorded.
Video ends when the window is closed, unmapped (i.e., iconified) or
shrunk beyond the video size (which defaults to the initial window
size).

This option disables options follow_mouse and select_region.

video_size
Set the video frame size. Default is the full desktop or window.

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

• Play a file on default ALSA device:

ffmpeg -i INPUT -f alsa default

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

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

AudioToolbox
AudioToolbox output device.

Allows native output to CoreAudio devices on OSX.

The output filename can be empty (or "-") to refer to the default
system output device or a number that refers to the device index as
shown using: "-list_devices true".

Alternatively, the audio input device can be chosen by index using the

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, UIDs and corresponding indices.

Options

AudioToolbox supports the following options:

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

Examples

• Print the list of supported devices and output a sine wave to the
default device:

$ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -list_devices true -

• Output a sine wave to the device with the index 2, overriding any
output filename:

$ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -audio_device_index 2 -

caca
CACA output device.

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

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

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

Options

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

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

driver
Set display driver.

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

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

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

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

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

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

Examples

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

ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -

• Show the list of available drivers and exit:

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

• Show the list of available dither colors and exit:

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

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

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

DeckLink is very picky about the formats it supports. Pixel format is
always uyvy422, framerate, 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. This option is deprecated, please use the "-sinks" option of
ffmpeg to list the available output devices.

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.

duplex_mode
Sets the decklink device duplex/profile mode. Must be unset, half,
full, one_sub_device_full, one_sub_device_half,
two_sub_device_full, four_sub_device_half Defaults to unset.

Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
one_sub_device_full, one_sub_device_half, two_sub_device_full,
four_sub_device_half

Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2: half,
full

timing_offset
Sets the genlock timing pixel offset on the used output. Defaults
to unset.

link
Sets the SDI video link configuration on the used output. Must be
unset, single link SDI, dual link SDI or quad link SDI. Defaults
to unset.

sqd Enable Square Division Quad Split mode for Quad-link SDI output.
Must be unset, true or false. Defaults to unset.

level_a
Enable SMPTE Level A mode on the used output. Must be unset, true
or false. Defaults to unset.

vanc_queue_size
Sets maximum output buffer size in bytes for VANC data. If the
buffering reaches this value, outgoing VANC data will be dropped.
Defaults to 1048576.

Examples

• List output devices:

ffmpeg -sinks decklink

• List supported formats:

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

• Play video clip:

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

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

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

fbdev
Linux framebuffer output device.

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

Options

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

Examples

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

ffmpeg -re -i INPUT -c:v rawvideo -pix_fmt bgra -f fbdev /dev/fb0

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

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"

sndio
sndio audio output device.

v4l2
Video4Linux2 output device.

xv
XV (XVideo) output device.

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

Options

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

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

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

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

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

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

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

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

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

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

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

Examples

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

ffmpeg -i INPUT OUTPUT -f xv display

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

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

RESAMPLER OPTIONS
The audio resampler supports the following named options.

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

uchl, used_chlayout
Set used input channel layout. Default is unset. 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.

ichl, in_chlayout
ochl, out_chlayout
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,
with a few API-only exceptions noted below. 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. The default width (alpha)
is 3 and can be changed by setting "param0".

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 (API only)
Set source width.

srch (API only)
Set source height.

dstw (API only)
Set destination width.

dsth (API only)
Set destination height.

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

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

src_range (boolean)
If value is set to 1, indicates source is full range. Default value
is 0, which indicates source is limited range.

dst_range (boolean)
If value is set to 1, enable full range for destination. Default
value is 0, which enables limited range.

gamma (boolean)
If value is set to 1, enable gamma correct scaling. Default value
is 0.

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:

• A ':'-separated list of key=value pairs.

• A ':'-separated list of value. In this case, the keys are assumed
to be the option names in the order they are declared. E.g. the
"fade" filter declares three options in this order -- type,
start_frame and nb_frames. Then the parameter list in:0:30 means
that the value in is assigned to the option type, 0 to start_frame
and 30 to nb_frames.

• A ':'-separated list of mixed direct value and long key=value
pairs. The direct value must precede the key=value pairs, and
follow the same constraints order of the previous point. The
following key=value pairs can be set in any preferred order.

If the option value itself is a list of items (e.g. the "format" filter
takes a list of pixel formats), the items in the list are usually
separated by |.

The list of arguments can be quoted using the character ' as initial
and ending mark, and the character \ for escaping the characters within
the quoted text; otherwise the argument string is considered terminated
when the next special character (belonging to the set []=;,) is
encountered.

A special syntax implemented in the ffmpeg CLI tool allows loading
option values from files. This is done be prepending a slash '/' to the
option name, then the supplied value is interpreted as a path from
which the actual value is loaded. E.g.

ffmpeg -i <INPUT> -vf drawtext=/text=/tmp/some_text <OUTPUT>

will load the text to be drawn from /tmp/some_text. API users wishing
to implement a similar feature should use the
"avfilter_graph_segment_*()" functions together with custom IO code.

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.

Leading and trailing whitespaces (space, tabs, or line feeds)
separating tokens in the filtergraph specification are ignored. This
means that the filtergraph can be expressed using empty lines and
spaces to improve readability.

For example, the filtergraph:

testsrc,split[L1],hflip[L2];[L1][L2] hstack

can be represented as:

testsrc,
split [L1], hflip [L2];

[L1][L2] hstack

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"

In order to avoid cumbersome escaping when using a commandline tool
accepting a filter specification as input, it is advisable to avoid
direct inclusion of the filter or options specification in the shell.

For example, in case of the drawtext filter, you might prefer to use
the textfile option in place of text to specify the text to render.

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;
deprecated, do not use

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.

CHANGING OPTIONS AT RUNTIME WITH A COMMAND
Some options can be changed during the operation of the filter using a
command. These options are marked 'T' on the output of ffmpeg -h
filter=<name of filter>. The name of the command is the name of the
option and the argument is the new value.

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.

ts_sync_mode
How strictly to sync streams based on secondary input timestamps;
it accepts one of the following values:

default
Frame from secondary input with the nearest lower or equal
timestamp to the primary input frame.

nearest
Frame from secondary input with the absolute nearest timestamp
to the primary input frame.

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.

aap
Apply Affine Projection algorithm to the first audio stream using the
second audio stream.

This adaptive filter is used to estimate unknown audio based on
multiple input audio samples. Affine projection algorithm can make
trade-offs between computation complexity with convergence speed.

A description of the accepted options follows.

order
Set the filter order.

projection
Set the projection order.

mu Set the filter mu.

delta
Set the coefficient to initialize internal covariance matrix.

out_mode
Set the filter output samples. It accepts the following values:

i Pass the 1st input.

d Pass the 2nd input.

o Pass difference between desired, 2nd input and error signal
estimate.

n Pass difference between input, 1st input and error signal
estimate.

e Pass error signal estimated samples.

Default value is o.

precision
Set which precision to use when processing samples.

auto
Auto pick internal sample format depending on other filters.

float
Always use single-floating point precision sample format.

double
Always use double-floating point precision sample format.

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.

mode
Set mode of compressor operation. Can be "upward" or "downward".
Default is "downward".

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.

Commands

This filter supports the all above options as commands.

acontrast
Simple audio dynamic range compression/expansion filter.

The filter accepts the following options:

contrast
Set contrast. Default is 33. Allowed range is between 0 and 100.

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

• Cross fade from one input to another:

ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac

• Cross fade from one input to another but without overlapping:

ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac

acrossover
Split audio stream into several bands.

This filter splits audio stream into two or more frequency ranges.
Summing all streams back will give flat output.

The filter accepts the following options:

split
Set split frequencies. Those must be positive and increasing.

order
Set filter order for each band split. This controls filter roll-off
or steepness of filter transfer function. Available values are:

2nd 12 dB per octave.

4th 24 dB per octave.

6th 36 dB per octave.

8th 48 dB per octave.

10th
60 dB per octave.

12th
72 dB per octave.

14th
84 dB per octave.

16th
96 dB per octave.

18th
108 dB per octave.

20th
120 dB per octave.

Default is 4th.

level
Set input gain level. Allowed range is from 0 to 1. Default value
is 1.

gains
Set output gain for each band. Default value is 1 for all bands.

precision
Set which precision to use when processing samples.

auto
Auto pick internal sample format depending on other filters.

float
Always use single-floating point precision sample format.

double
Always use double-floating point precision sample format.

Default value is "auto".

Examples

• Split input audio stream into two bands (low and high) with split
frequency of 1500 Hz, each band will be in separate stream:

ffmpeg -i in.flac -filter_complex 'acrossover=split=1500[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav

• Same as above, but with higher filter order:

ffmpeg -i in.flac -filter_complex 'acrossover=split=1500:order=8th[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav

• Same as above, but also with additional middle band (frequencies
between 1500 and 8000):

ffmpeg -i in.flac -filter_complex 'acrossover=split=1500 8000:order=8th[LOW][MID][HIGH]' -map '[LOW]' low.wav -map '[MID]' mid.wav -map '[HIGH]' high.wav

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, 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.

Commands

This filter supports the all above options as commands.

acue
Delay audio filtering until a given wallclock timestamp. See the cue
filter.

adeclick
Remove impulsive noise from input audio.

Samples detected as impulsive noise are replaced by interpolated
samples using autoregressive modelling.

window, w
Set window size, in milliseconds. Allowed range is from 10 to 100.
Default value is 55 milliseconds. This sets size of window which
will be processed at once.

overlap, o
Set window overlap, in percentage of window size. Allowed range is
from 50 to 95. Default value is 75 percent. Setting this to a very
high value increases impulsive noise removal but makes whole
process much slower.

arorder, a
Set autoregression order, in percentage of window size. Allowed
range is from 0 to 25. Default value is 2 percent. This option also
controls quality of interpolated samples using neighbour good
samples.

threshold, t
Set threshold value. Allowed range is from 1 to 100. Default value
is 2. This controls the strength of impulsive noise which is going
to be removed. The lower value, the more samples will be detected
as impulsive noise.

burst, b
Set burst fusion, in percentage of window size. Allowed range is 0
to 10. Default value is 2. If any two samples detected as noise
are spaced less than this value then any sample between those two
samples will be also detected as noise.

method, m
Set overlap method.

It accepts the following values:

add, a
Select overlap-add method. Even not interpolated samples are
slightly changed with this method.

save, s
Select overlap-save method. Not interpolated samples remain
unchanged.

Default value is "a".

adeclip
Remove clipped samples from input audio.

Samples detected as clipped are replaced by interpolated samples using
autoregressive modelling.

window, w
Set window size, in milliseconds. Allowed range is from 10 to 100.
Default value is 55 milliseconds. This sets size of window which
will be processed at once.

overlap, o
Set window overlap, in percentage of window size. Allowed range is
from 50 to 95. Default value is 75 percent.

arorder, a
Set autoregression order, in percentage of window size. Allowed
range is from 0 to 25. Default value is 8 percent. This option also
controls quality of interpolated samples using neighbour good
samples.

threshold, t
Set threshold value. Allowed range is from 1 to 100. Default value
is 10. Higher values make clip detection less aggressive.

hsize, n
Set size of histogram used to detect clips. Allowed range is from
100 to 9999. Default value is 1000. Higher values make clip
detection less aggressive.

method, m
Set overlap method.

It accepts the following values:

add, a
Select overlap-add method. Even not interpolated samples are
slightly changed with this method.

save, s
Select overlap-save method. Not interpolated samples remain
unchanged.

Default value is "a".

adecorrelate
Apply decorrelation to input audio stream.

The filter accepts the following options:

stages
Set decorrelation stages of filtering. Allowed range is from 1 to
16. Default value is 6.

seed
Set random seed used for setting delay in samples across channels.

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. If you want instead to delay in seconds,
append 's' to number.

all Use last set delay for all remaining channels. By default is
disabled. This option if enabled changes how option "delays" is
interpreted.

Examples

• Delay first channel by 1.5 seconds, the third channel by 0.5
seconds and leave the second channel (and any other channels that
may be present) unchanged.

adelay=1500|0|500

• 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

• Delay all channels by same number of samples:

adelay=delays=64S:all=1

adenorm
Remedy denormals in audio by adding extremely low-level noise.

This filter shall be placed before any filter that can produce
denormals.

A description of the accepted parameters follows.

level
Set level of added noise in dB. Default is -351. Allowed range is
from -451 to -90.

type
Set type of added noise.

dc Add DC signal.

ac Add AC signal.

square
Add square signal.

pulse
Add pulse signal.

Default is "dc".

Commands

This filter supports the all above options as commands.

aderivative, aintegral
Compute derivative/integral of audio stream.

Applying both filters one after another produces original audio.

adrc
Apply spectral dynamic range controller filter to input audio stream.

A description of the accepted options follows.

transfer
Set the transfer expression.

The expression can contain the following constants:

ch current channel number

sn current sample number

nb_channels
number of channels

t timestamp expressed in seconds

sr sample rate

p current frequency power value, in dB

f current frequency in Hz

Default value is "p".

attack
Set the attack in milliseconds. Default is 50 milliseconds.
Allowed range is from 1 to 1000 milliseconds.

release
Set the release in milliseconds. Default is 100 milliseconds.
Allowed range is from 5 to 2000 milliseconds.

channels
Set which channels to filter, by default "all" channels in audio
stream are filtered.

Commands

This filter supports the all above options as commands.

Examples

• Apply spectral compression to all frequencies with threshold of -50
dB and 1:6 ratio:

adrc=transfer='if(gt(p,-50),-50+(p-(-50))/6,p)':attack=50:release=100

• Similar to above but with 1:2 ratio and filtering only front center
channel:

adrc=transfer='if(gt(p,-50),-50+(p-(-50))/2,p)':attack=50:release=100:channels=FC

• Apply spectral noise gate to all frequencies with threshold of -85
dB and with short attack time and short release time:

adrc=transfer='if(lte(p,-85),p-800,p)':attack=1:release=5

• Apply spectral expansion to all frequencies with threshold of -10
dB and 1:2 ratio:

adrc=transfer='if(lt(p,-10),-10+(p-(-10))*2,p)':attack=50:release=100

• Apply limiter to max -60 dB to all frequencies, with attack of 2 ms
and release of 10 ms:

adrc=transfer='min(p,-60)':attack=2:release=10

adynamicequalizer
Apply dynamic equalization to input audio stream.

A description of the accepted options follows.

threshold
Set the detection threshold used to trigger equalization.
Threshold detection is using detection filter. Default value is 0.
Allowed range is from 0 to 100.

dfrequency
Set the detection frequency in Hz used for detection filter used to
trigger equalization. Default value is 1000 Hz. Allowed range is
between 2 and 1000000 Hz.

dqfactor
Set the detection resonance factor for detection filter used to
trigger equalization. Default value is 1. Allowed range is from
0.001 to 1000.

tfrequency
Set the target frequency of equalization filter. Default value is
1000 Hz. Allowed range is between 2 and 1000000 Hz.

tqfactor
Set the target resonance factor for target equalization filter.
Default value is 1. Allowed range is from 0.001 to 1000.

attack
Set the amount of milliseconds the signal from detection has to
rise above the detection threshold before equalization starts.
Default is 20. Allowed range is between 1 and 2000.

release
Set the amount of milliseconds the signal from detection has to
fall below the detection threshold before equalization ends.
Default is 200. Allowed range is between 1 and 2000.

ratio
Set the ratio by which the equalization gain is raised. Default is
1. Allowed range is between 0 and 30.

makeup
Set the makeup offset by which the equalization gain is raised.
Default is 0. Allowed range is between 0 and 100.

range
Set the max allowed cut/boost amount. Default is 50. Allowed range
is from 1 to 200.

mode
Set the mode of filter operation, can be one of the following:

listen
Output only isolated detection signal.

cutbelow
Cut frequencies below detection threshold.

cutabove
Cut frequencies above detection threshold.

boostbelow
Boost frequencies below detection threshold.

boostabove
Boost frequencies above detection threshold.

Default mode is cutbelow.

dftype
Set the type of detection filter, can be one of the following:

bandpass
lowpass
highpass
peak

Default type is bandpass.

tftype
Set the type of target filter, can be one of the following:

bell
lowshelf
highshelf

Default type is bell.

auto
Automatically gather threshold from detection filter. By default is
disabled. This option is useful to detect threshold in certain
time frame of input audio stream, in such case option value is
changed at runtime.

Available values are:

disabled
Disable using automatically gathered threshold value.

off Stop picking threshold value.

on Start picking threshold value.

adaptive
Adaptively pick threshold value, by calculating sliding window
entropy.

precision
Set which precision to use when processing samples.

auto
Auto pick internal sample format depending on other filters.

float
Always use single-floating point precision sample format.

double
Always use double-floating point precision sample format.

Commands

This filter supports the all above options as commands.

adynamicsmooth
Apply dynamic smoothing to input audio stream.

A description of the accepted options follows.

sensitivity
Set an amount of sensitivity to frequency fluctations. Default is
2. Allowed range is from 0 to 1e+06.

basefreq
Set a base frequency for smoothing. Default value is 22050.
Allowed range is from 2 to 1e+06.

Commands

This filter supports the all above options as commands.

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

• Make it sound as if there are twice as many instruments as are
actually playing:

aecho=0.8:0.88:60:0.4

• If delay is very short, then it sounds like a (metallic) robot
playing music:

aecho=0.8:0.88:6:0.4

• A longer delay will sound like an open air concert in the
mountains:

aecho=0.8:0.9:1000:0.3

• Same as above but with one more mountain:

aecho=0.8:0.9:1000|1800:0.3|0.25

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 50s (FM).

75fm
select 75s (FM).

50kf
select 50s (FM-KF).

75kf
select 75s (FM-KF).

Commands

This filter supports the all above options as commands.

aeval
Modify an audio signal according to the specified expressions.

This filter accepts one or more expressions (one for each channel),
which are evaluated and used to modify a corresponding audio signal.

It accepts the following parameters:

exprs
Set the '|'-separated expressions list for each separate channel.
If the number of input channels is greater than the number of
expressions, the last specified expression is used for the
remaining output channels.

channel_layout, c
Set output channel layout. If not specified, the channel layout is
specified by the number of expressions. If set to same, it will use
by default the same input channel layout.

Each expression in exprs can contain the following constants and
functions:

ch channel number of the current expression

n number of the evaluated sample, starting from 0

s sample rate

t time of the evaluated sample expressed in seconds

nb_in_channels
nb_out_channels
input and output number of channels

val(CH)
the value of input channel with number CH

Note: this filter is slow. For faster processing you should use a
dedicated filter.

Examples

• Half volume:

aeval=val(ch)/2:c=same

• Invert phase of the second channel:

aeval=val(0)|-val(1)

aexciter
An exciter is used to produce high sound that is not present in the
original signal. This is done by creating harmonic distortions of the
signal which are restricted in range and added to the original signal.
An Exciter raises the upper end of an audio signal without simply
raising the higher frequencies like an equalizer would do to create a
more "crisp" or "brilliant" sound.

The filter accepts the following options:

level_in
Set input level prior processing of signal. Allowed range is from
0 to 64. Default value is 1.

level_out
Set output level after processing of signal. Allowed range is from
0 to 64. Default value is 1.

amount
Set the amount of harmonics added to original signal. Allowed
range is from 0 to 64. Default value is 1.

drive
Set the amount of newly created harmonics. Allowed range is from
0.1 to 10. Default value is 8.5.

blend
Set the octave of newly created harmonics. Allowed range is from
-10 to 10. Default value is 0.

freq
Set the lower frequency limit of producing harmonics in Hz.
Allowed range is from 2000 to 12000 Hz. Default is 7500 Hz.

ceil
Set the upper frequency limit of producing harmonics. Allowed
range is from 9999 to 20000 Hz. If value is lower than 10000 Hz no
limit is applied.

listen
Mute the original signal and output only added harmonics. By
default is disabled.

Commands

This filter supports the all above options as commands.

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

losi
select logistic sigmoid

sinc
select sine cardinal function

isinc
select inverted sine cardinal function

quat
select quartic

quatr
select quartic root

qsin2
select squared quarter of sine wave

hsin2
select squared half of sine wave

nofade
no fade applied

silence
Set the initial gain for fade-in or final gain for fade-out.
Default value is 0.0.

unity
Set the initial gain for fade-out or final gain for fade-in.
Default value is 1.0.

Commands

This filter supports the all above options as commands.

Examples

• Fade in first 15 seconds of audio:

afade=t=in:ss=0:d=15

• Fade out last 25 seconds of a 900 seconds audio:

afade=t=out:st=875:d=25

afftdn
Denoise audio samples with FFT.

A description of the accepted parameters follows.

noise_reduction, nr
Set the noise reduction in dB, allowed range is 0.01 to 97.
Default value is 12 dB.

noise_floor, nf
Set the noise floor in dB, allowed range is -80 to -20. Default
value is -50 dB.

noise_type, nt
Set the noise type.

It accepts the following values:

white, w
Select white noise.

vinyl, v
Select vinyl noise.

shellac, s
Select shellac noise.

custom, c
Select custom noise, defined in "bn" option.

Default value is white noise.

band_noise, bn
Set custom band noise profile for every one of 15 bands. Bands are
separated by ' ' or '|'.

residual_floor, rf
Set the residual floor in dB, allowed range is -80 to -20. Default
value is -38 dB.

track_noise, tn
Enable noise floor tracking. By default is disabled. With this
enabled, noise floor is automatically adjusted.

track_residual, tr
Enable residual tracking. By default is disabled.

output_mode, om
Set the output mode.

It accepts the following values:

input, i
Pass input unchanged.

output, o
Pass noise filtered out.

noise, n
Pass only noise.

Default value is output.

adaptivity, ad
Set the adaptivity factor, used how fast to adapt gains adjustments
per each frequency bin. Value 0 enables instant adaptation, while
higher values react much slower. Allowed range is from 0 to 1.
Default value is 0.5.

floor_offset, fo
Set the noise floor offset factor. This option is used to adjust
offset applied to measured noise floor. It is only effective when
noise floor tracking is enabled. Allowed range is from -2.0 to
2.0. Default value is 1.0.

noise_link, nl
Set the noise link used for multichannel audio.

It accepts the following values:

none
Use unchanged channel's noise floor.

min Use measured min noise floor of all channels.

max Use measured max noise floor of all channels.

average
Use measured average noise floor of all channels.

Default value is min.

band_multiplier, bm
Set the band multiplier factor, used how much to spread bands
across frequency bins. Allowed range is from 0.2 to 5. Default
value is 1.25.

sample_noise, sn
Toggle capturing and measurement of noise profile from input audio.

It accepts the following values:

start, begin
Start sample noise capture.

stop, end
Stop sample noise capture and measure new noise band profile.

Default value is "none".

gain_smooth, gs
Set gain smooth spatial radius, used to smooth gains applied to
each frequency bin. Useful to reduce random music noise artefacts.
Higher values increases smoothing of gains. Allowed range is from
0 to 50. Default value is 0.

Commands

This filter supports the some above mentioned options as commands.

Examples

• Reduce white noise by 10dB, and use previously measured noise floor
of -40dB:

afftdn=nr=10:nf=-40

• Reduce white noise by 10dB, also set initial noise floor to -80dB
and enable automatic tracking of noise floor so noise floor will
gradually change during processing:

afftdn=nr=10:nf=-80:tn=1

• Reduce noise by 20dB, using noise floor of -40dB and using commands
to take noise profile of first 0.4 seconds of input audio:

asendcmd=0.0 afftdn sn start,asendcmd=0.4 afftdn sn stop,afftdn=nr=20:nf=-40

afftfilt
Apply arbitrary expressions to samples in frequency domain.

real
Set frequency domain real expression for each separate channel
separated by '|'. Default is "re". 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 '|'. Default is "im".

Each expression in real and imag can contain the following
constants and functions:

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

re current real part of frequency bin of current channel

im current imaginary part of frequency bin of current channel

real(b, ch)
Return the value of real part of frequency bin at location
(bin,channel)

imag(b, ch)
Return the value of imaginary part of frequency bin at location
(bin,channel)

win_size
Set window size. Allowed range is from 16 to 131072. Default is
4096

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
bohman
kaiser

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

• Leave almost only low frequencies in audio:

afftfilt="'real=re * (1-clip((b/nb)*b,0,1))':imag='im * (1-clip((b/nb)*b,0,1))'"

• Apply robotize effect:

afftfilt="real='hypot(re,im)*sin(0)':imag='hypot(re,im)*cos(0)':win_size=512:overlap=0.75"

• Apply whisper effect:

afftfilt="real='hypot(re,im)*cos((random(0)*2-1)*2*3.14)':imag='hypot(re,im)*sin((random(1)*2-1)*2*3.14)':win_size=128:overlap=0.8"

• Apply phase shift:

afftfilt="real=re*cos(1)-im*sin(1):imag=re*sin(1)+im*cos(1)"

afir
Apply an arbitrary Finite Impulse Response filter.

This filter is designed for applying long FIR filters, up to 60 seconds
long.

It can be used as component for digital crossover filters, room
equalization, cross talk cancellation, wavefield synthesis,
auralization, ambiophonics, ambisonics and spatialization.

This filter uses the streams higher than first one as FIR coefficients.
If the non-first stream holds a single channel, it will be used for all
input channels in the first stream, otherwise the number of channels in
the non-first stream must be same as the number of channels in the
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.

gtype
This option is deprecated, and does nothing.

irnorm
Set norm to be applied to IR coefficients before filtering.
Allowed range is from -1 to 2. IR coefficients are normalized with
calculated vector norm set by this option. For negative values, no
norm is calculated, and IR coefficients are not modified at all.
Default is 1.

irlink
For multichannel IR if this option is set to true, all IR channels
will be normalized with maximal measured gain of all IR channels
coefficients as set by "irnorm" option. When disabled, all IR
coefficients in each IR channel will be normalized independently.
Default is true.

irgain
Set gain to be applied to IR coefficients before filtering.
Allowed range is 0 to 1. This gain is applied after any gain
applied with irnorm option.

irfmt
Set format of IR stream. Can be "mono" or "input". Default is
"input".

maxir
Set max allowed Impulse Response filter duration in seconds.
Default is 30 seconds. Allowed range is 0.1 to 60 seconds.

response
This option is deprecated, and does nothing.

channel
This option is deprecated, and does nothing.

size
This option is deprecated, and does nothing.

rate
This option is deprecated, and does nothing.

minp
Set minimal partition size used for convolution. Default is 8192.
Allowed range is from 1 to 65536. Lower values decreases latency
at cost of higher CPU usage.

maxp
Set maximal partition size used for convolution. Default is 8192.
Allowed range is from 8 to 65536. Lower values may increase CPU
usage.

nbirs
Set number of input impulse responses streams which will be
switchable at runtime. Allowed range is from 1 to 32. Default is
1.

ir Set IR stream which will be used for convolution, starting from 0,
should always be lower than supplied value by "nbirs" option.
Default is 0. This option can be changed at runtime via commands.

precision
Set which precision to use when processing samples.

auto
Auto pick internal sample format depending on other filters.

float
Always use single-floating point precision sample format.

double
Always use double-floating point precision sample format.

Default value is auto.

irload
Set when to load IR stream. Can be "init" or "access". First one
load and prepares all IRs on initialization, second one once on
first access of specific IR. Default is "init".

Examples

• 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

• Apply true stereo processing given input stereo stream, and two
stereo impulse responses for left and right channel, the impulse
response files are files with names l_ir.wav and r_ir.wav, and
setting irnorm option value:

"pan=4C|c0=FL|c1=FL|c2=FR|c3=FR[a];amovie=l_ir.wav[LIR];amovie=r_ir.wav[RIR];[LIR][RIR]amerge[ir];[a][ir]afir=irfmt=input:irnorm=1.2,pan=stereo|FL<c0+c2|FR<c1+c3"

• Similar to above example, but with "irgain" explicitly set to
estimated value and with "irnorm" disabled:

"pan=4C|c0=FL|c1=FL|c2=FR|c3=FR[a];amovie=l_ir.wav[LIR];amovie=r_ir.wav[RIR];[LIR][RIR]amerge[ir];[a][ir]afir=irfmt=input:irgain=-5dB:irnom=-1,pan=stereo|FL<c0+c2|FR<c1+c3"

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, f
A '|'-separated list of requested sample formats.

sample_rates, r
A '|'-separated list of requested sample rates.

channel_layouts, cl
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

afreqshift
Apply frequency shift to input audio samples.

The filter accepts the following options:

shift
Specify frequency shift. Allowed range is -INT_MAX to INT_MAX.
Default value is 0.0.

level
Set output gain applied to final output. Allowed range is from 0.0
to 1.0. Default value is 1.0.

order
Set filter order used for filtering. Allowed range is from 1 to 16.
Default value is 8.

Commands

This filter supports the all above options as commands.

afwtdn
Reduce broadband noise from input samples using Wavelets.

A description of the accepted options follows.

sigma
Set the noise sigma, allowed range is from 0 to 1. Default value
is 0. This option controls strength of denoising applied to input
samples. Most useful way to set this option is via decibels, eg.
-45dB.

levels
Set the number of wavelet levels of decomposition. Allowed range
is from 1 to 12. Default value is 10. Setting this too low make
denoising performance very poor.

wavet
Set wavelet type for decomposition of input frame. They are sorted
by number of coefficients, from lowest to highest. More
coefficients means worse filtering speed, but overall better
quality. Available wavelets are:

sym2
sym4
rbior68
deb10
sym10
coif5
bl3

percent
Set percent of full denoising. Allowed range is from 0 to 100
percent. Default value is 85 percent or partial denoising.

profile
If enabled, first input frame will be used as noise profile. If
first frame samples contain non-noise performance will be very
poor.

adaptive
If enabled, input frames are analyzed for presence of noise. If
noise is detected with high possibility then input frame profile
will be used for processing following frames, until new noise frame
is detected.

samples
Set size of single frame in number of samples. Allowed range is
from 512 to 65536. Default frame size is 8192 samples.

softness
Set softness applied inside thresholding function. Allowed range is
from 0 to 10. Default softness is 1.

Commands

This filter supports the all above options as commands.

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.

mode
Set the mode of operation. Can be "upward" or "downward". Default
is "downward". If set to "upward" mode, higher parts of signal will
be amplified, expanding dynamic range in upward direction.
Otherwise, in case of "downward" lower parts of signal will be
reduced.

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.
Setting this to 0 disables reduction and then filter behaves like
expander.

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".

Commands

This filter supports the all above options as commands.

aiir
Apply an arbitrary Infinite Impulse Response filter.

It accepts the following parameters:

zeros, z
Set B/numerator/zeros/reflection coefficients.

poles, p
Set A/denominator/poles/ladder coefficients.

gains, k
Set channels gains.

dry_gain
Set input gain.

wet_gain
Set output gain.

format, f
Set coefficients format.

ll lattice-ladder function

sf analog transfer function

tf digital transfer function

zp Z-plane zeros/poles, cartesian (default)

pr Z-plane zeros/poles, polar radians

pd Z-plane zeros/poles, polar degrees

sp S-plane zeros/poles

process, r
Set type of processing.

d direct processing

s serial processing

p parallel processing

precision, e
Set filtering precision.

dbl double-precision floating-point (default)

flt single-precision floating-point

i32 32-bit integers

i16 16-bit integers

normalize, n
Normalize filter coefficients, by default is enabled. Enabling it
will normalize magnitude response at DC to 0dB.

mix How much to use filtered signal in output. Default is 1. Range is
between 0 and 1.

response
Show IR frequency response, magnitude(magenta), phase(green) and
group delay(yellow) in additional video stream. By default it is
disabled.

channel
Set for which IR channel to display frequency response. By default
is first channel displayed. This option is used only when response
is enabled.

size
Set video stream size. This option is used only when response is
enabled.

Coefficients in "tf" and "sf" format are separated by spaces and are in
ascending order.

Coefficients in "zp" format are separated by spaces and order of
coefficients doesn't matter. Coefficients in "zp" format are complex
numbers with i imaginary unit.

Different coefficients and gains can be provided for every channel, in
such case use '|' to separate coefficients or gains. Last provided
coefficients will be used for all remaining channels.

Examples

• Apply 2 pole elliptic notch at around 5000Hz for 48000 Hz sample
rate:

aiir=k=1:z=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:p=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf:r=d

• Same as above but in "zp" format:

aiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp:r=s

• Apply 3-rd order analog normalized Butterworth low-pass filter,
using analog transfer function format:

aiir=z=1.3057 0 0 0:p=1.3057 2.3892 2.1860 1:f=sf:r=d

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.

latency
Compensate the delay introduced by using the lookahead buffer set
with attack parameter. Also flush the valid audio data in the
lookahead buffer when the stream hits EOF.

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

k kHz

width, w
Specify the band-width of a filter in width_type units.

mix, m
How much to use filtered signal in output. Default is 1. Range is
between 0 and 1.

channels, c
Specify which channels to filter, by default all available are
filtered.

normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it
will normalize magnitude response at DC to 0dB.

order, o
Set the filter order, can be 1 or 2. Default is 2.

transform, a
Set transform type of IIR filter.

di
dii
tdi
tdii
latt
svf
zdf

precision, r
Set precision of filtering.

auto
Pick automatic sample format depending on surround filters.

s16 Always use signed 16-bit.

s32 Always use signed 32-bit.

f32 Always use float 32-bit.

f64 Always use float 64-bit.

Commands

This filter supports the following commands:

frequency, f
Change allpass frequency. Syntax for the command is : "frequency"

width_type, t
Change allpass width_type. Syntax for the command is :
"width_type"

width, w
Change allpass width. Syntax for the command is : "width"

mix, m
Change allpass mix. Syntax for the command is : "mix"

aloop
Loop audio samples.

The filter accepts the following options:

loop
Set the number of loops. Setting this value to -1 will result in
infinite loops. Default is 0.

size
Set maximal number of samples. Default is 0.

start
Set first sample of loop. Default is 0.

time
Set the time of loop start in seconds. Only used if option named
start is set to -1.

amerge
Merge two or more audio streams into a single multi-channel stream.

The filter accepts the following options:

inputs
Set the number of inputs. Default is 2.

If the channel layouts of the inputs are disjoint, and therefore
compatible, the channel layout of the output will be set accordingly
and the channels will be reordered as necessary. If the channel layouts
of the inputs are not disjoint, the output will have all the channels
of the first input then all the channels of the second input, in that
order, and the channel layout of the output will be the default value
corresponding to the total number of channels.

For example, if the first input is in 2.1 (FL+FR+LF) and the second
input is FC+BL+BR, then the output will be in 5.1, with the channels in
the following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of
the first input, b1 is the first channel of the second input).

On the other hand, if both input are in stereo, the output channels
will be in the default order: a1, a2, b1, b2, and the channel layout
will be arbitrarily set to 4.0, which may or may not be the expected
value.

All inputs must have the same sample rate, and format.

If inputs do not have the same duration, the output will stop with the
shortest.

Examples

• Merge two mono files into a stereo stream:

amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge

• Multiple merges assuming 1 video stream and 6 audio streams in
input.mkv:

ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv

amix
Mixes multiple audio inputs into a single output.

Note that this filter only supports float samples (the amerge and pan
audio filters support many formats). If the amix input has integer
samples then aresample will be automatically inserted to perform the
conversion to float samples.

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.

weights
Specify weight of each input audio stream as a sequence of numbers
separated by a space. If fewer weights are specified compared to
number of inputs, the last weight is assigned to the remaining
inputs. Default weight for each input is 1.

normalize
Always scale inputs instead of only doing summation of samples.
Beware of heavy clipping if inputs are not normalized prior or
after filtering by this filter if this option is disabled. By
default is enabled.

Examples

• This 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:

ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT

• This will mix one vocal and one music input audio stream to a
single output with the same duration as the longest input. The
music will have quarter the weight as the vocals, and the inputs
are not normalized:

ffmpeg -i VOCALS -i MUSIC -filter_complex amix=inputs=2:duration=longest:dropout_transition=0:weights="1 0.25":normalize=0 OUTPUT

Commands

This filter supports the following commands:

weights
normalize
Syntax is same as option with same name.

amultiply
Multiply first audio stream with second audio stream and store result
in output audio stream. Multiplication is done by multiplying each
sample from first stream with sample at same position from second
stream.

With this element-wise multiplication one can create amplitude fades
and amplitude modulations.

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

• 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 Hertz. 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=...

anlmdn
Reduce broadband noise in audio samples using Non-Local Means
algorithm.

Each sample is adjusted by looking for other samples with similar
contexts. This context similarity is defined by comparing their
surrounding patches of size p. Patches are searched in an area of r
around the sample.

The filter accepts the following options:

strength, s
Set denoising strength. Allowed range is from 0.00001 to 10000.
Default value is 0.00001.

patch, p
Set patch radius duration. Allowed range is from 1 to 100
milliseconds. Default value is 2 milliseconds.

research, r
Set research radius duration. Allowed range is from 2 to 300
milliseconds. Default value is 6 milliseconds.

output, o
Set the output mode.

It accepts the following values:

i Pass input unchanged.

o Pass noise filtered out.

n Pass only noise.

Default value is o.

smooth, m
Set smooth factor. Default value is 11. Allowed range is from 1 to
1000.

Commands

This filter supports the all above options as commands.

anlmf, anlms
Apply Normalized Least-Mean-(Squares|Fourth) algorithm to the first
audio stream using the second audio stream.

This adaptive filter is used to mimic a desired filter by finding the
filter coefficients that relate to producing the least mean square of
the error signal (difference between the desired, 2nd input audio
stream and the actual signal, the 1st input audio stream).

A description of the accepted options follows.

order
Set filter order.

mu Set filter mu.

eps Set the filter eps.

leakage
Set the filter leakage.

out_mode
It accepts the following values:

i Pass the 1st input.

d Pass the 2nd input.

o Pass difference between desired, 2nd input and error signal
estimate.

n Pass difference between input, 1st input and error signal
estimate.

e Pass error signal estimated samples.

Default value is o.

precision
Set which precision to use when processing samples.

auto
Auto pick internal sample format depending on other filters.

float
Always use single-floating point precision sample format.

double
Always use double-floating point precision sample format.

Examples

• One of many usages of this filter is noise reduction, input audio
is filtered with same samples that are delayed by fixed amount, one
such example for stereo audio is:

asplit[a][b],[a]adelay=32S|32S[a],[b][a]anlms=order=128:leakage=0.0005:mu=.5:out_mode=o

Commands

This filter supports the same commands as options, excluding option
"order".

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.

pad_dur
Specify the duration of samples of silence to add. See the Time
duration section in the ffmpeg-utils(1) manual for the accepted
syntax. Used only if set to non-negative value.

whole_dur
Specify the minimum total duration in the output audio stream. See
the Time duration section in the ffmpeg-utils(1) manual for the
accepted syntax. Used only if set to non-negative value. 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_dur

If neither the pad_len nor the whole_len nor pad_dur nor whole_dur
option is set, the filter will add silence to the end of the input
stream indefinitely.

Note that for ffmpeg 4.4 and earlier a zero pad_dur or whole_dur also
caused the filter to add silence indefinitely.

Examples

• Add 1024 samples of silence to the end of the input:

apad=pad_len=1024

• Make sure the audio output will contain at least 10000 samples, pad
the input with silence if required:

apad=whole_len=10000

• Use ffmpeg to pad the audio input with silence, so that the video
stream will always result the shortest and will be converted until
the end in the output file when using the shortest option:

ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT

aphaser
Add a phasing effect to the input audio.

A phaser filter creates series of peaks and troughs in the frequency
spectrum. The position of the peaks and troughs are modulated so that
they vary over time, creating a sweeping effect.

A description of the accepted parameters follows.

in_gain
Set input gain. Default is 0.4.

out_gain
Set output gain. Default is 0.74

delay
Set delay in milliseconds. Default is 3.0.

decay
Set decay. Default is 0.4.

speed
Set modulation speed in Hz. Default is 0.5.

type
Set modulation type. Default is triangular.

It accepts the following values:

triangular, t
sinusoidal, s

aphaseshift
Apply phase shift to input audio samples.

The filter accepts the following options:

shift
Specify phase shift. Allowed range is from -1.0 to 1.0. Default
value is 0.0.

level
Set output gain applied to final output. Allowed range is from 0.0
to 1.0. Default value is 1.0.

order
Set filter order used for filtering. Allowed range is from 1 to 16.
Default value is 8.

Commands

This filter supports the all above options as commands.

apsnr
Measure Audio Peak Signal-to-Noise Ratio.

This filter takes two audio streams for input, and outputs first audio
stream. Results are in dB per channel at end of either input.

apsyclip
Apply Psychoacoustic clipper to input audio stream.

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].

clip
Set the clipping start value. Default value is 0dBFS or 1.

diff
Output only difference samples, useful to hear introduced
distortions. By default is disabled.

adaptive
Set strength of adaptive distortion applied. Default value is 0.5.
Allowed range is from 0 to 1.

iterations
Set number of iterations of psychoacoustic clipper. Allowed range
is from 1 to 20. Default value is 10.

level
Auto level output signal. Default is disabled. This normalizes
audio back to 0dBFS if enabled.

Commands

This filter supports the all above options as commands.

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 "Resampler Options" section
in the ffmpeg-resampler(1) manual for the complete list of supported
options.

Examples

• Resample the input audio to 44100Hz:

aresample=44100

• Stretch/squeeze samples to the given timestamps, with a maximum of
1000 samples per second compensation:

aresample=async=1000

areverse
Reverse an audio clip.

Warning: This filter requires memory to buffer the entire clip, so
trimming is suggested.

Examples

• Take the first 5 seconds of a clip, and reverse it.

atrim=end=5,areverse

arls
Apply Recursive Least Squares algorithm to the first audio stream using
the second audio stream.

This adaptive filter is used to mimic a desired filter by recursively
finding the filter coefficients that relate to producing the minimal
weighted linear least squares cost function of the error signal
(difference between the desired, 2nd input audio stream and the actual
signal, the 1st input audio stream).

A description of the accepted options follows.

order
Set the filter order.

lambda
Set the forgetting factor.

delta
Set the coefficient to initialize internal covariance matrix.

out_mode
Set the filter output samples. It accepts the following values:

i Pass the 1st input.

d Pass the 2nd input.

o Pass difference between desired, 2nd input and error signal
estimate.

n Pass difference between input, 1st input and error signal
estimate.

e Pass error signal estimated samples.

Default value is o.

precision
Set which precision to use when processing samples.

auto
Auto pick internal sample format depending on other filters.

float
Always use single-floating point precision sample format.

double
Always use double-floating point precision sample format.

arnndn
Reduce noise from speech using Recurrent Neural Networks.

This filter accepts the following options:

model, m
Set train model file to load. This option is always required.

mix Set how much to mix filtered samples into final output. Allowed
range is from -1 to 1. Default value is 1. Negative values are
special, they set how much to keep filtered noise in the final
filter output. Set this option to -1 to hear actual noise removed
from input signal.

Commands

This filter supports the all above options as commands.

asdr
Measure Audio Signal-to-Distortion Ratio.

This filter takes two audio streams for input, and outputs first audio
stream. Results are in dB per channel at end of either input.

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.

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.

asisdr
Measure Audio Scaled-Invariant Signal-to-Distortion Ratio.

This filter takes two audio streams for input, and outputs first audio
stream. Results are in dB per channel at end of either input.

asoftclip
Apply audio soft clipping.

Soft clipping is a type of distortion effect where the amplitude of a
signal is saturated along a smooth curve, rather than the abrupt shape
of hard-clipping.

This filter accepts the following options:

type
Set type of soft-clipping.

It accepts the following values:

hard
tanh
atan
cubic
exp
alg
quintic
sin
erf

threshold
Set threshold from where to start clipping. Default value is 0dB or
1.

output
Set gain applied to output. Default value is 0dB or 1.

param
Set additional parameter which controls sigmoid function.

oversample
Set oversampling factor.

Commands

This filter supports the all above options as commands.

aspectralstats
Display frequency domain statistical information about the audio
channels. Statistics are calculated and stored as metadata for each
audio channel and for each audio frame.

It accepts the following option:

win_size
Set the window length in samples. Default value is 2048. Allowed
range is from 32 to 65536.

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
bohman
kaiser

Default is "hann".

overlap
Set window overlap. Allowed range is from 0 to 1. Default value is
0.5.

measure
Select the parameters which are measured. The metadata keys can be
used as flags, default is all which measures everything. none
disables all measurement.

A list of each metadata key follows:

mean
variance
centroid
spread
skewness
kurtosis
entropy
flatness
crest
flux
slope
decrease
rolloff

asr
Automatic Speech Recognition

This filter uses PocketSphinx for speech recognition. To enable
compilation of this filter, you need to configure FFmpeg with
"--enable-pocketsphinx".

It accepts the following options:

rate
Set sampling rate of input audio. Defaults is 16000. This need to
match speech models, otherwise one will get poor results.

hmm Set dictionary containing acoustic model files.

dict
Set pronunciation dictionary.

lm Set language model file.

lmctl
Set language model set.

lmname
Set which language model to use.

logfn
Set output for log messages.

The filter exports recognized speech as the frame metadata
"lavfi.asr.text".

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 through RMS
measurement. Default is 0.05 (50 milliseconds). Allowed range is
"[0 - 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: Bit_depth Crest_factor
DC_offset Dynamic_range Entropy Flat_factor Max_difference
Max_level Mean_difference Min_difference Min_level Noise_floor
Noise_floor_count Number_of_Infs Number_of_NaNs Number_of_denormals
Peak_count Abs_Peak_count Peak_level RMS_difference RMS_peak
RMS_trough Zero_crossings Zero_crossings_rate

and for "Overall": Bit_depth DC_offset Entropy Flat_factor
Max_difference Max_level Mean_difference Min_difference Min_level
Noise_floor Noise_floor_count Number_of_Infs Number_of_NaNs
Number_of_denormals Number_of_samples Peak_count Abs_Peak_count
Peak_level RMS_difference RMS_level RMS_peak RMS_trough

For example, a full key looks like "lavfi.astats.1.DC_offset" or
"lavfi.astats.Overall.Peak_count".

Read below for the description of the keys.

reset
Set the number of frames over which cumulative stats are calculated
before being reset. Default is disabled.

measure_perchannel
Select the parameters which are measured per channel. The metadata
keys can be used as flags, default is all which measures
everything. none disables all per channel measurement.

measure_overall
Select the parameters which are measured overall. The metadata keys
can be used as flags, default is all which measures everything.
none disables all overall measurement.

A description of the measure keys follow:

none
no measures

all all measures

Bit_depth
overall bit depth of audio, i.e. number of bits used for each
sample

Crest_factor
standard ratio of peak to RMS level (note: not in dB)

DC_offset
mean amplitude displacement from zero

Dynamic_range
measured dynamic range of audio in dB

Entropy
entropy measured across whole audio, entropy of value near 1.0 is
typically measured for white noise

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)

Max_difference
maximal difference between two consecutive samples

Max_level
maximal sample level

Mean_difference
mean difference between two consecutive samples, i.e. the average
of each difference between two consecutive samples

Min_difference
minimal difference between two consecutive samples

Min_level
minimal sample level

Noise_floor
minimum local peak measured in dBFS over a short window

Noise_floor_count
number of occasions (not the number of samples) that the signal
attained Noise floor

Number_of_Infs
number of samples with an infinite value

Number_of_NaNs
number of samples with a NaN (not a number) value

Number_of_denormals
number of samples with a subnormal value

Number_of_samples
number of samples

Peak_count
number of occasions (not the number of samples) that the signal
attained either Min_level or Max_level

Abs_Peak_count
number of occasions that the absolute samples taken from the signal
attained max absolute value of Min_level and Max_level

Peak_level
standard peak level measured in dBFS

RMS_difference
Root Mean Square difference between two consecutive samples

RMS_level
standard RMS level measured in dBFS

RMS_peak
RMS_trough
peak and through values for RMS level measured over a short window,
measured in dBFS.

Zero crossings
number of points where the waveform crosses the zero level axis

Zero crossings rate
rate of Zero crossings and number of audio samples

asubboost
Boost subwoofer frequencies.

The filter accepts the following options:

dry Set dry gain, how much of original signal is kept. Allowed range is
from 0 to 1. Default value is 1.0.

wet Set wet gain, how much of filtered signal is kept. Allowed range is
from 0 to 1. Default value is 1.0.

boost
Set max boost factor. Allowed range is from 1 to 12. Default value
is 2.

decay
Set delay line decay gain value. Allowed range is from 0 to 1.
Default value is 0.0.

feedback
Set delay line feedback gain value. Allowed range is from 0 to 1.
Default value is 0.9.

cutoff
Set cutoff frequency in Hertz. Allowed range is 50 to 900. Default
value is 100.

slope
Set slope amount for cutoff frequency. Allowed range is 0.0001 to
1. Default value is 0.5.

delay
Set delay. Allowed range is from 1 to 100. Default value is 20.

channels
Set the channels to process. Default value is all available.

Commands

This filter supports the all above options as commands.

asubcut
Cut subwoofer frequencies.

This filter allows to set custom, steeper roll off than highpass
filter, and thus is able to more attenuate frequency content in
stop-band.

The filter accepts the following options:

cutoff
Set cutoff frequency in Hertz. Allowed range is 2 to 200. Default
value is 20.

order
Set filter order. Available values are from 3 to 20. Default value
is 10.

level
Set input gain level. Allowed range is from 0 to 1. Default value
is 1.

Commands

This filter supports the all above options as commands.

asupercut
Cut super frequencies.

The filter accepts the following options:

cutoff
Set cutoff frequency in Hertz. Allowed range is 20000 to 192000.
Default value is 20000.

order
Set filter order. Available values are from 3 to 20. Default value
is 10.

level
Set input gain level. Allowed range is from 0 to 1. Default value
is 1.

Commands

This filter supports the all above options as commands.

asuperpass
Apply high order Butterworth band-pass filter.

The filter accepts the following options:

centerf
Set center frequency in Hertz. Allowed range is 2 to 999999.
Default value is 1000.

order
Set filter order. Available values are from 4 to 20. Default value
is 4.

qfactor
Set Q-factor. Allowed range is from 0.01 to 100. Default value is
1.

level
Set input gain level. Allowed range is from 0 to 2. Default value
is 1.

Commands

This filter supports the all above options as commands.

asuperstop
Apply high order Butterworth band-stop filter.

The filter accepts the following options:

centerf
Set center frequency in Hertz. Allowed range is 2 to 999999.
Default value is 1000.

order
Set filter order. Available values are from 4 to 20. Default value
is 4.

qfactor
Set Q-factor. Allowed range is from 0.01 to 100. Default value is
1.

level
Set input gain level. Allowed range is from 0 to 2. Default value
is 1.

Commands

This filter supports the all above options as commands.

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, 100.0] range.

Note that tempo greater than 2 will skip some samples rather than blend
them in. If for any reason this is a concern it is always possible to
daisy-chain several instances of atempo to achieve the desired product
tempo.

Examples

• Slow down audio to 80% tempo:

atempo=0.8

• To speed up audio to 300% tempo:

atempo=3

• To speed up audio to 300% tempo by daisy-chaining two atempo
instances:

atempo=sqrt(3),atempo=sqrt(3)

Commands

This filter supports the following commands:

tempo
Change filter tempo scale factor. Syntax for the command is :
"tempo"

atilt
Apply spectral tilt filter to audio stream.

This filter apply any spectral roll-off slope over any specified
frequency band.

The filter accepts the following options:

freq
Set central frequency of tilt in Hz. Default is 10000 Hz.

slope
Set slope direction of tilt. Default is 0. Allowed range is from -1
to 1.

width
Set width of tilt. Default is 1000. Allowed range is from 100 to
10000.

order
Set order of tilt filter.

level
Set input volume level. Allowed range is from 0 to 4. Default is
1.

Commands

This filter supports the all above options as commands.

atrim
Trim the input so that the output contains one continuous subpart of
the input.

It accepts the following parameters:

start
Timestamp (in seconds) of the start of the section to keep. I.e.
the audio sample with the timestamp start will be the first sample
in the output.

end Specify time of the first audio sample that will be dropped, i.e.
the audio sample immediately preceding the one with the timestamp
end will be the last sample in the output.

start_pts
Same as start, except this option sets the start timestamp in
samples instead of seconds.

end_pts
Same as end, except this option sets the end timestamp in samples
instead of seconds.

duration
The maximum duration of the output in seconds.

start_sample
The number of the first sample that should be output.

end_sample
The number of the first sample that should be dropped.

start, end, and duration are expressed as time duration specifications;
see the Time duration section in the ffmpeg-utils(1) manual.

Note that the first two sets of the start/end options and the duration
option look at the frame timestamp, while the _sample options simply
count the samples that pass through the filter. So start/end_pts and
start/end_sample will give different results when the timestamps are
wrong, inexact or do not start at zero. Also note that this filter does
not modify the timestamps. If you wish to have the output timestamps
start at zero, insert the asetpts filter after the atrim filter.

If multiple start or end options are set, this filter tries to be
greedy and keep all samples that match at least one of the specified
constraints. To keep only the part that matches all the constraints at
once, chain multiple atrim filters.

The defaults are such that all the input is kept. So it is possible to
set e.g. just the end values to keep everything before the specified
time.

Examples:

• Drop everything except the second minute of input:

ffmpeg -i INPUT -af atrim=60:120

• Keep only the first 1000 samples:

ffmpeg -i INPUT -af atrim=end_sample=1000

axcorrelate
Calculate normalized windowed cross-correlation between two input audio
streams.

Resulted samples are always between -1 and 1 inclusive. If result is 1
it means two input samples are highly correlated in that selected
segment. Result 0 means they are not correlated at all. If result is
-1 it means two input samples are out of phase, which means they cancel
each other.

The filter accepts the following options:

size
Set size of segment over which cross-correlation is calculated.
Default is 256. Allowed range is from 2 to 131072.

algo
Set algorithm for cross-correlation. Can be "slow" or "fast" or
"best". Default is "best". Fast algorithm assumes mean values over
any given segment are always zero and thus need much less
calculations to make. This is generally not true, but is valid for
typical audio streams.

Examples

• Calculate correlation between channels in stereo audio stream:

ffmpeg -i stereo.wav -af channelsplit,axcorrelate=size=1024:algo=fast correlation.wav

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

k kHz

width, w
Specify the band-width of a filter in width_type units.

mix, m
How much to use filtered signal in output. Default is 1. Range is
between 0 and 1.

channels, c
Specify which channels to filter, by default all available are
filtered.

normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it
will normalize magnitude response at DC to 0dB.

transform, a
Set transform type of IIR filter.

di
dii
tdi
tdii
latt
svf
zdf

precision, r
Set precision of filtering.

auto
Pick automatic sample format depending on surround filters.

s16 Always use signed 16-bit.

s32 Always use signed 32-bit.

f32 Always use float 32-bit.

f64 Always use float 64-bit.

block_size, b
Set block size used for reverse IIR processing. If this value is
set to high enough value (higher than impulse response length
truncated when reaches near zero values) filtering will become
linear phase otherwise if not big enough it will just produce nasty
artifacts.

Note that filter delay will be exactly this many samples when set
to non-zero value.

Commands

This filter supports the following commands:

frequency, f
Change bandpass frequency. Syntax for the command is : "frequency"

width_type, t
Change bandpass width_type. Syntax for the command is :
"width_type"

width, w
Change bandpass width. Syntax for the command is : "width"

mix, m
Change bandpass mix. Syntax for the command is : "mix"

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

k kHz

width, w
Specify the band-width of a filter in width_type units.

mix, m
How much to use filtered signal in output. Default is 1. Range is
between 0 and 1.

channels, c
Specify which channels to filter, by default all available are
filtered.

normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it
will normalize magnitude response at DC to 0dB.

transform, a
Set transform type of IIR filter.

di
dii
tdi
tdii
latt
svf
zdf

precision, r
Set precision of filtering.

auto
Pick automatic sample format depending on surround filters.

s16 Always use signed 16-bit.

s32 Always use signed 32-bit.

f32 Always use float 32-bit.

f64 Always use float 64-bit.

Note that filter delay will be exactly this many samples when set
to non-zero value.

Commands

This filter supports the following commands:

frequency, f
Change bandreject frequency. Syntax for the command is :
"frequency"

width_type, t
Change bandreject width_type. Syntax for the command is :
"width_type"

width, w
Change bandreject width. Syntax for the command is : "width"

mix, m
Change bandreject mix. Syntax for the command is : "mix"

bass, lowshelf
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

k kHz

width, w
Determine how steep is the filter's shelf transition.

poles, p
Set number of poles. Default is 2.

mix, m
How much to use filtered signal in output. Default is 1. Range is
between 0 and 1.

channels, c
Specify which channels to filter, by default all available are
filtered.

normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it
will normalize magnitude response at DC to 0dB.

transform, a
Set transform type of IIR filter.

di
dii
tdi
tdii
latt
svf
zdf

precision, r
Set precision of filtering.

auto
Pick automatic sample format depending on surround filters.

s16 Always use signed 16-bit.

s32 Always use signed 32-bit.

f32 Always use float 32-bit.

f64 Always use float 64-bit.

Note that filter delay will be exactly this many samples when set
to non-zero value.

Commands

This filter supports the following commands:

frequency, f
Change bass frequency. Syntax for the command is : "frequency"

width_type, t
Change bass width_type. Syntax for the command is : "width_type"

width, w
Change bass width. Syntax for the command is : "width"

gain, g
Change bass gain. Syntax for the command is : "gain"

mix, m
Change bass mix. Syntax for the command is : "mix"

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.

Commands

This filter supports the following commands:

a0
a1
a2
b0
b1
b2 Change biquad parameter. Syntax for the command is : "value"

mix, m
How much to use filtered signal in output. Default is 1. Range is
between 0 and 1.

channels, c
Specify which channels to filter, by default all available are
filtered.

normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it
will normalize magnitude response at DC to 0dB.

transform, a
Set transform type of IIR filter.

di
dii
tdi
tdii
latt
svf
zdf

precision, r
Set precision of filtering.

auto
Pick automatic sample format depending on surround filters.

s16 Always use signed 16-bit.

s32 Always use signed 32-bit.

f32 Always use float 32-bit.

f64 Always use float 64-bit.

Note that filter delay will be exactly this many samples when set
to non-zero value.

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. Mixing different types of mappings is not allowed
and will result in a parse error.

channel_layout
The channel layout of the output stream. If not specified, then
filter will guess it based on the out_channel names or the number
of mappings. Guessed layouts will not necessarily contain channels
in the order of the mappings.

If no mapping is present, the filter will implicitly map input channels
to output channels, preserving indices.

Examples

• 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".

channels
A channel layout describing the channels to be extracted as
separate output streams or "all" to extract each input channel as a
separate stream. The default is "all".

Choosing channels not present in channel layout in the input will
result in an error.

Examples

• 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

• Extract only LFE from a 5.1 WAV file:

ffmpeg -i in.wav -filter_complex 'channelsplit=channel_layout=5.1:channels=LFE[LFE]'
-map '[LFE]' lfe.wav

chorus
Add a chorus effect to the audio.

Can make a single vocal sound like a chorus, but can also be applied to
instrumentation.

Chorus resembles an echo effect with a short delay, but whereas with
echo the delay is constant, with chorus, it is varied using using
sinusoidal or triangular modulation. The modulation depth defines the
range the modulated delay is played before or after the delay. Hence
the delayed sound will sound slower or faster, that is the delayed
sound tuned around the original one, like in a chorus where some vocals
are slightly off key.

It accepts the following parameters:

in_gain
Set input gain. Default is 0.4.

out_gain
Set output gain. Default is 0.4.

delays
Set delays. A typical delay is around 40ms to 60ms.

decays
Set decays.

speeds
Set speeds.

depths
Set depths.

Examples

• A single delay:

chorus=0.7:0.9:55:0.4:0.25:2

• Two delays:

chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3

• Fuller sounding chorus with three delays:

chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3

compand
Compress or expand the audio's dynamic range.

It accepts the following parameters:

attacks
decays
A list of times in seconds for each channel over which the
instantaneous level of the input signal is averaged to determine
its volume. attacks refers to increase of volume and decays refers
to decrease of volume. For most situations, the attack time
(response to the audio getting louder) should be shorter than the
decay time, because the human ear is more sensitive to sudden loud
audio than sudden soft audio. A typical value for attack is 0.3
seconds and a typical value for decay is 0.8 seconds. If specified
number of attacks & decays is lower than number of channels, the
last set attack/decay will be used for all remaining channels.

points
A list of points for the transfer function, specified in dB
relative to the maximum possible signal amplitude. Each key points
list must be defined using the following syntax:
"x0/y0|x1/y1|x2/y2|...." or "x0/y0 x1/y1 x2/y2 ...."

The input values must be in strictly increasing order but the
transfer function does not have to be monotonically rising. The
point "0/0" is assumed but may be overridden (by "0/out-dBn").
Typical values for the transfer function are "-70/-70|-60/-20|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

• Make music with both quiet and loud passages suitable for listening
to in a noisy environment:

compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2

Another example for audio with whisper and explosion parts:

compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0

• A noise gate for when the noise is at a lower level than the
signal:

compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1

• Here is another noise gate, this time for when the noise is at a
higher level than the signal (making it, in some ways, similar to
squelch):

compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1

• 2:1 compression starting at -6dB:

compand=points=-80/-80|-6/-6|0/-3.8|20/3.5

• 2:1 compression starting at -9dB:

compand=points=-80/-80|-9/-9|0/-5.3|20/2.9

• 2:1 compression starting at -12dB:

compand=points=-80/-80|-12/-12|0/-6.8|20/1.9

• 2:1 compression starting at -18dB:

compand=points=-80/-80|-18/-18|0/-9.8|20/0.7

• 3:1 compression starting at -15dB:

compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2

• Compressor/Gate:

compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6

• 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

• Hard limiter at -6dB:

compand=attacks=0:points=-80/-80|-6/-6|20/-6

• Hard limiter at -12dB:

compand=attacks=0:points=-80/-80|-12/-12|20/-12

• Hard noise gate at -35 dB:

compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20

• 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 locations. 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 a
distance of ~30 cm between microphones makes one microphone capture the
signal in antiphase to the other microphone. That makes the final mix
sound 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.

The filter 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 in degrees Celsius. This is the temperature of the
environment. Default is 20.

Commands

This filter supports the all above options as commands.

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.

slope
Set curve slope of low shelf filter. Default is 0.5. Allowed range
is from 0.01 to 1.

level_in
Set input gain. Default is 0.9.

level_out
Set output gain. Default is 1.

block_size
Set block size used for reverse IIR processing. If this value is
set to high enough value (higher than impulse response length
truncated when reaches near zero values) filtering will become
linear phase otherwise if not big enough it will just produce nasty
artifacts.

Note that filter delay will be exactly this many samples when set
to non-zero value.

Commands

This filter supports the all above options as commands.

crystalizer
Simple algorithm for audio noise sharpening.

This filter linearly increases differences between each audio sample.

The filter accepts the following options:

i Sets the intensity of effect (default: 2.0). Must be in range
between -10.0 to 0 (unchanged sound) to 10.0 (maximum effect). To
inverse filtering use negative value.

c Enable clipping. By default is enabled.

Commands

This filter supports the all above options as commands.

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.

deesser
Apply de-essing to the audio samples.

i Set intensity for triggering de-essing. Allowed range is from 0 to
1. Default is 0.

m Set amount of ducking on treble part of sound. Allowed range is
from 0 to 1. Default is 0.5.

f How much of original frequency content to keep when de-essing.
Allowed range is from 0 to 1. Default is 0.5.

s Set the output mode.

It accepts the following values:

i Pass input unchanged.

o Pass ess filtered out.

e Pass only ess.

Default value is o.

dialoguenhance
Enhance dialogue in stereo audio.

This filter accepts stereo input and produce surround (3.0) channels
output. The newly produced front center channel have enhanced speech
dialogue originally available in both stereo channels. This filter
outputs front left and front right channels same as available in stereo
input.

The filter accepts the following options:

original
Set the original center factor to keep in front center channel
output. Allowed range is from 0 to 1. Default value is 1.

enhance
Set the dialogue enhance factor to put in front center channel
output. Allowed range is from 0 to 3. Default value is 1.

voice
Set the voice detection factor. Allowed range is from 2 to 32.
Default value is 2.

Commands

This filter supports the all above options as commands.

drmeter
Measure audio dynamic range.

DR values of 14 and higher is found in very dynamic material. DR of 8
to 13 is found in transition material. And anything less that 8 have
very poor dynamics and is very compressed.

The filter accepts the following options:

length
Set window length in seconds used to split audio into segments of
equal length. Default is 3 seconds.

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.

framelen, 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.

gausssize, 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.

peak, 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.

maxgain, 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.

targetrms, 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.

coupling, 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.

correctdc, 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.

altboundary, 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.

compress, 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.

threshold, t
Set the target threshold value. This specifies the lowest
permissible magnitude level for the audio input which will be
normalized. If input frame volume is above this value frame will
be normalized. Otherwise frame may not be normalized at all. The
default value is set to 0, which means all input frames will be
normalized. This option is mostly useful if digital noise is not
wanted to be amplified.

channels, h
Specify which channels to filter, by default all available channels
are filtered.

overlap, o
Specify overlap for frames. If set to 0 (default) no frame
overlapping is done. Using >0 and <1 values will make less
conservative gain adjustments, like when framelen option is set to
smaller value, if framelen option value is compensated for non-zero
overlap then gain adjustments will be smoother across time compared
to zero overlap case.

curve, v
Specify the peak mapping curve expression which is going to be used
when calculating gain applied to frames. The max output frame gain
will still be limited by other options mentioned previously for
this filter.

The expression can contain the following constants:

ch current channel number

sn current sample number

nb_channels
number of channels

t timestamp expressed in seconds

sr sample rate

p current frame peak value

Commands

This filter supports the all above options as commands.

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

k kHz

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.

mix, m
How much to use filtered signal in output. Default is 1. Range is
between 0 and 1.

channels, c
Specify which channels to filter, by default all available are
filtered.

normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it
will normalize magnitude response at DC to 0dB.

transform, a
Set transform type of IIR filter.

di
dii
tdi
tdii
latt
svf
zdf

precision, r
Set precision of filtering.

auto
Pick automatic sample format depending on surround filters.

s16 Always use signed 16-bit.

s32 Always use signed 32-bit.

f32 Always use float 32-bit.

f64 Always use float 64-bit.

Note that filter delay will be exactly this many samples when set
to non-zero value.

Examples

• Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:

equalizer=f=1000:t=h:width=200:g=-10

• Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz
with Q 2:

equalizer=f=1000:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5

Commands

This filter supports the following commands:

frequency, f
Change equalizer frequency. Syntax for the command is :
"frequency"

width_type, t
Change equalizer width_type. Syntax for the command is :
"width_type"

width, w
Change equalizer width. Syntax for the command is : "width"

gain, g
Change equalizer gain. Syntax for the command is : "gain"

mix, m
Change equalizer mix. Syntax for the command is : "mix"

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.

Commands

This filter supports the all above options as commands.

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

• lowpass at 1000 Hz:

firequalizer=gain='if(lt(f,1000), 0, -INF)'

• lowpass at 1000 Hz with gain_entry:

firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'

• custom equalization:

firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'

• higher delay with zero phase to compensate delay:

firequalizer=delay=0.1:fixed=on:zero_phase=on

• 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.

size
Set size of frame in number of samples which will be processed at
once. Default value is 1024. Allowed range is from 1024 to 96000.

hrir
Set format of hrir stream. Default value is stereo. Alternative
value is multich. If value is set to stereo, number of additional
streams should be greater or equal to number of input channels in
first input stream. Also each additional stream should have stereo
number of channels. If value is set to multich, number of
additional streams should be exactly one. Also number of input
channels of additional stream should be equal or greater than twice
number of channels of first input stream.

Examples

• 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
-filter_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];[0:a][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"
output.wav

• Full example using wav files as coefficients with amovie filters
for 7.1 downmix, but now in multich hrir format.

ffmpeg -i input.wav -filter_complex "amovie=minp.wav[hrirs];[0:a][hrirs]headphone=map=FL|FR|FC|LFE|BL|BR|SL|SR:hrir=multich"
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

k kHz

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.

mix, m
How much to use filtered signal in output. Default is 1. Range is
between 0 and 1.

channels, c
Specify which channels to filter, by default all available are
filtered.

normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it
will normalize magnitude response at DC to 0dB.

transform, a
Set transform type of IIR filter.

di
dii
tdi
tdii
latt
svf
zdf

precision, r
Set precision of filtering.

auto
Pick automatic sample format depending on surround filters.

s16 Always use signed 16-bit.

s32 Always use signed 32-bit.

f32 Always use float 32-bit.

f64 Always use float 64-bit.

Note that filter delay will be exactly this many samples when set
to non-zero value.

Commands

This filter supports the following commands:

frequency, f
Change highpass frequency. Syntax for the command is : "frequency"

width_type, t
Change highpass width_type. Syntax for the command is :
"width_type"

width, w
Change highpass width. Syntax for the command is : "width"

mix, m
Change highpass mix. Syntax for the command is : "mix"

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.

latency, l
Enable latency compensation, by default is disabled. Only used if
plugin have inputs.

Examples

• List all available plugins within amp (LADSPA example plugin)
library:

ladspa=file=amp

• List all available controls and their valid ranges for "vcf_notch"
plugin from "VCF" library:

ladspa=f=vcf:p=vcf_notch:c=help

• Simulate low quality audio equipment using "Computer Music Toolkit"
(CMT) plugin library:

ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12

• Add reverberation to the audio using TAP-plugins (Tom's Audio
Processing plugins):

ladspa=file=tap_reverb:tap_reverb

• Generate white noise, with 0.2 amplitude:

ladspa=file=cmt:noise_source_white:c=c0=.2

• Generate 20 bpm clicks using plugin "C* Click - Metronome" from the
"C* Audio Plugin Suite" (CAPS) library:

ladspa=file=caps:Click:c=c1=20'

• Apply "C* Eq10X2 - Stereo 10-band equaliser" effect:

ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2

• Increase volume by 20dB using fast lookahead limiter from Steve
Harris "SWH Plugins" collection:

ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2

• 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

• Reduce stereo image using "Narrower" from the "C* Audio Plugin
Suite" (CAPS) library:

ladspa=caps:Narrower

• Another white noise, now using "C* Audio Plugin Suite" (CAPS)
library:

ladspa=caps:White:.2

• Some fractal noise, using "C* Audio Plugin Suite" (CAPS) library:

ladspa=caps:Fractal:c=c1=1

• 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. In dynamic mode, to accurately detect true peaks,
the audio stream will be upsampled to 192 kHz. 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 - 50.0. 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 by linearly scaling the source audio. "measured_I",
"measured_LRA", "measured_TP", and "measured_thresh" must all be
specified. Target LRA shouldn't be lower than source LRA and the
change in integrated loudness shouldn't result in a true peak which
exceeds the target TP. If any of these conditions aren't met,
normalization mode will revert to dynamic. 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

k kHz

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.

mix, m
How much to use filtered signal in output. Default is 1. Range is
between 0 and 1.

channels, c
Specify which channels to filter, by default all available are
filtered.

normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it
will normalize magnitude response at DC to 0dB.

transform, a
Set transform type of IIR filter.

di
dii
tdi
tdii
latt
svf
zdf

precision, r
Set precision of filtering.

auto
Pick automatic sample format depending on surround filters.

s16 Always use signed 16-bit.

s32 Always use signed 32-bit.

f32 Always use float 32-bit.

f64 Always use float 64-bit.

Note that filter delay will be exactly this many samples when set
to non-zero value.

Examples

• Lowpass only LFE channel, it LFE is not present it does nothing:

lowpass=c=LFE

Commands

This filter supports the following commands:

frequency, f
Change lowpass frequency. Syntax for the command is : "frequency"

width_type, t
Change lowpass width_type. Syntax for the command is :
"width_type"

width, w
Change lowpass width. Syntax for the command is : "width"

mix, m
Change lowpass mix. Syntax for the command is : "mix"

lv2
Load a LV2 (LADSPA Version 2) plugin.

To enable compilation of this filter you need to configure FFmpeg with
"--enable-lv2".

plugin, p
Specifies the plugin URI. You may need to escape ':'.

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.

Examples

• Apply bass enhancer plugin from Calf:

lv2=p=http\\\\://calf.sourceforge.net/plugins/BassEnhancer:c=amount=2

• Apply vinyl plugin from Calf:

lv2=p=http\\\\://calf.sourceforge.net/plugins/Vinyl:c=drone=0.2|aging=0.5

• Apply bit crusher plugin from ArtyFX:

lv2=p=http\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3

Commands

This filter supports all options that are exported by plugin as
commands.

mcompand
Multiband Compress or expand the audio's dynamic range.

The input audio is divided into bands using 4th order Linkwitz-Riley
IIRs. This is akin to the crossover of a loudspeaker, and results in
flat frequency response when absent compander action.

It accepts the following parameters:

args
This option syntax is: attack,decay,[attack,decay..] soft-knee
points crossover_frequency [delay [initial_volume [gain]]] |
attack,decay ... For explanation of each item refer to compand
filter documentation.

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".

The filter accepts the following exported read-only options:

track_gain
Exported track gain in dB at end of stream.

track_peak
Exported track peak at end of stream.

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.

To enable compilation of this filter, you need to configure FFmpeg with
"--enable-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

Commands

This filter supports the following commands:

tempo
Change filter tempo scale factor. Syntax for the command is :
"tempo"

pitch
Change filter pitch scale factor. Syntax for the command is :
"pitch"

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.

mode
Set mode of compressor operation. Can be "upward" or "downward".
Default is "downward".

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.

Commands

This filter supports the all above options as commands.

Examples

• Full ffmpeg example taking 2 audio inputs, 1st input to be
compressed depending on the signal of 2nd input and later
compressed signal to be merged with 2nd input:

ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"

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.

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.

Commands

This filter supports the all above options as commands.

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
"lavfi.silence_start" or "lavfi.silence_start.X" metadata key is set on
the first frame whose timestamp equals or exceeds the detection
duration and it contains the timestamp of the first frame of the
silence.

The "lavfi.silence_duration" or "lavfi.silence_duration.X" and
"lavfi.silence_end" or "lavfi.silence_end.X" metadata keys are set on
the first frame after the silence. If mono is enabled, and each channel
is evaluated separately, the ".X" suffixed keys are used, and "X"
corresponds to the channel number.

The filter accepts the following options:

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.

duration, d
Set silence duration until notification (default is 2 seconds). See
the Time duration section in the ffmpeg-utils(1) manual for the
accepted syntax.

mono, m
Process each channel separately, instead of combined. By default is
disabled.

Examples

• Detect 5 seconds of silence with -50dB noise tolerance:

silencedetect=n=-50dB:d=5

• Complete example with ffmpeg to detect silence with 0.0001 noise
tolerance in silence.mp3:

ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -

silenceremove
Remove silence from the beginning, middle or end of the audio.

The filter accepts the following options:

start_periods
This value is used to indicate if audio should be trimmed at
beginning of the audio. A value of zero indicates no silence should
be trimmed from the beginning. When specifying a non-zero value, it
trims audio up until it finds non-silence. Normally, when trimming
silence from beginning of audio the start_periods will be 1 but it
can be increased to higher values to trim all audio up to specific
count of non-silence periods. Default value is 0.

start_duration
Specify the amount of time that non-silence must be detected before
it stops trimming audio. By increasing the duration, bursts of
noises can be treated as silence and trimmed off. Default value is
0.

start_threshold
This indicates what sample value should be treated as silence. For
digital audio, a value of 0 may be fine but for audio recorded from
analog, you may wish to increase the value to account for
background noise. Can be specified in dB (in case "dB" is appended
to the specified value) or amplitude ratio. Default value is 0.

start_silence
Specify max duration of silence at beginning that will be kept
after trimming. Default is 0, which is equal to trimming all
samples detected as silence.

start_mode
Specify mode of detection of silence end at start of multi-channel
audio. Can be any or all. Default is any. With any, any sample
from any channel that is detected as non-silence will trigger end
of silence trimming at start of audio stream. With all, only if
every sample from every channel is detected as non-silence will
trigger end of silence trimming at start of audio stream, limited
usage.

stop_periods
Set the count for trimming silence from the end of audio. When
specifying a positive value, it trims audio after it finds
specified silence period. 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 stop_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.

stop_silence
Specify max duration of silence at end that will be kept after
trimming. Default is 0, which is equal to trimming all samples
detected as silence.

stop_mode
Specify mode of detection of silence start after start of
multi-channel audio. Can be any or all. Default is all. With any,
any sample from any channel that is detected as silence will
trigger start of silence trimming after start of audio stream,
limited usage. With all, only if every sample from every channel
is detected as silence will trigger start of silence trimming after
start of audio stream.

detection
Set how is silence detected.

avg Mean of absolute values of samples in moving window.

rms Root squared mean of absolute values of samples in moving
window.

peak
Maximum of absolute values of samples in moving window.

median
Median of absolute values of samples in moving window.

ptp Absolute of max peak to min peak difference of samples in
moving window.

dev Standard deviation of values of samples in moving window.

Default value is "rms".

window
Set duration in number of seconds used to calculate size of window
in number of samples for detecting silence. Using 0 will
effectively disable any windowing and use only single sample per
channel for silence detection. In that case it may be needed to
also set start_silence and/or stop_silence to nonzero values with
also start_duration and/or stop_duration to nonzero values.
Default value is 0.02. Allowed range is from 0 to 10.

timestamp
Set processing mode of every audio frame output timestamp.

write
Full timestamps rewrite, keep only the start time for the first
output frame.

copy
Non-dropped frames are left with same timestamp as input audio
frame.

Defaults value is "write".

Examples

• The following example shows how this filter can be used to start a
recording that does not contain the delay at the start which
usually occurs between pressing the record button and the start of
the performance:

silenceremove=start_periods=1:start_duration=5:start_threshold=0.02

• Trim all silence encountered from beginning to end where there is
more than 1 second of silence in audio:

silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-90dB

• Trim all digital silence samples, using peak detection, from
beginning to end where there is more than 0 samples of digital
silence in audio and digital silence is detected in all channels at
same positions in stream:

silenceremove=window=0:detection=peak:stop_mode=all:start_mode=all:stop_periods=-1:stop_threshold=0

• Trim every 2nd encountered silence period from beginning to end
where there is more than 1 second of silence per silence period in
audio:

silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB

• Similar as above, but keep maximum of 0.5 seconds of silence from
each trimmed period:

silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB:stop_silence=0.5

• Similar as above, but keep maximum of 1.5 seconds of silence from
start of audio:

silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB:stop_silence=0.5:start_periods=1:start_duration=1:start_silence=1.5:stop_threshold=-90dB

Commands

This filter supports some above options as commands.

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.

framesize
Set custom frame size in number of samples. Default is 1024.
Allowed range is from 1024 to 96000. Only used if option type is
set to freq.

normalize
Should all IRs be normalized upon importing SOFA file. By default
is enabled.

interpolate
Should nearest IRs be interpolated with neighbor IRs if exact
position does not match. By default is disabled.

minphase
Minphase all IRs upon loading of SOFA file. By default is disabled.

anglestep
Set neighbor search angle step. Only used if option interpolate is
enabled.

radstep
Set neighbor search radius step. Only used if option interpolate is
enabled.

Examples

• Using ClubFritz6 sofa file:

sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1

• Using ClubFritz12 sofa file and bigger radius with small rotation:

sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5

• 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"

speechnorm
Speech Normalizer.

This filter expands or compresses each half-cycle of audio samples
(local set of samples all above or all below zero and between two
nearest zero crossings) depending on threshold value, so audio reaches
target peak value under conditions controlled by below options.

The filter accepts the following options:

peak, p
Set the expansion target peak value. This specifies the highest
allowed absolute amplitude level for the normalized audio input.
Default value is 0.95. Allowed range is from 0.0 to 1.0.

expansion, e
Set the maximum expansion factor. Allowed range is from 1.0 to
50.0. Default value is 2.0. This option controls maximum local
half-cycle of samples expansion. The maximum expansion would be
such that local peak value reaches target peak value but never to
surpass it and that ratio between new and previous peak value does
not surpass this option value.

compression, c
Set the maximum compression factor. Allowed range is from 1.0 to
50.0. Default value is 2.0. This option controls maximum local
half-cycle of samples compression. This option is used only if
threshold option is set to value greater than 0.0, then in such
cases when local peak is lower or same as value set by threshold
all samples belonging to that peak's half-cycle will be compressed
by current compression factor.

threshold, t
Set the threshold value. Default value is 0.0. Allowed range is
from 0.0 to 1.0. This option specifies which half-cycles of
samples will be compressed and which will be expanded. Any
half-cycle samples with their local peak value below or same as
this option value will be compressed by current compression factor,
otherwise, if greater than threshold value they will be expanded
with expansion factor so that it could reach peak target value but
never surpass it.

raise, r
Set the expansion raising amount per each half-cycle of samples.
Default value is 0.001. Allowed range is from 0.0 to 1.0. This
controls how fast expansion factor is raised per each new
half-cycle until it reaches expansion value. Setting this options
too high may lead to distortions.

fall, f
Set the compression raising amount per each half-cycle of samples.
Default value is 0.001. Allowed range is from 0.0 to 1.0. This
controls how fast compression factor is raised per each new
half-cycle until it reaches compression value.

channels, h
Specify which channels to filter, by default all available channels
are filtered.

invert, i
Enable inverted filtering, by default is disabled. This inverts
interpretation of threshold option. When enabled any half-cycle of
samples with their local peak value below or same as threshold
option will be expanded otherwise it will be compressed.

link, l
Link channels when calculating gain applied to each filtered
channel sample, by default is disabled. When disabled each
filtered channel gain calculation is independent, otherwise when
this option is enabled the minimum of all possible gains for each
filtered channel is used.

rms, m
Set the expansion target RMS value. This specifies the highest
allowed RMS level for the normalized audio input. Default value is
0.0, thus disabled. Allowed range is from 0.0 to 1.0.

Commands

This filter supports the all above options as commands.

Examples

• Weak and slow amplification:

speechnorm=e=3:r=0.00001:l=1

• Moderate and slow amplification:

speechnorm=e=6.25:r=0.00001:l=1

• Strong and fast amplification:

speechnorm=e=12.5:r=0.0001:l=1

• Very strong and fast amplification:

speechnorm=e=25:r=0.0001:l=1

• Extreme and fast amplification:

speechnorm=e=50:r=0.0001:l=1

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.

ms>rl
Mid/Side to Right/Left.

lr>l-r
Left/Right to Left - 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.

Commands

This filter supports the all above options as commands.

Examples

• Apply karaoke like effect:

stereotools=mlev=0.015625

• 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.

Commands

This filter supports the all above options except "delay" as commands.

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.

lfe_mode
Set LFE mode, can be add or sub. Default is add. In add mode, LFE
channel is created from input audio and added to output. In sub
mode, LFE channel is created from input audio and added to output
but also all non-LFE output channels are subtracted with output LFE
channel.

smooth
Set temporal smoothness strength, used to gradually change factors
when transforming stereo sound in time. Allowed range is from 0.0
to 1.0. Useful to improve output quality with focus option values
greater than 0.0. Default is 0.0. Only values inside this range
and without edges are effective.

angle
Set angle of stereo surround transform, Allowed range is from 0 to
360. Default is 90.

focus
Set focus of stereo surround transform, Allowed range is from -1 to
1. Default is 0.

fc_in
Set front center input volume. By default, this is 1.

fc_out
Set front center output volume. By default, this is 1.

fl_in
Set front left input volume. By default, this is 1.

fl_out
Set front left output volume. By default, this is 1.

fr_in
Set front right input volume. By default, this is 1.

fr_out
Set front right output volume. By default, this is 1.

sl_in
Set side left input volume. By default, this is 1.

sl_out
Set side left output volume. By default, this is 1.

sr_in
Set side right input volume. By default, this is 1.

sr_out
Set side right output volume. By default, this is 1.

bl_in
Set back left input volume. By default, this is 1.

bl_out
Set back left output volume. By default, this is 1.

br_in
Set back right input volume. By default, this is 1.

br_out
Set back right output volume. By default, this is 1.

bc_in
Set back center input volume. By default, this is 1.

bc_out
Set back 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.

allx
Set spread usage of stereo image across X axis for all channels.
Allowed range is from -1 to 15. By default this value is negative
-1, and thus unused.

ally
Set spread usage of stereo image across Y axis for all channels.
Allowed range is from -1 to 15. By default this value is negative
-1, and thus unused.

fcx, flx, frx, blx, brx, slx, srx, bcx
Set spread usage of stereo image across X axis for each channel.
Allowed range is from 0.06 to 15. By default this value is 0.5.

fcy, fly, fry, bly, bry, sly, sry, bcy
Set spread usage of stereo image across Y axis for each channel.
Allowed range is from 0.06 to 15. By default this value is 0.5.

win_size
Set window size. Allowed range is from 1024 to 65536. Default size
is 4096.

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
bohman
kaiser

Default is "hann".

overlap
Set window overlap. If set to 1, the recommended overlap for
selected window function will be picked. Default is 0.5.

tiltshelf
Boost or cut the lower frequencies and cut or boost higher 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 3000 Hz.

width_type, t
Set method to specify band-width of filter.

h Hz

q Q-Factor

o octave

s slope

k kHz

width, w
Determine how steep is the filter's shelf transition.

poles, p
Set number of poles. Default is 2.

mix, m
How much to use filtered signal in output. Default is 1. Range is
between 0 and 1.

channels, c
Specify which channels to filter, by default all available are
filtered.

normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it
will normalize magnitude response at DC to 0dB.

transform, a
Set transform type of IIR filter.

di
dii
tdi
tdii
latt
svf
zdf

precision, r
Set precision of filtering.

auto
Pick automatic sample format depending on surround filters.

s16 Always use signed 16-bit.

s32 Always use signed 32-bit.

f32 Always use float 32-bit.

f64 Always use float 64-bit.

Note that filter delay will be exactly this many samples when set
to non-zero value.

Commands

This filter supports some options as commands.

treble, highshelf
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

k kHz

width, w
Determine how steep is the filter's shelf transition.

poles, p
Set number of poles. Default is 2.

mix, m
How much to use filtered signal in output. Default is 1. Range is
between 0 and 1.

channels, c
Specify which channels to filter, by default all available are
filtered.

normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it
will normalize magnitude response at DC to 0dB.

transform, a
Set transform type of IIR filter.

di
dii
tdi
tdii
latt
svf
zdf

precision, r
Set precision of filtering.

auto
Pick automatic sample format depending on surround filters.

s16 Always use signed 16-bit.

s32 Always use signed 32-bit.

f32 Always use float 32-bit.

f64 Always use float 64-bit.

Note that filter delay will be exactly this many samples when set
to non-zero value.

Commands

This filter supports the following commands:

frequency, f
Change treble frequency. Syntax for the command is : "frequency"

width_type, t
Change treble width_type. Syntax for the command is : "width_type"

width, w
Change treble width. Syntax for the command is : "width"

gain, g
Change treble gain. Syntax for the command is : "gain"

mix, m
Change treble mix. Syntax for the command is : "mix"

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.

virtualbass
Apply audio Virtual Bass filter.

This filter accepts stereo input and produce stereo with LFE (2.1)
channels output. The newly produced LFE channel have enhanced virtual
bass originally obtained from both stereo channels. This filter
outputs front left and front right channels unchanged as available in
stereo input.

The filter accepts the following options:

cutoff
Set the virtual bass cutoff frequency. Default value is 250 Hz.
Allowed range is from 100 to 500 Hz.

strength
Set the virtual bass strength. Allowed range is from 0.5 to 3.
Default value is 3.

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.

replaygain_noclip
Prevent clipping by limiting the gain applied.

Default value for replaygain_noclip is 1.

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; deprecated, do not use

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.

Examples

• Halve the input audio volume:

volume=volume=0.5
volume=volume=1/2
volume=volume=-6.0206dB

In all the above example the named key for volume can be omitted,
for example like in:

volume=0.5

• Increase input audio power by 6 decibels using fixed-point
precision:

volume=volume=6dB:precision=fixed

• Fade volume after time 10 with an annihilation period of 5 seconds:

volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame

volumedetect
Detect the volume of the input video.

The filter has no parameters. It supports only 16-bit signed integer
samples, so the input will be converted when needed. Statistics about
the volume will be printed in the log when the input stream end is
reached.

In particular it will show the mean volume (root mean square), maximum
volume (on a per-sample basis), and the beginning of a histogram of the
registered volume values (from the maximum value to a cumulated 1/1000
of the samples).

All volumes are in decibels relative to the maximum PCM value.

Examples

Here is an excerpt of the output:

[Parsed_volumedetect_0 0xa23120] mean_volume: -27 dB
[Parsed_volumedetect_0 0xa23120] max_volume: -4 dB
[Parsed_volumedetect_0 0xa23120] histogram_4db: 6
[Parsed_volumedetect_0 0xa23120] histogram_5db: 62
[Parsed_volumedetect_0 0xa23120] histogram_6db: 286
[Parsed_volumedetect_0 0xa23120] histogram_7db: 1042
[Parsed_volumedetect_0 0xa23120] histogram_8db: 2551
[Parsed_volumedetect_0 0xa23120] histogram_9db: 4609
[Parsed_volumedetect_0 0xa23120] histogram_10db: 8409

It means that:

• The mean square energy is approximately -27 dB, or 10^-2.7.

• The largest sample is at -4 dB, or more precisely between -4 dB and
-5 dB.

• There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

In other words, raising the volume by +4 dB does not cause any
clipping, raising it by +5 dB causes clipping for 6 samples, etc.

whisper
It runs automatic speech recognition using the OpenAI's Whisper model.

It requires the whisper.cpp library
(https://github.com/ggml-org/whisper.cpp) as a prerequisite. After
installing the library it can be enabled using: "./configure
--enable-whisper".

The filter has following options:

model
The file path of the downloaded whisper.cpp model (mandatory).

language
The language to use for transcription ('auto' for auto-detect).
Default value: "auto"

queue
The maximum size that will be queued into the filter before
processing the audio with whisper. Using a small value the audio
stream will be processed more often, but the transcription quality
will be lower and the required processing power will be higher.
Using a large value (e.g. 10-20s) will produce more accurate
results using less CPU (as using the whisper-cli tool), but the
transcription latency will be higher, thus not useful to process
real-time streams. Consider using the vad_model option associated
with a large queue value. Default value: "3"

use_gpu
If the GPU support should be enabled. Default value: "true"

gpu_device
The GPU device index to use. Default value: "0"

destination
If set, the transcription output will be sent to the specified file
or URL (use one of the FFmpeg AVIO protocols); otherwise, the
output will be logged as info messages. The output will also be
set in the "lavfi.whisper.text" frame metadata. If the destination
is a file and it already exists, it will be overwritten.

format
The destination format string; it could be "text" (only the
transcribed text will be sent to the destination), "srt" (subtitle
format) or "json". Default value: "text"

vad_model
Path to the VAD model file. If set, the filter will load an
additional voice activity detection module
(https://github.com/snakers4/silero-vad) that will be used to
fragment the audio queue; use this option setting a valid path
obtained from the whisper.cpp repository (e.g.
"../whisper.cpp/models/ggml-silero-v5.1.2.bin") and increase the
queue parameter to a higher value (e.g. 20).

vad_threshold
The VAD threshold to use. Default value: "0.5"

vad_min_speech_duration
The minimum VAD speaking duration. Default value: "0.1"

vad_min_silence_duration
The minimum VAD silence duration. Default value: "0.5"

Examples

• Run a transcription with srt file generation:

ffmpeg -i input.mp4 -vn -af "whisper=model=../whisper.cpp/models/ggml-base.en.bin\
:language=en\
:queue=3\
:destination=output.srt\
:format=srt" -f null -

• Run a transcription and send the output in JSON format to an HTTP
service:

ffmpeg -i input.mp4 -vn -af "whisper=model=../whisper.cpp/models/ggml-base.en.bin\
:language=en\
:queue=3\
:destination=http\\://localhost\\:3000\
:format=json' -f null -

• Transcribe the microphone input using the VAD option:

ffmpeg -loglevel warning -f pulse -i default \
-af 'highpass=f=200,lowpass=f=3000,whisper=model=../whisper.cpp/models/ggml-medium.bin\
:language=en\
:queue=10\
:destination=-\
:format=json\
:vad_model=../whisper.cpp/models/ggml-silero-v5.1.2.bin' -f null -

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/buffersrc.h.

It accepts the following parameters:

time_base
The timebase which will be used for timestamps of submitted frames.
It must be either a floating-point number or in
numerator/denominator form.

sample_rate
The sample rate of the incoming audio buffers.

sample_fmt
The sample format of the incoming audio buffers. Either a sample
format name or its corresponding integer representation from the
enum AVSampleFormat in libavutil/samplefmt.h

channel_layout
The channel layout of the incoming audio buffers. Either a channel
layout name from channel_layout_map in libavutil/channel_layout.c
or its corresponding integer representation from the AV_CH_LAYOUT_*
macros in libavutil/channel_layout.h

channels
The number of channels of the incoming audio buffers. If both
channels and channel_layout are specified, then they must be
consistent.

Examples

abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo

will instruct the source to accept planar 16bit signed stereo at
44100Hz. Since the sample format with name "s16p" corresponds to the
number 6 and the "stereo" channel layout corresponds to the value 0x3,
this is equivalent to:

abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3

aevalsrc
Generate an audio signal specified by an expression.

This source accepts in input one or more expressions (one for each
channel), which are evaluated and used to generate a corresponding
audio signal.

This source accepts the following options:

exprs
Set the '|'-separated expressions list for each separate channel.
In case the channel_layout option is not specified, the selected
channel layout depends on the number of provided expressions.
Otherwise the last specified expression is applied to the remaining
output channels.

channel_layout, c
Set the channel layout. The number of channels in the specified
layout must be equal to the number of specified expressions.

If not specified, or the expressed duration is negative, the audio
is supposed to be generated forever.

nb_samples, n
Set the number of samples per channel per each output frame,
default to 1024.

sample_rate, s
Specify the sample rate, default to 44100.

Each expression in exprs can contain the following constants:

n number of the evaluated sample, starting from 0

t time of the evaluated sample expressed in seconds, starting from 0

s sample rate

Examples

• Generate silence:

aevalsrc=0

• Generate a sin signal with frequency of 440 Hz, set sample rate to
8000 Hz:

aevalsrc="sin(440*2*PI*t):s=8000"

• Generate a two channels signal, specify the channel layout (Front
Center + Back Center) explicitly:

aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"

• Generate white noise:

aevalsrc="-2+random(0)"

• Generate an amplitude modulated signal:

aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"

• Generate 2.5 Hz binaural beats on a 360 Hz carrier:

aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"

afdelaysrc
Generate a fractional delay FIR coefficients.

The resulting stream can be used with afir filter for filtering the
audio signal.

The filter accepts the following options:

delay, d
Set the fractional delay. Default is 0.

sample_rate, r
Set the sample rate, default is 44100.

nb_samples, n
Set the number of samples per each frame. Default is 1024.

taps, t
Set the number of filter coefficients in output audio stream.
Default value is 0.

channel_layout, c
Specifies the channel layout, and can be a string representing a
channel layout. The default value of channel_layout is "stereo".

afireqsrc
Generate a FIR equalizer coefficients.

The resulting stream can be used with afir filter for filtering the
audio signal.

The filter accepts the following options:

preset, p
Set equalizer preset. Default preset is "flat".

Available presets are:

custom
flat
acoustic
bass
beats
classic
clear
deep bass
dubstep
electronic
hard-style
hip-hop
jazz
metal
movie
pop
r&b
rock
vocal booster

gains, g
Set custom gains for each band. Only used if the preset option is
set to "custom". Gains are separated by white spaces and each gain
is set in dBFS. Default is "0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0".

bands, b
Set the custom bands from where custom equalizer gains are set.
This must be in strictly increasing order. Only used if the preset
option is set to "custom". Bands are separated by white spaces and
each band represent frequency in Hz. Default is "25 40 63 100 160
250 400 630 1000 1600 2500 4000 6300 10000 16000 24000".

taps, t
Set number of filter coefficients in output audio stream. Default
value is 4096.

sample_rate, r
Set sample rate of output audio stream, default is 44100.

nb_samples, n
Set number of samples per each frame in output audio stream.
Default is 1024.

interp, i
Set interpolation method for FIR equalizer coefficients. Can be
"linear" or "cubic".

phase, h
Set phase type of FIR filter. Can be "linear" or "min":
minimum-phase. Default is minimum-phase filter.

afirsrc
Generate a FIR coefficients using frequency sampling method.

The resulting stream can be used with afir filter for filtering the
audio signal.

The filter accepts the following options:

taps, t
Set number of filter coefficients in output audio stream. Default
value is 1025.

frequency, f
Set frequency points from where magnitude and phase are set. This
must be in non decreasing order, and first element must be 0, while
last element must be 1. Elements are separated by white spaces.

magnitude, m
Set magnitude value for every frequency point set by frequency.
Number of values must be same as number of frequency points.
Values are separated by white spaces.

phase, p
Set phase value for every frequency point set by frequency. Number
of values must be same as number of frequency points. Values are
separated by white spaces.

sample_rate, r
Set sample rate, default is 44100.

nb_samples, n
Set number of samples per each frame. Default is 1024.

win_func, w
Set window function. Default is blackman.

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.

duration, d
Set the duration of the sourced audio. 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 audio
is supposed to be generated forever.

Examples

• Set the sample rate to 48000 Hz and the channel layout to
AV_CH_LAYOUT_MONO.

anullsrc=r=48000:cl=4

• Do the same operation with a more obvious syntax:

anullsrc=r=48000:cl=mono

All the parameters need to be explicitly defined.

flite
Synthesize a voice utterance using the libflite library.

To enable compilation of this filter you need to configure FFmpeg with
"--enable-libflite".

Note that versions of the flite library prior to 2.0 are not
thread-safe.

The filter accepts the following options:

list_voices
If set to 1, list the names of the available voices and exit
immediately. Default value is 0.

nb_samples, n
Set the maximum number of samples per frame. Default value is 512.

textfile
Set the filename containing the text to speak.

text
Set the text to speak.

voice, v
Set the voice to use for the speech synthesis. Default value is
"kal". See also the list_voices option.

Examples

• Read from file speech.txt, and synthesize the text using the
standard flite voice:

flite=textfile=speech.txt

• Read the specified text selecting the "slt" voice:

flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

• Input text to ffmpeg:

ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

• Make ffplay speak the specified text, using "flite" and the "lavfi"
device:

ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'

For more information about libflite, check:
<http://www.festvox.org/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, violet and velvet. 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.

density
Set the density (0.0 - 1.0) for the velvet noise generator, default
is 0.05.

Examples

• 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

hilbert
Generate odd-tap Hilbert transform FIR coefficients.

The resulting stream can be used with afir filter for phase-shifting
the signal by 90 degrees.

This is used in many matrix coding schemes and for analytic signal
generation. The process is often written as a multiplication by i (or
j), the imaginary unit.

The filter accepts the following options:

sample_rate, s
Set sample rate, default is 44100.

taps, t
Set length of FIR filter, default is 22051.

nb_samples, n
Set number of samples per each frame.

win_func, w
Set window function to be used when generating FIR coefficients.

sinc
Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or
band-reject FIR coefficients.

The resulting stream can be used with afir filter for filtering the
audio signal.

The filter accepts the following options:

sample_rate, r
Set sample rate, default is 44100.

nb_samples, n
Set number of samples per each frame. Default is 1024.

hp Set high-pass frequency. Default is 0.

lp Set low-pass frequency. Default is 0. If high-pass frequency is
lower than low-pass frequency and low-pass frequency is higher than
0 then filter will create band-pass filter coefficients, otherwise
band-reject filter coefficients.

phase
Set filter phase response. Default is 50. Allowed range is from 0
to 100.

beta
Set Kaiser window beta.

att Set stop-band attenuation. Default is 120dB, allowed range is from
40 to 180 dB.

round
Enable rounding, by default is disabled.

hptaps
Set number of taps for high-pass filter.

lptaps
Set number of taps for low-pass filter.

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

• Generate a simple 440 Hz sine wave:

sine

• Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5
seconds:

sine=220:4:d=5
sine=f=220:b=4:d=5
sine=frequency=220:beep_factor=4:duration=5

• 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.

addroi
Mark a region of interest in a video frame.

The frame data is passed through unchanged, but metadata is attached to
the frame indicating regions of interest which can affect the behaviour
of later encoding. Multiple regions can be marked by applying the
filter multiple times.

x Region distance in pixels from the left edge of the frame.

y Region distance in pixels from the top edge of the frame.

w Region width in pixels.

h Region height in pixels.

The parameters x, y, w and h are expressions, and may contain the
following variables:

iw Width of the input frame.

ih Height of the input frame.

qoffset
Quantisation offset to apply within the region.

This must be a real value in the range -1 to +1. A value of zero
indicates no quality change. A negative value asks for better
quality (less quantisation), while a positive value asks for worse
quality (greater quantisation).

The range is calibrated so that the extreme values indicate the
largest possible offset - if the rest of the frame is encoded with
the worst possible quality, an offset of -1 indicates that this
region should be encoded with the best possible quality anyway.
Intermediate values are then interpolated in some codec-dependent
way.

For example, in 10-bit H.264 the quantisation parameter varies
between -12 and 51. A typical qoffset value of -1/10 therefore
indicates that this region should be encoded with a QP around
one-tenth of the full range better than the rest of the frame. So,
if most of the frame were to be encoded with a QP of around 30,
this region would get a QP of around 24 (an offset of approximately
-1/10 * (51 - -12) = -6.3). An extreme value of -1 would indicate
that this region should be encoded with the best possible quality
regardless of the treatment of the rest of the frame - that is,
should be encoded at a QP of -12.

clear
If set to true, remove any existing regions of interest marked on
the frame before adding the new one.

Examples

• Mark the centre quarter of the frame as interesting.

addroi=iw/4:ih/4:iw/2:ih/2:-1/10

• Mark the 100-pixel-wide region on the left edge of the frame as
very uninteresting (to be encoded at much lower quality than the
rest of the frame).

addroi=0:0:100:ih:+1/5

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]

amplify
Amplify differences between current pixel and pixels of adjacent frames
in same pixel location.

This filter accepts the following options:

radius
Set frame radius. Default is 2. Allowed range is from 1 to 63. For
example radius of 3 will instruct filter to calculate average of 7
frames.

factor
Set factor to amplify difference. Default is 2. Allowed range is
from 0 to 65535.

threshold
Set threshold for difference amplification. Any difference greater
or equal to this value will not alter source pixel. Default is 10.
Allowed range is from 0 to 65535.

tolerance
Set tolerance for difference amplification. Any difference lower to
this value will not alter source pixel. Default is 0. Allowed
range is from 0 to 65535.

low Set lower limit for changing source pixel. Default is 65535.
Allowed range is from 0 to 65535. This option controls maximum
possible value that will decrease source pixel value.

high
Set high limit for changing source pixel. Default is 65535. Allowed
range is from 0 to 65535. This option controls maximum possible
value that will increase source pixel value.

planes
Set which planes to filter. Default is all. Allowed range is from 0
to 15.

Commands

This filter supports the following commands that corresponds to option
of same name:

factor
threshold
tolerance
low
high
planes

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 9.
Must be odd number in range [5, 129].

p Set what planes of frame filter will use for averaging. Default is
all.

a Set what variant of algorithm filter will use for averaging.
Default is "p" parallel. Alternatively can be set to "s" serial.

Parallel can be faster then serial, while other way around is never
true. Parallel will abort early on first change being greater then
thresholds, while serial will continue processing other side of
frames if they are equal or below thresholds.

0s
1s
2s Set sigma for 1st plane, 2nd plane or 3rd plane. Default is 32767.
Valid range is from 0 to 32767. This options controls weight for
each pixel in radius defined by size. Default value means every
pixel have same weight. Setting this option to 0 effectively
disables filtering.

Commands

This filter supports same commands as options except option "s". The
command accepts the same syntax of the corresponding option.

avgblur
Apply average blur filter.

The filter accepts the following options:

sizeX
Set horizontal radius size.

planes
Set which planes to filter. By default all planes are filtered.

sizeY
Set vertical radius size, if zero it will be same as "sizeX".
Default is 0.

Commands

This filter supports same commands as options. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

backgroundkey
Turns a static background into transparency.

The filter accepts the following option:

threshold
Threshold for scene change detection.

similarity
Similarity percentage with the background.

blend
Set the blend amount for pixels that are not similar.

Commands

This filter supports the all above options as commands.

bbox
Compute the bounding box for the non-black pixels in the input frame
luma plane.

This filter computes the bounding box containing all the pixels with a
luma 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 luma value. Default is 16.

Commands

This filter supports the all above options as commands.

bilateral
Apply bilateral filter, spatial smoothing while preserving edges.

The filter accepts the following options:

sigmaS
Set sigma of gaussian function to calculate spatial weight.
Allowed range is 0 to 512. Default is 0.1.

sigmaR
Set sigma of gaussian function to calculate range weight. Allowed
range is 0 to 1. Default is 0.1.

planes
Set planes to filter. Default is first only.

Commands

This filter supports the all above options as commands.

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, blackdetect_vulkan
Detect video intervals that are (almost) completely black. Can be
useful to detect chapter transitions, commercials, or invalid
recordings.

The filter outputs its detection analysis to both the log as well as
frame metadata. If a black segment of at least the specified minimum
duration is found, a line with the start and end timestamps as well as
duration is printed to the log with level "info". In addition, a log
line with level "debug" is printed per frame showing the black amount
detected for that frame.

The filter also attaches metadata to the first frame of a black segment
with key "lavfi.black_start" and to the first frame after the black
segment ends with key "lavfi.black_end". The value is the frame's
timestamp. This metadata is added regardless of the minimum duration
specified.

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 luma value for which a
pixel is considered "black". The provided value is scaled according
to the following equation:

<absolute_threshold> = <luma_minimum_value> + <pixel_black_th> * <luma_range_size>

luma_range_size and luma_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.

alpha
If true, check the alpha channel instead of the luma channel.
Detects frames which are (almost) transparent, instead of frames
which are almost black.

Default value is disabled.

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
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
and
average
bleach
burn
darken
difference
divide
dodge
exclusion
extremity
freeze
geometric
glow
grainextract
grainmerge
hardlight
hardmix
hardoverlay
harmonic
heat
interpolate
lighten
linearlight
multiply
multiply128
negation
normal
or
overlay
phoenix
pinlight
reflect
screen
softdifference
softlight
stain
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 for the plane being filtered. It is the
ratio between the dimensions of the current plane to the luma
plane, e.g. for a "yuv420p" frame, the values are "1,1" for the
luma plane and "0.5,0.5" for the 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

• Apply transition from bottom layer to top layer in first 10
seconds:

blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'

• Apply linear horizontal transition from top layer to bottom layer:

blend=all_expr='A*(X/W)+B*(1-X/W)'

• Apply 1x1 checkerboard effect:

blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'

• Apply uncover left effect:

blend=all_expr='if(gte(N*SW+X,W),A,B)'

• Apply uncover down effect:

blend=all_expr='if(gte(Y-N*SH,0),A,B)'

• Apply uncover up-left effect:

blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'

• Split diagonally video and shows top and bottom layer on each side:

blend=all_expr='if(gt(X,Y*(W/H)),A,B)'

• Display differences between the current and the previous frame:

tblend=all_mode=grainextract

Commands

This filter supports same commands as options.

blockdetect
Determines blockiness of frames without altering the input frames.

Based on Remco Muijs and Ihor Kirenko: "A no-reference blocking
artifact measure for adaptive video processing." 2005 13th European
signal processing conference.

The filter accepts the following options:

period_min
period_max
Set minimum and maximum values for determining pixel grids
(periods). Default values are [3,24].

planes
Set planes to filter. Default is first only.

Examples

• Determine blockiness for the first plane and search for periods
within [8,32]:

blockdetect=period_min=8:period_max=32:planes=1

blurdetect
Determines blurriness of frames without altering the input frames.

Based on Marziliano, Pina, et al. "A no-reference perceptual blur
metric." Allows for a block-based abbreviation.

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".

radius
Define the radius to search around an edge pixel for local maxima.

block_pct
Determine blurriness only for the most significant blocks, given in
percentage.

block_width
Determine blurriness for blocks of width block_width. If set to any
value smaller 1, no blocks are used and the whole image is
processed as one no matter of block_height.

block_height
Determine blurriness for blocks of height block_height. If set to
any value smaller 1, no blocks are used and the whole image is
processed as one no matter of block_width.

planes
Set planes to filter. Default is first only.

Examples

• Determine blur for 80% of most significant 32x32 blocks:

blurdetect=block_width=32:block_height=32:block_pct=80

bm3d
Denoise frames using Block-Matching 3D algorithm.

The filter accepts the following options.

sigma
Set denoising strength. Default value is 1. Allowed range is from
0 to 999.9. The denoising algorithm is very sensitive to sigma, so
adjust it according to the source.

block
Set local patch size. This sets dimensions in 2D.

bstep
Set sliding step for processing blocks. Default value is 4.
Allowed range is from 1 to 64. Smaller values allows processing
more reference blocks and is slower.

group
Set maximal number of similar blocks for 3rd dimension. Default
value is 1. When set to 1, no block matching is done. Larger
values allows more blocks in single group. Allowed range is from 1
to 256.

range
Set radius for search block matching. Default is 9. Allowed range
is from 1 to INT32_MAX.

mstep
Set step between two search locations for block matching. Default
is 1. Allowed range is from 1 to 64. Smaller is slower.

thmse
Set threshold of mean square error for block matching. Valid range
is 0 to INT32_MAX.

hdthr
Set thresholding parameter for hard thresholding in 3D transformed
domain. Larger values results in stronger hard-thresholding
filtering in frequency domain.

estim
Set filtering estimation mode. Can be "basic" or "final". Default
is "basic".

ref If enabled, filter will use 2nd stream for block matching. Default
is disabled for "basic" value of estim option, and always enabled
if value of estim is "final".

planes
Set planes to filter. Default is all available except alpha.

Examples

• Basic filtering with bm3d:

bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic

• Same as above, but filtering only luma:

bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic:planes=1

• Same as above, but with both estimation modes:

split[a][b],[a]bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic[a],[b][a]bm3d=sigma=3:block=4:bstep=2:group=16:estim=final:ref=1

• Same as above, but prefilter with nlmeans filter instead:

split[a][b],[a]nlmeans=s=3:r=7:p=3[a],[b][a]bm3d=sigma=3:block=4:bstep=2:group=16:estim=final:ref=1

boxblur
Apply a boxblur algorithm to the input video.

It accepts the following parameters:

luma_radius, lr
luma_power, lp
chroma_radius, cr
chroma_power, cp
alpha_radius, ar
alpha_power, ap

A description of the accepted options follows.

luma_radius, lr
chroma_radius, cr
alpha_radius, ar
Set an expression for the box radius in pixels used for blurring
the corresponding input plane.

The radius value must be a non-negative number, and must not be
greater than the value of the expression "min(w,h)/2" for the luma
and alpha planes, and of "min(cw,ch)/2" for the chroma planes.

Default value for luma_radius is "2". If not specified,
chroma_radius and alpha_radius default to the corresponding value
set for luma_radius.

The expressions can contain the following constants:

w
h The input width and height in pixels.

cw
ch The input chroma image width and height in pixels.

hsub
vsub
The horizontal and vertical chroma subsample values. For
example, for the pixel format "yuv422p", hsub is 2 and vsub is
1.

luma_power, lp
chroma_power, cp
alpha_power, ap
Specify how many times the boxblur filter is applied to the
corresponding plane.

Default value for luma_power is 2. If not specified, chroma_power
and alpha_power default to the corresponding value set for
luma_power.

A value of 0 will disable the effect.

Examples

• Apply a boxblur filter with the luma, chroma, and alpha radii set
to 2:

boxblur=luma_radius=2:luma_power=1
boxblur=2:1

• Set the luma radius to 2, and alpha and chroma radius to 0:

boxblur=2:1:cr=0:ar=0

• Set the luma and chroma radii to a fraction of the video dimension:

boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1

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. Accepts one of the following
values:

0, all
Deinterlace all frames.

1, interlaced
Only deinterlace frames marked as interlaced.

The default value is "all".

ccrepack
Repack CEA-708 closed captioning side data

This filter fixes various issues seen with commercial encoders related
to upstream malformed CEA-708 payloads, specifically incorrect number
of tuples (wrong cc_count for the target FPS), and incorrect ordering
of tuples (i.e. the CEA-608 tuples are not at the first entries in the
payload).

cas
Apply Contrast Adaptive Sharpen filter to video stream.

The filter accepts the following options:

strength
Set the sharpening strength. Default value is 0.

planes
Set planes to filter. Default value is to filter all planes except
alpha plane.

Commands

This filter supports same commands as options.

chromahold
Remove all color information for all colors except for certain one.

The filter accepts the following options:

color
The color which will not be replaced with neutral chroma.

similarity
Similarity percentage with the above color. 0.01 matches only the
exact key color, while 1.0 matches everything.

blend
Blend percentage. 0.0 makes pixels either fully gray, or not gray
at all. Higher values result in more preserved 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.

Commands

This filter supports same commands as options. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

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.

Commands

This filter supports same commands as options. 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

• Make every green pixel in the input image transparent:

ffmpeg -i input.png -vf chromakey=green out.png

• 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

chromanr
Reduce chrominance noise.

The filter accepts the following options:

thres
Set threshold for averaging chrominance values. Sum of absolute
difference of Y, U and V pixel components of current pixel and
neighbour pixels lower than this threshold will be used in
averaging. Luma component is left unchanged and is copied to
output. Default value is 30. Allowed range is from 1 to 200.

sizew
Set horizontal radius of rectangle used for averaging. Allowed
range is from 1 to 100. Default value is 5.

sizeh
Set vertical radius of rectangle used for averaging. Allowed range
is from 1 to 100. Default value is 5.

stepw
Set horizontal step when averaging. Default value is 1. Allowed
range is from 1 to 50. Mostly useful to speed-up filtering.

steph
Set vertical step when averaging. Default value is 1. Allowed
range is from 1 to 50. Mostly useful to speed-up filtering.

threy
Set Y threshold for averaging chrominance values. Set finer
control for max allowed difference between Y components of current
pixel and neighbour pixels. Default value is 200. Allowed range is
from 1 to 200.

threu
Set U threshold for averaging chrominance values. Set finer
control for max allowed difference between U components of current
pixel and neighbour pixels. Default value is 200. Allowed range is
from 1 to 200.

threv
Set V threshold for averaging chrominance values. Set finer
control for max allowed difference between V components of current
pixel and neighbour pixels. Default value is 200. Allowed range is
from 1 to 200.

distance
Set distance type used in calculations.

manhattan
Absolute difference.

euclidean
Difference squared.

Default distance type is manhattan.

Commands

This filter supports same commands as options. The command accepts the
same syntax of the corresponding option.

chromashift
Shift chroma pixels horizontally and/or vertically.

The filter accepts the following options:

cbh Set amount to shift chroma-blue horizontally.

cbv Set amount to shift chroma-blue vertically.

crh Set amount to shift chroma-red horizontally.

crv Set amount to shift chroma-red vertically.

edge
Set edge mode, can be smear, default, or warp.

Commands

This filter supports the all above options as commands.

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
dcip3

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.

fill
Fill with CIE colors. By default is enabled.

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:

block
Display block partition structure using the luma plane.

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

• Visualize forward predicted MVs of all frames using ffplay:

ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv_type=fp

• 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.

pl Preserve lightness when changing color balance. Default is
disabled.

Examples

• Add red color cast to shadows:

colorbalance=rs=.3

Commands

This filter supports the all above options as commands.

colorcontrast
Adjust color contrast between RGB components.

The filter accepts the following options:

rc Set the red-cyan contrast. Defaults is 0.0. Allowed range is from
-1.0 to 1.0.

gm Set the green-magenta contrast. Defaults is 0.0. Allowed range is
from -1.0 to 1.0.

by Set the blue-yellow contrast. Defaults is 0.0. Allowed range is
from -1.0 to 1.0.

rcw
gmw
byw Set the weight of each "rc", "gm", "by" option value. Default value
is 0.0. Allowed range is from 0.0 to 1.0. If all weights are 0.0
filtering is disabled.

pl Set the amount of preserving lightness. Default value is 0.0.
Allowed range is from 0.0 to 1.0.

Commands

This filter supports the all above options as commands.

colorcorrect
Adjust color white balance selectively for blacks and whites. This
filter operates in YUV colorspace.

The filter accepts the following options:

rl Set the red shadow spot. Allowed range is from -1.0 to 1.0.
Default value is 0.

bl Set the blue shadow spot. Allowed range is from -1.0 to 1.0.
Default value is 0.

rh Set the red highlight spot. Allowed range is from -1.0 to 1.0.
Default value is 0.

bh Set the blue highlight spot. Allowed range is from -1.0 to 1.0.
Default value is 0.

saturation
Set the amount of saturation. Allowed range is from -3.0 to 3.0.
Default value is 1.

analyze
If set to anything other than "manual" it will analyze every frame
and use derived parameters for filtering output frame.

Possible values are:

manual
average
minmax
median

Default value is "manual".

Commands

This filter supports the all above options as commands.

colorchannelmixer
Adjust video input frames by re-mixing color channels.

This filter modifies a color channel by adding the values associated to
the other channels of the same pixels. For example if the value to
modify is red, the output value will be:

The filter accepts the following options:

rr
rg
rb
ra Adjust contribution of input red, green, blue and alpha channels
for output red channel. Default is 1 for rr, and 0 for rg, rb and
ra.

gr
gg
gb
ga Adjust contribution of input red, green, blue and alpha channels
for output green channel. Default is 1 for gg, and 0 for gr, gb
and ga.

br
bg
bb
ba Adjust contribution of input red, green, blue and alpha channels
for output blue channel. Default is 1 for bb, and 0 for br, bg and
ba.

ar
ag
ab
aa Adjust contribution of input red, green, blue and alpha channels
for output alpha channel. Default is 1 for aa, and 0 for ar, ag
and ab.

Allowed ranges for options are "[-2.0, 2.0]".

pc Set preserve color mode. The accepted values are:

none
Disable color preserving, this is default.

lum Preserve luminance.

max Preserve max value of RGB triplet.

avg Preserve average value of RGB triplet.

sum Preserve sum value of RGB triplet.

nrm Preserve normalized value of RGB triplet.

pwr Preserve power value of RGB triplet.

pa Set the preserve color amount when changing colors. Allowed range
is from "[0.0, 1.0]". Default is 0.0, thus disabled.

Examples

• Convert source to grayscale:

colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3

• Simulate sepia tones:

colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131

Commands

This filter supports the all above options as commands.

colordetect
Analyze the video frames to determine the effective value range and
alpha mode.

The filter accepts the following options:

mode
Set of properties to detect. Unavailable properties, such as alpha
mode for an input image without an alpha channel, will be ignored
automatically.

Accepts a combination of the following flags:

color_range
Detect if the source contains luma pixels outside the limited
(MPEG) range, which indicates that this is a full range YUV
source.

alpha_mode
Detect if the source contains color values above the alpha
channel, which indicates that the alpha channel is independent
(straight), rather than premultiplied.

all Enable detection of all of the above properties. This is the
default.

colorize
Overlay a solid color on the video stream.

The filter accepts the following options:

hue Set the color hue. Allowed range is from 0 to 360. Default value
is 0.

saturation
Set the color saturation. Allowed range is from 0 to 1. Default
value is 0.5.

lightness
Set the color lightness. Allowed range is from 0 to 1. Default
value is 0.5.

mix Set the mix of source lightness. By default is set to 1.0. Allowed
range is from 0.0 to 1.0.

Commands

This filter supports the all above options as commands.

colorkey
RGB colorspace color keying. This filter operates on 8-bit RGB format
frames by setting the alpha component of each pixel which falls within
the similarity radius of the key color to 0. The alpha value for pixels
outside the similarity radius depends on the value of the blend option.

The filter accepts the following options:

color
Set the color for which alpha will be set to 0 (full transparency).
See "Color" section in the ffmpeg-utils manual. Default is
"black".

similarity
Set the radius from the key color within which other colors also
have full transparency. The computed distance is related to the
unit fractional distance in 3D space between the RGB values of the
key color and the pixel's color. Range is 0.01 to 1.0. 0.01 matches
within a very small radius around the exact key color, while 1.0
matches everything. Default is 0.01.

blend
Set how the alpha value for pixels that fall outside the similarity
radius is computed. 0.0 makes pixels either fully transparent or
fully opaque. Higher values result in semi-transparent pixels,
with greater transparency the more similar the pixel color is to
the key color. Range is 0.0 to 1.0. Default is 0.0.

Examples

• Make every green pixel in the input image transparent:

ffmpeg -i input.png -vf colorkey=green out.png

• Overlay a greenscreen-video on top of a static background image.

ffmpeg -i background.png -i video.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.flv

Commands

This filter supports same commands as options. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

colorhold
Remove all color information for all RGB colors except for certain one.

The filter accepts the following options:

color
The color which will not be replaced with neutral gray.

similarity
Similarity percentage with the above color. 0.01 matches only the
exact key color, while 1.0 matches everything.

blend
Blend percentage. 0.0 makes pixels fully gray. Higher values
result in more preserved color.

Commands

This filter supports same commands as options. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

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.

preserve
Set preserve color mode. The accepted values are:

none
Disable color preserving, this is default.

lum Preserve luminance.

max Preserve max value of RGB triplet.

avg Preserve average value of RGB triplet.

sum Preserve sum value of RGB triplet.

nrm Preserve normalized value of RGB triplet.

pwr Preserve power value of RGB triplet.

Examples

• Make video output darker:

colorlevels=rimin=0.058:gimin=0.058:bimin=0.058

• Increase contrast:

colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96

• Make video output lighter:

colorlevels=rimax=0.902:gimax=0.902:bimax=0.902

• Increase brightness:

colorlevels=romin=0.5:gomin=0.5:bomin=0.5

Commands

This filter supports the all above options as commands.

colormap
Apply custom color maps to video stream.

This filter needs three input video streams. First stream is video
stream that is going to be filtered out. Second and third video stream
specify color patches for source color to target color mapping.

The filter accepts the following options:

patch_size
Set the source and target video stream patch size in pixels.

nb_patches
Set the max number of used patches from source and target video
stream. Default value is number of patches available in additional
video streams. Max allowed number of patches is 64.

type
Set the adjustments used for target colors. Can be "relative" or
"absolute". Defaults is "absolute".

kernel
Set the kernel used to measure color differences between mapped
colors.

The accepted values are:

euclidean
weuclidean

Default is "euclidean".

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

colortemperature
Adjust color temperature in video to simulate variations in ambient
color temperature.

The filter accepts the following options:

temperature
Set the temperature in Kelvin. Allowed range is from 1000 to 40000.
Default value is 6500 K.

mix Set mixing with filtered output. Allowed range is from 0 to 1.
Default value is 1.

pl Set the amount of preserving lightness. Allowed range is from 0 to
1. Default value is 0.

Commands

This filter supports same commands as options.

convolution
Apply convolution of 3x3, 5x5, 7x7 or horizontal/vertical up to 49
elements.

The filter accepts the following options:

0m
1m
2m
3m Set matrix for each plane. Matrix is sequence of 9, 25 or 49
signed integers in square mode, and from 1 to 49 odd number of
signed integers in row mode.

0rdiv
1rdiv
2rdiv
3rdiv
Set multiplier for calculated value for each plane. If unset or 0,
it will be 1/sum of all matrix elements.

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.

0mode
1mode
2mode
3mode
Set matrix mode for each plane. Can be square, row or column.
Default is square.

Commands

This filter supports the all above options as commands.

Examples

• 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"

• 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"

• 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"

• 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"

• 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"

• 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

• List all filters available:

coreimage=list_filters=true

• Use the CIBoxBlur filter with default options to blur an image:

coreimage=filter=CIBoxBlur@default

• 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

• 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

corr
Obtain the correlation between two input videos.

This filter takes two input videos.

Both input videos 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 per component, average, min and max correlation is printed
through the logging system.

The filter stores the calculated correlation of each frame in frame
metadata.

This filter also supports the framesync options.

In the below example the input file main.mpg being processed is
compared with the reference file ref.mpg.

ffmpeg -i main.mpg -i ref.mpg -lavfi corr -f null -

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

• Cover a rectangular object by the supplied image of a given video
using ffmpeg:

ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv

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;
deprecated, do not use

t The timestamp expressed in seconds. It's NAN if the input timestamp
is unknown.

The expression for out_w may depend on the value of out_h, and the
expression for out_h may depend on out_w, but they cannot depend on x
and y, as x and y are evaluated after out_w and out_h.

The x and y parameters specify the expressions for the position of the
top-left corner of the output (non-cropped) area. They are evaluated
for each frame. If the evaluated value is not valid, it is approximated
to the nearest valid value.

The expression for x may depend on y, and the expression for y may
depend on x.

Examples

• Crop area with size 100x100 at position (12,34).

crop=100:100:12:34

Using named options, the example above becomes:

crop=w=100:h=100:x=12:y=34

• Crop the central input area with size 100x100:

crop=100:100

• Crop the central input area with size 2/3 of the input video:

crop=2/3*in_w:2/3*in_h

• Crop the input video central square:

crop=out_w=in_h
crop=in_h

• Delimit the rectangle with the top-left corner placed at position
100:100 and the right-bottom corner corresponding to the
right-bottom corner of the input image.

crop=in_w-100:in_h-100:100:100

• Crop 10 pixels from the left and right borders, and 20 pixels from
the top and bottom borders

crop=in_w-2*10:in_h-2*20

• Keep only the bottom right quarter of the input image:

crop=in_w/2:in_h/2:in_w/2:in_h/2

• Crop height for getting Greek harmony:

crop=in_w:1/PHI*in_w

• Apply trembling effect:

crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)

• Apply erratic camera effect depending on timestamp:

crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)

• Set x depending on the value of y:

crop=in_w/2:in_h/2:y:10+10*sin(n/10)

Commands

This filter supports the following commands:

w, out_w
h, out_h
x
y Set width/height of the output video and the horizontal/vertical
position in the input video. The command accepts the same syntax
of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

cropdetect
Auto-detect the crop size.

It calculates the necessary cropping parameters and prints the
recommended parameters via the logging system. The detected dimensions
correspond to the non-black or video area of the input video according
to mode.

It accepts the following parameters:

mode
Depending on mode crop detection is based on either the mere black
value of surrounding pixels or a combination of motion vectors and
edge pixels.

black
Detect black pixels surrounding the playing video. For fine
control use option limit.

mvedges
Detect the playing video by the motion vectors inside the video
and scanning for edge pixels typically forming the border of a
playing video.

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.

skip
Set the number of initial frames for which evaluation is skipped.
Default is 2. Range is 0 to INT_MAX.

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.

mv_threshold
Set motion in pixel units as threshold for motion detection. It
defaults to 8.

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 "5/255", and default value for high is
"15/255".

Examples

• Find video area surrounded by black borders:

ffmpeg -i file.mp4 -vf cropdetect,metadata=mode=print -f null -

• Find an embedded video area, generate motion vectors beforehand:

ffmpeg -i file.mp4 -vf mestimate,cropdetect=mode=mvedges,metadata=mode=print -f null -

• Find an embedded video area, use motion vectors from decoder:

ffmpeg -flags2 +export_mvs -i file.mp4 -vf cropdetect=mode=mvedges,metadata=mode=print -f null -

Commands

This filter supports the following commands:

limit
The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current
value.

cue
Delay video filtering until a given wallclock timestamp. The filter
first passes on preroll amount of frames, then it buffers at most
buffer amount of frames and waits for the cue. After reaching the cue
it forwards the buffered frames and also any subsequent frames coming
in its input.

The filter can be used synchronize the output of multiple ffmpeg
processes for realtime output devices like decklink. By putting the
delay in the filtering chain and pre-buffering frames the process can
pass on data to output almost immediately after the target wallclock
timestamp is reached.

Perfect frame accuracy cannot be guaranteed, but the result is good
enough for some use cases.

cue The cue timestamp expressed in a UNIX timestamp in microseconds.
Default is 0.

preroll
The duration of content to pass on as preroll expressed in seconds.
Default is 0.

buffer
The maximum duration of content to buffer before waiting for the
cue expressed in seconds. Default is 0.

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 will be defined to pass smoothly through all these new
coordinates. The new defined points need to be strictly increasing over
the x-axis, and their x and y values must be in the [0;1] interval. The
curve is formed by using a natural or monotonic cubic spline
interpolation, depending on the interp option (default: "natural"). The
"natural" spline produces a smoother curve in general while the
monotonic ("pchip") spline guarantees the transitions between the
specified points to be monotonic. 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.

interp
Specify the kind of interpolation. Available algorithms are:

natural
Natural cubic spline using a piece-wise cubic polynomial that
is twice continuously differentiable.

pchip
Monotonic cubic spline using a piecewise cubic Hermite
interpolating polynomial (PCHIP).

To avoid some filtergraph syntax conflicts, each key points list need
to be defined using the following syntax: "x0/y0 x1/y1 x2/y2 ...".

Commands

This filter supports same commands as options.

Examples

• Increase slightly the middle level of blue:

curves=blue='0/0 0.5/0.58 1/1'

• 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)"

• The previous example can also be achieved with the associated
built-in preset:

curves=preset=vintage

• Or simply:

curves=vintage

• Use a Photoshop preset and redefine the points of the green
component:

curves=psfile='MyCurvesPresets/purple.acv':green='0/0 0.45/0.53 1/1'

• 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.

format
Set display number format. Can be "hex", or "dec". Default is
"hex".

components
Set pixel components to display. By default all pixel components
are displayed.

Commands

This filter supports same commands as options excluding "size" option.

dblur
Apply Directional blur filter.

The filter accepts the following options:

angle
Set angle of directional blur. Default is 45.

radius
Set radius of directional blur. Default is 5.

planes
Set which planes to filter. By default all planes are filtered.

Commands

This filter supports same commands as options. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

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.

Commands

This filter supports the all above options as commands.

deblock
Remove blocking artifacts from input video.

The filter accepts the following options:

filter
Set filter type, can be weak or strong. Default is strong. This
controls what kind of deblocking is applied.

block
Set size of block, allowed range is from 4 to 512. Default is 8.

alpha
beta
gamma
delta
Set blocking detection thresholds. Allowed range is 0 to 1.
Defaults are: 0.098 for alpha and 0.05 for the rest. Using higher
threshold gives more deblocking strength. Setting alpha controls
threshold detection at exact edge of block. Remaining options
controls threshold detection near the edge. Each one for
below/above or left/right. Setting any of those to 0 disables
deblocking.

planes
Set planes to filter. Default is to filter all available planes.

Examples

• Deblock using weak filter and block size of 4 pixels.

deblock=filter=weak:block=4

• Deblock using strong filter, block size of 4 pixels and custom
thresholds for deblocking more edges.

deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05

• Similar as above, but filter only first plane.

deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05:planes=1

• Similar as above, but filter only second and third plane.

deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05:planes=6

Commands

This filter supports the all above options as commands.

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.

mixed
Set whether or not the input only partially contains content to be
decimated. Default is "false". If enabled video output stream
will be in variable frame rate.

deconvolve
Apply 2D deconvolution 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.

noise
Set noise when doing divisions. Default is 0.0000001. Useful when
width and height are not same and not power of 2 or if stream prior
to convolving had noise.

The "deconvolve" filter also supports the framesync options.

dedot
Reduce cross-luminance (dot-crawl) and cross-color (rainbows) from
video.

It accepts the following options:

m Set mode of operation. Can be combination of dotcrawl for
cross-luminance reduction and/or rainbows for cross-color
reduction.

lt Set spatial luma threshold. Lower values increases reduction of
cross-luminance.

tl Set tolerance for temporal luma. Higher values increases reduction
of cross-luminance.

tc Set tolerance for chroma temporal variation. Higher values
increases reduction of cross-color.

ct Set temporal chroma threshold. Lower values increases reduction of
cross-color.

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.

Commands

This filter supports the all above options as commands.

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.

show
When set to 1, a green rectangle is drawn on the screen to simplify
finding the right x, y, w, and h parameters. The default value is
0.

The rectangle is drawn on the outermost pixels which will be
(partly) replaced with interpolated values. The values of the next
pixels immediately outside this rectangle in each direction will be
used to compute the interpolated pixel values inside the rectangle.

Examples

• Set a rectangle covering the area with top left corner coordinates
0,0 and size 100x77:

delogo=x=0:y=0:w=100:h=77

derain
Remove the rain in the input image/video by applying the derain methods
based on convolutional neural networks. Supported models:

• Recurrent Squeeze-and-Excitation Context Aggregation Net (RESCAN).
See
<http://openaccess.thecvf.com/content_ECCV_2018/papers/Xia_Li_Recurrent_Squeeze-and-Excitation_Context_ECCV_2018_paper.pdf>.

Training as well as model generation scripts are provided in the
repository at <https://github.com/XueweiMeng/derain_filter.git>.

The filter accepts the following options:

filter_type
Specify which filter to use. This option accepts the following
values:

derain
Derain filter. To conduct derain filter, you need to use a
derain model.

dehaze
Dehaze filter. To conduct dehaze filter, you need to use a
dehaze model.

Default value is derain.

dnn_backend
Specify which DNN backend to use for model loading and execution.
This option accepts the following values:

tensorflow
TensorFlow backend. To enable this backend you need to install
the TensorFlow for C library (see
<https://www.tensorflow.org/install/lang_c>) and configure
FFmpeg with "--enable-libtensorflow"

model
Set path to model file specifying network architecture and its
parameters. Note that different backends use different file
formats. TensorFlow can load files for only its format.

To get full functionality (such as async execution), please use the
dnn_processing filter.

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.

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.

Commands

This filter supports the all above options as commands.

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

Commands

This filter supports the all above options as commands.

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

• 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

• 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

dnn_classify
Do classification with deep neural networks based on bounding boxes.

The filter accepts the following options:

dnn_backend
Specify which DNN backend to use for model loading and execution.
This option accepts only openvino now, tensorflow backends will be
added.

model
Set path to model file specifying network architecture and its
parameters. Note that different backends use different file
formats.

input
Set the input name of the dnn network.

output
Set the output name of the dnn network.

confidence
Set the confidence threshold (default: 0.5).

labels
Set path to label file specifying the mapping between label id and
name. Each label name is written in one line, tailing spaces and
empty lines are skipped. The first line is the name of label id 0,
and the second line is the name of label id 1, etc. The label id
is considered as name if the label file is not provided.

backend_configs
Set the configs to be passed into backend

For tensorflow backend, you can set its configs with sess_config
options, please use tools/python/tf_sess_config.py to get the
configs for your system.

dnn_detect
Do object detection with deep neural networks.

The filter accepts the following options:

dnn_backend
Specify which DNN backend to use for model loading and execution.
This option accepts only openvino now, tensorflow backends will be
added.

model
Set path to model file specifying network architecture and its
parameters. Note that different backends use different file
formats.

input
Set the input name of the dnn network.

output
Set the output name of the dnn network.

confidence
Set the confidence threshold (default: 0.5).

labels
Set path to label file specifying the mapping between label id and
name. Each label name is written in one line, tailing spaces and
empty lines are skipped. The first line is the name of label id 0
(usually it is 'background'), and the second line is the name of
label id 1, etc. The label id is considered as name if the label
file is not provided.

backend_configs
Set the configs to be passed into backend. To use async execution,
set async (default: set). Roll back to sync execution if the
backend does not support async.

dnn_processing
Do image processing with deep neural networks. It works together with
another filter which converts the pixel format of the Frame to what the
dnn network requires.

The filter accepts the following options:

dnn_backend
Specify which DNN backend to use for model loading and execution.
This option accepts the following values:

openvino
OpenVINO backend. To enable this backend you need to build and
install the OpenVINO for C library (see
<https://github.com/openvinotoolkit/openvino/blob/master/build-instruction.md>)
and configure FFmpeg with "--enable-libopenvino"
(--extra-cflags=-I... --extra-ldflags=-L... might be needed if
the header files and libraries are not installed into system
path)

torch
Libtorch backend. To enable this backend you need to build and
install Libtroch for C++ library. Please download cxx11 ABI
version (see <https://pytorch.org/get-started/locally>) and
configure FFmpeg with "--enable-libtorch
--extra-cflags=-I/libtorch_root/libtorch/include
--extra-cflags=-I/libtorch_root/libtorch/include/torch/csrc/api/include
--extra-ldflags=-L/libtorch_root/libtorch/lib/"

model
Set path to model file specifying network architecture and its
parameters. Note that different backends use different file
formats. TensorFlow, OpenVINO and Libtorch backend can load files
for only its format.

input
Set the input name of the dnn network.

output
Set the output name of the dnn network.

backend_configs
Set the configs to be passed into backend. To use async execution,
set async (default: set). Roll back to sync execution if the
backend does not support async.

For tensorflow backend, you can set its configs with sess_config
options, please use tools/python/tf_sess_config.py to get the
configs of TensorFlow backend for your system.

Examples

• Remove rain in rgb24 frame with can.pb (see derain filter):

./ffmpeg -i rain.jpg -vf format=rgb24,dnn_processing=dnn_backend=tensorflow:model=can.pb:input=x:output=y derain.jpg

• Handle the Y channel with srcnn.pb (see sr filter) for frame with
yuv420p (planar YUV formats supported):

./ffmpeg -i 480p.jpg -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=tensorflow:model=srcnn.pb:input=x:output=y -y srcnn.jpg

• Handle the Y channel with espcn.pb (see sr filter), which changes
frame size, for format yuv420p (planar YUV formats supported),
please use tools/python/tf_sess_config.py to get the configs of
TensorFlow backend for your system.

./ffmpeg -i 480p.jpg -vf format=yuv420p,dnn_processing=dnn_backend=tensorflow:model=espcn.pb:input=x:output=y:backend_configs=sess_config=0x10022805320e09cdccccccccccec3f20012a01303801 -y tmp.espcn.jpg

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. A value
of "fill" will create a filled box. Default value is 3.

See below for the list of accepted constants.

replace
Applicable if the input has alpha. With value 1, the pixels of the
painted box will overwrite the video's color and alpha pixels.
Default is 0, which composites the box onto the input, leaving the
video's alpha intact.

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.

box_source
Box source can be set as side_data_detection_bboxes if you want to
use box data in detection bboxes of side data.

If box_source is set, the x, y, width and height will be ignored
and still use box data in detection bboxes of side data. So please
do not use this parameter if you were not sure about the box
source.

t The thickness of the drawn box.

These constants allow the x, y, w, h and t expressions to refer to
each other, so you may for example specify "y=x/dar" or "h=w/dar".

Examples

• Draw a black box around the edge of the input image:

drawbox

• Draw a box with color red and an opacity of 50%:

drawbox=10:20:200:60:red@0.5

The previous example can be specified as:

drawbox=x=10:y=20:w=200:h=60:color=red@0.5

• Fill the box with pink color:

drawbox=x=10:y=10:w=100:h=100:color=pink@0.5:t=fill

• 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

Commands

This filter supports same commands as options. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

drawgraph
Draw a graph using input video 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".

rate, r
Set the output frame rate. Default value is 25.

The foreground color expressions can use the following variables:

MIN Minimal value of metadata value.

MAX Maximal value of metadata value.

VAL Current metadata key value.

The color is defined as 0xAABBGGRR.

Example using metadata from signalstats filter:

signalstats,drawgraph=lavfi.signalstats.YAVG:min=0:max=255

Example using metadata from ebur128 filter:

ebur128=metadata=1,adrawgraph=lavfi.r128.M:min=-120:max=5

drawgrid
Draw a grid on the input image.

It accepts the following parameters:

x
y The expressions which specify the coordinates of some point of grid
intersection (meant to configure offset). Both default to 0.

width, w
height, h
The expressions which specify the width and height of the grid
cell, if 0 they are interpreted as the input width and height,
respectively, minus "thickness", so image gets framed. Default to
0.

color, c
Specify the color of the grid. For the general syntax of this
option, check the "Color" section in the ffmpeg-utils manual. If
the special value "invert" is used, the grid color is the same as
the video with inverted luma.

thickness, t
The expression which sets the thickness of the grid line. Default
value is 1.

See below for the list of accepted constants.

replace
Applicable if the input has alpha. With 1 the pixels of the painted
grid will overwrite the video's color and alpha pixels. Default is
0, which composites the grid onto the input, leaving the video's
alpha intact.

The parameters for x, y, w and h and t are expressions containing the
following constants:

dar The input display aspect ratio, it is the same as (w / h) * sar.

hsub
vsub
horizontal and vertical chroma subsample values. For example for
the pixel format "yuv422p" hsub is 2 and vsub is 1.

in_h, ih
in_w, iw
The input grid cell width and height.

sar The input sample aspect ratio.

x
y The x and y coordinates of some point of grid intersection (meant
to configure offset).

w
h The width and height of the drawn cell.

t The thickness of the drawn cell.

These constants allow the x, y, w, h and t expressions to refer to
each other, so you may for example specify "y=x/dar" or "h=w/dar".

Examples

• Draw a grid with cell 100x100 pixels, thickness 2 pixels, with
color red and an opacity of 50%:

drawgrid=width=100:height=100:thickness=2:color=red@0.5

• Draw a white 3x3 grid with an opacity of 50%:

drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5

Commands

This filter supports same commands as options. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

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" and "--enable-libharfbuzz". 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 value must be specified using one of the following
formats:

*<"boxborderw=10" set the width of all the borders to 10>
*<"boxborderw=10|20" set the width of the top and bottom borders to
10>
and the width of the left and right borders to 20

*<"boxborderw=10|20|30" set the width of the top border to 10, the
width>
of the bottom border to 30 and the width of the left and right borders to 20

*<"boxborderw=10|20|30|40" set the borders width to 10 (top), 20
(right),>
30 (bottom), 40 (left)

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. The default value of line_spacing
is 0.

text_align
Set the vertical and horizontal alignment of the text with respect
to the box boundaries. The value is combination of flags, one for
the vertical alignment (T=top, M=middle, B=bottom) and one for the
horizontal alignment (L=left, C=center, R=right). Please note that
tab characters are only supported with the left horizontal
alignment.

y_align
Specify what the y value is referred to. Possible values are:

*<"text" the top of the highest glyph of the first text line is
placed at y>
*<"baseline" the baseline of the first text line is placed at y>
*<"font" the baseline of the first text line is placed at y plus
the>
ascent (in pixels) defined in the font metrics

The default value of y_align is "text" for backward compatibility.

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".

boxw
Set the width of the box to be drawn around text. The default
value of boxw is computed automatically to match the text width

boxh
Set the height of the box to be drawn around text. The default
value of boxh is computed automatically to match the text height

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). Value will be rounded
to nearest integer. Minimum value is "1". Drop-frame timecode is
supported for frame rates 30 & 60.

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.

text_source
Text source should be set as side_data_detection_bboxes if you want
to use text data in detection bboxes of side data.

If text source is set, text and textfile will be ignored and still
use text data in detection bboxes of side data. So please do not
use this parameter if you are not sure about the text source.

reload
The textfile will be reloaded at specified frame interval. Be sure
to update textfile atomically, or it may be read partially, or even
fail. Range is 0 to INT_MAX. Default is 0.

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

font_a
the ascent size defined in the font metrics

font_d
the descent size defined in the font metrics

top_a
the maximum ascender of the glyphs of the first text line

bottom_d
the maximum descender of the glyphs of the last text line

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 to each
other, so you can for example specify "y=x/dar".

pict_type
A one character description of the current frame's picture type.

pkt_pos
The current packet's position in the input file or stream (in
bytes, from the start of the input). A value of -1 indicates this
info is not available.

duration
The current packet's duration, in seconds.

pkt_size
The current packet's size (in bytes).

Text expansion

If expansion is set to "strftime", the filter recognizes sequences
accepted by the "strftime" C function in the provided text and expands
them accordingly. Check the documentation of "strftime". This feature
is deprecated in favor of "normal" expansion with the "gmtime" or
"localtime" expansion functions.

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 with the textfile option
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" C function format string. The
format string is extended to support the variable %[1-6]N which
prints fractions of the second with optionally specified number of
digits.

localtime
The time at which the filter is running, expressed in the local
time zone. It can accept an argument: a "strftime" C function
format string. The format string is extended to support the
variable %[1-6]N which prints fractions of the second with
optionally specified number of digits.

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.

Available metadata can be identified by inspecting entries starting
with TAG included within each frame section printed by running
"ffprobe -show_frames".

String metadata generated in filters leading to the drawtext filter
are also available.

n, frame_num
The frame number, starting from 0.

pict_type
A one 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 "hms", a third argument "24HH" may be
supplied to present the hour part of the formatted timestamp in 24h
format (00-23).

If the format is set to "localtime" or "gmtime", a third argument
may be supplied: a "strftime" C function format string. By
default, YYYY-MM-DD HH:MM:SS format will be used.

Commands

This filter supports altering parameters via commands:

reinit
Alter existing filter parameters.

Syntax for the argument is the same as for filter invocation, e.g.

fontsize=56:fontcolor=green:text='Hello World'

Full filter invocation with sendcmd would look like this:

sendcmd=c='56.0 drawtext reinit fontsize=56\:fontcolor=green\:text=Hello\\ World'

If the entire argument can't be parsed or applied as valid values
then the filter will continue with its existing parameters.

The following options are also supported as commands:

*<x>
*<y>
*<alpha>
*<fontsize>
*<fontcolor>
*<boxcolor>
*<bordercolor>
*<shadowcolor>
*<box>
*<boxw>
*<boxh>
*<boxborderw>
*<line_spacing>
*<text_align>
*<shadowx>
*<shadowy>
*<borderw>

Examples

• Draw "Test Text" with font FreeSerif, using the default values for
the optional parameters.

drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'"

• Draw 'Test Text' with font FreeSerif of size 24 at position x=100
and y=50 (counting from the top-left corner of the screen), text is
yellow with a red box around it. Both the text and the box have an
opacity of 20%.

drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\
x=100: y=50: fontsize=24: fontcolor=yellow@0.2: box=1: boxcolor=red@0.2"

Note that the double quotes are not necessary if spaces are not
used within the parameter list.

• Show the text at the center of the video frame:

drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2"

• Show 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)"

• Show a text line sliding from right to left in the last row of the
video frame. The file LONG_LINE is assumed to contain a single line
with no newlines.

drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t"

• Show the content of file CREDITS off the bottom of the frame and
scroll up.

drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t"

• Draw a single green letter "g", at the center of the input video.
The glyph baseline is placed at half screen height.

drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent"

• Show text for 1 second every 3 seconds:

drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'"

• Use fontconfig to set the font. Note that the colons need to be
escaped.

drawtext='fontfile=Linux Libertine O-40\\:style=Semibold:text=FFmpeg'

• Draw "Test Text" with font size dependent on height of the video.

drawtext="text='Test Text': fontsize=h/30: x=(w-text_w)/2: y=(h-text_h*2)"

• Print the date of a real-time encoding (see documentation for the
"strftime" C function):

drawtext='fontfile=FreeSans.ttf:text=%{localtime\:%a %b %d %Y}'

• Show text fading in and out (appearing/disappearing):

#!/bin/sh
DS=1.0 # display start
DE=10.0 # display end
FID=1.5 # fade in duration
FOD=5 # fade out duration
ffplay -f lavfi "color,drawtext=text=TEST:fontsize=50:fontfile=FreeSerif.ttf:fontcolor_expr=ff0000%{eif\\\\: clip(255*(1*between(t\\, $DS + $FID\\, $DE - $FOD) + ((t - $DS)/$FID)*between(t\\, $DS\\, $DS + $FID) + (-(t - $DE)/$FOD)*between(t\\, $DE - $FOD\\, $DE) )\\, 0\\, 255) \\\\: x\\\\: 2 }"

• 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

• Plot special lavf.image2dec.source_basename metadata onto each
frame if such metadata exists. Otherwise, plot the string "NA".
Note that image2 demuxer must have option -export_path_metadata 1
for the special metadata fields to be available for filters.

drawtext="fontsize=20:fontcolor=white:fontfile=FreeSans.ttf:text='%{metadata\:lavf.image2dec.source_basename\:NA}':x=10:y=10"

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/>.

For more information about libharfbuzz, check:
<https://github.com/harfbuzz/harfbuzz>.

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.

canny
Apply Canny edge detector on all selected planes.

Default value is wires.

planes
Select planes for filtering. By default all available planes are
filtered.

Examples

• Standard edge detection with custom values for the hysteresis
thresholding:

edgedetect=low=0.1:high=0.4

• Painting effect without thresholding:

edgedetect=mode=colormix:high=0

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. Default is disabled.

use_alpha
Include alpha values in the quantization calculation. Allows
creating palettized output images (e.g. PNG8) with multiple alpha
smooth blending.

entropy
Measure graylevel entropy in histogram of color channels of video
frames.

It accepts the following parameters:

mode
Can be either normal or diff. Default is normal.

diff mode measures entropy of histogram delta values, absolute
differences between neighbour histogram values.

epx
Apply the EPX magnification filter which is designed for pixel art.

It accepts the following option:

n Set the scaling dimension: 2 for "2xEPX", 3 for "3xEPX". Default
is 3.

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 -1000.0 to 1000.0. 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; deprecated, do not use

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

Commands

This filter supports the all above options as commands.

estdif
Deinterlace the input video ("estdif" stands for "Edge Slope Tracing
Deinterlacing Filter").

Spatial only filter that uses edge slope tracing algorithm to
interpolate missing lines. It accepts the following parameters:

mode
The interlacing mode to adopt. It accepts one of the following
values:

frame
Output one frame for each frame.

field
Output one frame for each field.

The default value is "field".

parity
The picture field parity assumed for the input interlaced video. It
accepts one of the following values:

tff Assume the top field is first.

bff Assume the bottom field is first.

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. Accepts one of the following
values:

all Deinterlace all frames.

interlaced
Only deinterlace frames marked as interlaced.

The default value is "all".

rslope
Specify the search radius for edge slope tracing. Default value is
1. Allowed range is from 1 to 15.

redge
Specify the search radius for best edge matching. Default value is
2. Allowed range is from 0 to 15.

ecost
Specify the edge cost for edge matching. Default value is 2.
Allowed range is from 0 to 50.

mcost
Specify the middle cost for edge matching. Default value is 1.
Allowed range is from 0 to 50.

dcost
Specify the distance cost for edge matching. Default value is 1.
Allowed range is from 0 to 50.

interp
Specify the interpolation used. Default is 4-point interpolation.
It accepts one of the following values:

2p Two-point interpolation.

4p Four-point interpolation.

6p Six-point interpolation.

Commands

This filter supports same commands as options.

exposure
Adjust exposure of the video stream.

The filter accepts the following options:

exposure
Set the exposure correction in EV. Allowed range is from -3.0 to
3.0 EV Default value is 0 EV.

black
Set the black level correction. Allowed range is from -1.0 to 1.0.
Default value is 0.

Commands

This filter supports same commands as options.

extractplanes
Extract color channel components from input video stream into separate
grayscale video streams.

The filter accepts the following option:

planes
Set plane(s) to extract.

Available values for planes are:

y
u
v
a
r
g
b

Choosing planes not available in the input will result in an error.
That means you cannot select "r", "g", "b" planes with "y", "u",
"v" planes at same time.

Examples

• Extract luma, u and v color channel component from input video
frame into 3 grayscale outputs:

ffmpeg -i video.avi -filter_complex 'extractplanes=y+u+v[y][u][v]' -map '[y]' y.avi -map '[u]' u.avi -map '[v]' v.avi

fade
Apply a fade-in/out effect to the input video.

It accepts the following parameters:

type, t
The effect type can be either "in" for a fade-in, or "out" for a
fade-out effect. Default is "in".

start_frame, s
Specify the number of the frame to start applying the fade effect
at. Default is 0.

nb_frames, n
The number of frames that the fade effect lasts. At the end of the
fade-in effect, the output video will have the same intensity as
the input video. At the end of the fade-out transition, the output
video will be filled with the selected color. Default is 25.

alpha
If set to 1, fade only alpha channel, if one exists on the input.
Default value is 0.

start_time, st
Specify the timestamp (in seconds) of the frame to start to apply
the fade effect. If both start_frame and start_time are specified,
the fade will start at whichever comes last. Default is 0.

duration, d
The number of seconds for which the fade effect has to last. At the
end of the fade-in effect the output video will have the same
intensity as the input video, at the end of the fade-out transition
the output video will be filled with the selected color. If both
duration and nb_frames are specified, duration is used. Default is
0 (nb_frames is used by default).

color, c
Specify the color of the fade. Default is "black".

Examples

• Fade in the first 30 frames of video:

fade=in:0:30

The command above is equivalent to:

fade=t=in:s=0:n=30

• Fade out the last 45 frames of a 200-frame video:

fade=out:155:45
fade=type=out:start_frame=155:nb_frames=45

• Fade in the first 25 frames and fade out the last 25 frames of a
1000-frame video:

fade=in:0:25, fade=out:975:25

• Make the first 5 frames yellow, then fade in from frame 5-24:

fade=in:5:20:color=yellow

• Fade in alpha over first 25 frames of video:

fade=in:0:25:alpha=1

• Make the first 5.5 seconds black, then fade in for 0.5 seconds:

fade=t=in:st=5.5:d=0.5

feedback
Apply feedback video filter.

This filter pass cropped input frames to 2nd output. From there it can
be filtered with other video filters. After filter receives frame from
2nd input, that frame is combined on top of original frame from 1st
input and passed to 1st output.

The typical usage is filter only part of frame.

The filter accepts the following options:

x
y Set the top left crop position.

w
h Set the crop size.

Examples

• Blur only top left rectangular part of video frame size 100x100
with gblur filter.

[in][blurin]feedback=x=0:y=0:w=100:h=100[out][blurout];[blurout]gblur=8[blurin]

• Draw black box on top left part of video frame of size 100x100 with
drawbox filter.

[in][blurin]feedback=x=0:y=0:w=100:h=100[out][blurout];[blurout]drawbox=x=0:y=0:w=100:h=100:t=100[blurin]

• Pixelize rectangular part of video frame of size 100x100 with
pixelize filter.

[in][blurin]feedback=x=320:y=240:w=100:h=100[out][blurout];[blurout]pixelize[blurin]

fftdnoiz
Denoise frames using 3D FFT (frequency domain filtering).

The filter accepts the following options:

sigma
Set the noise sigma constant. This sets denoising strength.
Default value is 1. Allowed range is from 0 to 30. Using very high
sigma with low overlap may give blocking artifacts.

amount
Set amount of denoising. By default all detected noise is reduced.
Default value is 1. Allowed range is from 0 to 1.

block
Set size of block in pixels, Default is 32, can be 8 to 256.

overlap
Set block overlap. Default is 0.5. Allowed range is from 0.2 to
0.8.

method
Set denoising method. Default is "wiener", can also be "hard".

prev
Set number of previous frames to use for denoising. By default is
set to 0.

next
Set number of next frames to to use for denoising. By default is
set to 0.

planes
Set planes which will be filtered, by default are all available
filtered except alpha.

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.

WS
HS The size of FFT array for horizontal and vertical processing.

Examples

• High-pass:

fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'

• Low-pass:

fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'

• Sharpen:

fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'

• 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
optionally followed by "t" output frame will use only top field, or
in case of "b" it will use only bottom field. If line starts with
"#" or ";" that line is skipped.

mode
Can be item "absolute" or "relative" or "pattern". Default is
"absolute". The "pattern" mode is same as "relative" mode, except
at last entry of file if there are more frames to process than
"hint" file is seek back to start.

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

fillborders
Fill borders of the input video, without changing video stream
dimensions. Sometimes video can have garbage at the four edges and you
may not want to crop video input to keep size multiple of some number.

This filter accepts the following options:

left
Number of pixels to fill from left border.

right
Number of pixels to fill from right border.

top Number of pixels to fill from top border.

bottom
Number of pixels to fill from bottom border.

mode
Set fill mode.

It accepts the following values:

smear
fill pixels using outermost pixels

mirror
fill pixels using mirroring (half sample symmetric)

fixed
fill pixels with constant value

reflect
fill pixels using reflecting (whole sample symmetric)

wrap
fill pixels using wrapping

fade
fade pixels to constant value

margins
fill pixels at top and bottom with weighted averages pixels
near borders

Default is smear.

color
Set color for pixels in fixed or fade mode. Default is black.

Commands

This filter supports same commands as options. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

find_rect
Find a rectangular object in the input video.

The object to search for must be specified as a gray8 image specified
with the object option.

For each possible match, a score is computed. If the score reaches the
specified threshold, the object is considered found.

If the input video contains multiple instances of the object, the
filter will find only one of them.

When an object is found, the following metadata entries are set in the
matching frame:

lavfi.rect.w
width of object

lavfi.rect.h
height of object

lavfi.rect.x
x position of object

lavfi.rect.y
y position of object

lavfi.rect.score
match score of the found object

It accepts the following options:

object
Filepath of the object image, needs to be in gray8.

threshold
Detection threshold, expressed as a decimal number in the range
0-1.

A threshold value of 0.01 means only exact matches, a threshold of
0.99 means almost everything matches.

Default value is 0.5.

mipmaps
Number of mipmaps, default is 3.

xmin, ymin, xmax, ymax
Specifies the rectangle in which to search.

discard
Discard frames where object is not detected. Default is disabled.

Examples

• Cover a rectangular object by the supplied image of a given video
using ffmpeg:

ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv

• Find the position of an object in each frame using ffprobe and
write it to a log file:

ffprobe -f lavfi movie=test.mp4,find_rect=object=object.pgm:threshold=0.3 \
-show_entries frame=pkt_pts_time:frame_tags=lavfi.rect.x,lavfi.rect.y \
-of csv -o find_rect.csv

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".

color_spaces
A '|'-separated list of color space names, such as
"color_spaces=bt709|bt470bg|bt2020nc".

color_ranges
A '|'-separated list of color range names, such as
"color_ranges=tv|pc".

Examples

• Convert the input video to the yuv420p format

format=pix_fmts=yuv420p

Convert the input video to any of the formats in the list

format=pix_fmts=yuv420p|yuv444p|yuv410p

fps
Convert the video to specified constant frame rate by duplicating or
dropping frames as necessary.

It accepts the following parameters:

fps The desired output frame rate. It accepts expressions containing
the following constants:

source_fps
The input's frame rate

ntsc
NTSC frame rate of "30000/1001"

pal PAL frame rate of 25.0

film
Film frame rate of 24.0

ntsc_film
NTSC-film frame rate of "24000/1001"

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

• A typical usage in order to set the fps to 25:

fps=fps=25

• Sets the fps to 24, using abbreviation and rounding method to round
to nearest:

fps=fps=film:round=near

framepack
Pack two different video streams into a stereoscopic video, setting
proper metadata on supported codecs. The two views should have the same
size and framerate and processing will stop when the shorter video
ends. Please note that you may conveniently adjust view properties with
the scale and fps filters.

It accepts the following parameters:

format
The desired packing format. Supported values are:

sbs The views are next to each other (default).

tab The views are on top of each other.

lines
The views are packed by line.

columns
The views are packed by column.

frameseq
The views are temporally interleaved.

Some examples:

# Convert left and right views into a frame-sequential video
ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT

# Convert views into a side-by-side video with the same output resolution as the input
ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT

framerate
Change the frame rate by interpolating new video output frames from the
source frames.

This filter is not designed to function correctly with interlaced
media. If you wish to change the frame rate of interlaced media then
you are required to deinterlace before this filter and re-interlace
after this filter.

A description of the accepted options follows.

fps Specify the output frames per second. This option can also be
specified as a value alone. The default is 50.

interp_start
Specify the start of a range where the output frame will be created
as a linear interpolation of two frames. The range is [0-255], the
default is 15.

interp_end
Specify the end of a range where the output frame will be created
as a linear interpolation of two frames. The range is [0-255], the
default is 240.

scene
Specify the level at which a scene change is detected as a value
between 0 and 100 to indicate a new scene; a low value reflects a
low probability for the current frame to introduce a new scene,
while a higher value means the current frame is more likely to be
one. The default is 8.2.

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.

freezedetect
Detect frozen video.

This filter logs a message and sets frame metadata when it detects that
the input video has no significant change in content during a specified
duration. Video freeze detection calculates the mean average absolute
difference of all the components of video frames and compares it to a
noise floor.

The printed times and duration are expressed in seconds. The
"lavfi.freezedetect.freeze_start" metadata key is set on the first
frame whose timestamp equals or exceeds the detection duration and it
contains the timestamp of the first frame of the freeze. The
"lavfi.freezedetect.freeze_duration" and
"lavfi.freezedetect.freeze_end" metadata keys are set on the first
frame after the freeze.

The filter accepts the following options:

noise, n
Set noise tolerance. Can be specified in dB (in case "dB" is
appended to the specified value) or as a difference ratio between 0
and 1. Default is -60dB, or 0.001.

duration, d
Set freeze duration until notification (default is 2 seconds).

freezeframes
Freeze video frames.

This filter freezes video frames using frame from 2nd input.

The filter accepts the following options:

first
Set number of first frame from which to start freeze.

last
Set number of last frame from which to end freeze.

replace
Set number of frame from 2nd input which will be used instead of
replaced frames.

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 a color
description as specified in the "Color" section in the ffmpeg-utils
manual, a position (specified as X/Y, where X and Y are floating point
numbers) and/or a string.

The number and types of parameters depend on the loaded effect. If an
effect parameter is not specified, the default value is set.

Examples

• Apply the distort0r effect, setting the first two double
parameters:

frei0r=filter_name=distort0r:filter_params=0.5|0.01

• Apply the colordistance effect, taking a color as the first
parameter:

frei0r=colordistance:0.2/0.3/0.4
frei0r=colordistance:violet
frei0r=colordistance:0x112233

• Apply the perspective effect, specifying the top left and top right
image positions:

frei0r=perspective:0.2/0.2|0.8/0.2

For more information, see <http://frei0r.dyne.org>

Commands

This filter supports the filter_params option as commands.

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 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).

fsync
Synchronize video frames with an external mapping from a file.

For each input PTS given in the map file it either drops or creates as
many frames as necessary to recreate the sequence of output frames
given in the map file.

This filter is useful to recreate the output frames of a framerate
conversion by the fps filter, recorded into a map file using the ffmpeg
option "-stats_mux_pre", and do further processing to the corresponding
frames e.g. quality comparison.

Each line of the map file must contain three items per input frame, the
input PTS (decimal), the output PTS (decimal) and the output TIMEBASE
(decimal/decimal), separated by a space. This file format corresponds
to the output of "-stats_mux_pre_fmt="{ptsi} {pts} {tb}"".

The filter assumes the map file is sorted by increasing input PTS.

The filter accepts the following options:

file, f
The filename of the map file to be used.

Example:

# Convert a video to 25 fps and record a MAP_FILE file with the default format of this filter
ffmpeg -i INPUT -vf fps=fps=25 -stats_mux_pre MAP_FILE -stats_mux_pre_fmt "{ptsi} {pts} {tb}" OUTPUT

# Sort MAP_FILE by increasing input PTS
sort -n MAP_FILE

# Use INPUT, OUTPUT and the MAP_FILE from above to compare the corresponding frames in INPUT and OUTPUT via SSIM
ffmpeg -i INPUT -i OUTPUT -filter_complex '[0:v]fsync=file=MAP_FILE[ref];[1:v][ref]ssim' -f null -

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. Default 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.

Commands

This filter supports same commands as options. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

geq
Apply generic equation to each pixel.

The filter accepts the following options:

lum_expr, lum
Set the luma 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 luma 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 luma 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.

psum(x,y), lumsum(x, y), cbsum(x,y), crsum(x,y), rsum(x,y), gsum(x,y),
bsum(x,y), alphasum(x,y)
Sum of sample values in the rectangle from (0,0) to (x,y), this
allows obtaining sums of samples within a rectangle. See the
functions without the sum postfix.

interpolation
Set one of interpolation methods:

nearest, n
bilinear, b

Default is bilinear.

For functions, if x and y are outside the area, the value will be
automatically clipped to the closer edge.

Please note that this filter can use multiple threads in which case
each slice will have its own expression state. If you want to use only
a single expression state because your expressions depend on previous
state then you should limit the number of filter threads to 1.

Examples

• Flip the image horizontally:

geq=p(W-X\,Y)

• Generate a bidimensional sine wave, with angle "PI/3" and a
wavelength of 100 pixels:

geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128

• Generate a fancy enigmatic moving light:

nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128

• Generate a quick emboss effect:

format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2'

• Modify RGB components depending on pixel position:

geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)'

• Create a radial gradient that is the same size as the input (also
see the vignette filter):

geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray

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

• Apply the filter with a 3.5 strength and radius of 8:

gradfun=3.5:8

• Specify radius, omitting the strength (which will fall-back to the
default value):

gradfun=radius=8

graphmonitor
Show various filtergraph stats.

With this filter one can debug complete filtergraph. Especially issues
with links filling with queued frames.

The filter accepts the following options:

size, s
Set video output size. Default is hd720.

opacity, o
Set video opacity. Default is 0.9. Allowed range is from 0 to 1.

mode, m
Set output mode flags.

Available values for flags are:

full
No any filtering. Default.

compact
Show only filters with queued frames.

nozero
Show only filters with non-zero stats.

noeof
Show only filters with non-eof stat.

nodisabled
Show only filters that are enabled in timeline.

flags, f
Set flags which enable which stats are shown in video.

Available values for flags are:

none
All flags turned off.

all All flags turned on.

queue
Display number of queued frames in each link.

frame_count_in
Display number of frames taken from filter.

frame_count_out
Display number of frames given out from filter.

frame_count_delta
Display delta number of frames between above two values.

pts Display current filtered frame pts.

pts_delta
Display pts delta between current and previous frame.

time
Display current filtered frame time.

time_delta
Display time delta between current and previous frame.

timebase
Display time base for filter link.

format
Display used format for filter link.

size
Display video size or number of audio channels in case of audio
used by filter link.

rate
Display video frame rate or sample rate in case of audio used
by filter link.

eof Display link output status.

sample_count_in
Display number of samples taken from filter.

sample_count_out
Display number of samples given out from filter.

sample_count_delta
Display delta number of samples between above two values.

disabled
Show the timeline filter status.

rate, r
Set upper limit for video rate of output stream, Default value is
25. This guarantee that output video frame rate will not be higher
than this value.

grayworld
A color constancy filter that applies color correction based on the
grayworld assumption

See:
<https://www.researchgate.net/publication/275213614_A_New_Color_Correction_Method_for_Underwater_Imaging>

The algorithm uses linear light, so input data should be linearized
beforehand (and possibly correctly tagged).

ffmpeg -i INPUT -vf zscale=transfer=linear,grayworld,zscale=transfer=bt709,format=yuv420p OUTPUT

greyedge
A color constancy variation filter which estimates scene illumination
via grey edge algorithm and corrects the scene colors accordingly.

See: <https://staff.science.uva.nl/th.gevers/pub/GeversTIP07.pdf>

The filter accepts the following options:

difford
The order of differentiation to be applied on the scene. Must be
chosen in the range [0,2] and default value is 1.

minknorm
The Minkowski parameter to be used for calculating the Minkowski
distance. Must be chosen in the range [0,20] and default value is
1. Set to 0 for getting max value instead of calculating Minkowski
distance.

sigma
The standard deviation of Gaussian blur to be applied on the scene.
Must be chosen in the range [0,1024.0] and default value = 1.
floor( sigma * break_off_sigma(3) ) can't be equal to 0 if difford
is greater than 0.

Examples

• Grey Edge:

greyedge=difford=1:minknorm=5:sigma=2

• Max Edge:

greyedge=difford=1:minknorm=0:sigma=2

guided
Apply guided filter for edge-preserving smoothing, dehazing and so on.

The filter accepts the following options:

radius
Set the box radius in pixels. Allowed range is 1 to 20. Default is
3.

eps Set regularization parameter (with square). Allowed range is 0 to
1. Default is 0.01.

mode
Set filter mode. Can be "basic" or "fast". Default is "basic".

sub Set subsampling ratio for "fast" mode. Range is 2 to 64. Default
is 4. No subsampling occurs in "basic" mode.

guidance
Set guidance mode. Can be "off" or "on". Default is "off". If
"off", single input is required. If "on", two inputs of the same
resolution and pixel format are required. The second input serves
as the guidance.

planes
Set planes to filter. Default is first only.

Commands

This filter supports the all above options as commands.

Examples

• Edge-preserving smoothing with guided filter:

ffmpeg -i in.png -vf guided out.png

• Dehazing, structure-transferring filtering, detail enhancement with
guided filter. For the generation of guidance image, refer to
paper "Guided Image Filtering". See:
<http://kaiminghe.com/publications/pami12guidedfilter.pdf>.

ffmpeg -i in.png -i guidance.png -filter_complex guided=guidance=on out.png

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:

clut
Set which CLUT video frames will be processed from second input
stream, can be first or all. Default is all.

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).

This filter also supports the framesync options.

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>.

Commands

This filter supports the "interp" option as commands.

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.

colors_mode
Set colors mode. It accepts the following values:

whiteonblack
blackonwhite
whiteongray
blackongray
coloronblack
coloronwhite
colorongray
blackoncolor
whiteoncolor
grayoncolor

Default is "whiteonblack".

Examples

• Calculate and draw histogram:

ffplay -i input -vf histogram

hqdn3d
This is a high precision/quality 3d denoise filter. It aims to reduce
image noise, producing smooth images and making still images really
still. It should enhance compressibility.

It accepts the following optional parameters:

luma_spatial
A non-negative floating point number which specifies spatial luma
strength. It defaults to 4.0.

chroma_spatial
A non-negative floating point number which specifies spatial chroma
strength. It defaults to 3.0*luma_spatial/4.0.

luma_tmp
A floating point number which specifies luma temporal strength. It
defaults to 6.0*luma_spatial/4.0.

chroma_tmp
A floating point number which specifies chroma temporal strength.
It defaults to luma_tmp*chroma_spatial/luma_spatial.

Commands

This filter supports same commands as options. The command accepts the
same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current
value.

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:

• 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.

• 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.

• 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 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).

The following additional parameters are accepted:

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.

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 accepts 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.

hsvhold
Turns a certain HSV range into gray values.

This filter measures color difference between set HSV color in options
and ones measured in video stream. Depending on options, output colors
can be changed to be gray or not.

The filter accepts the following options:

hue Set the hue value which will be used in color difference
calculation. Allowed range is from -360 to 360. Default value is
0.

sat Set the saturation value which will be used in color difference
calculation. Allowed range is from -1 to 1. Default value is 0.

val Set the value which will be used in color difference calculation.
Allowed range is from -1 to 1. Default value is 0.

similarity
Set similarity percentage with the key color. Allowed range is
from 0 to 1. Default value is 0.01.

0.00001 matches only the exact key color, while 1.0 matches
everything.

blend
Blend percentage. Allowed range is from 0 to 1. Default value is
0.

0.0 makes pixels either fully gray, or not gray at all.

Higher values result in more gray pixels, with a higher gray pixel
the more similar the pixels color is to the key color.

hsvkey
Turns a certain HSV range into transparency.

This filter measures color difference between set HSV color in options
and ones measured in video stream. Depending on options, output colors
can be changed to transparent by adding alpha channel.